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authorSean Bartell <wingedtachikoma@gmail.com>2011-06-24 01:25:21 -0400
committerNils Wallménius <nils@rockbox.org>2012-03-18 12:00:39 +0100
commitb5716df4cb2837bbbc42195cf1aefcf03e21d6a6 (patch)
tree130cd712e2e00893b6df9959a375a8d9523a1aca /apps
parent24bd9d5393dbe39a5c6194877bc00ede669b1d5d (diff)
downloadrockbox-b5716df4cb2837bbbc42195cf1aefcf03e21d6a6.tar.gz
rockbox-b5716df4cb2837bbbc42195cf1aefcf03e21d6a6.zip
Build librbcodec with DSP and metadata.
All associated files are moved to /lib/rbcodec. Change-Id: I572ddd2b8a996aae1e98c081d06b1ed356dce222
Diffstat (limited to 'apps')
-rw-r--r--apps/SOURCES54
-rw-r--r--apps/compressor.c363
-rw-r--r--apps/compressor.h29
-rw-r--r--apps/dsp.c1573
-rw-r--r--apps/dsp.h125
-rw-r--r--apps/dsp_arm.S561
-rw-r--r--apps/dsp_arm_v6.S127
-rw-r--r--apps/dsp_asm.h86
-rw-r--r--apps/dsp_cf.S611
-rw-r--r--apps/eq.c268
-rw-r--r--apps/eq.h50
-rw-r--r--apps/eq_arm.S89
-rw-r--r--apps/eq_cf.S91
-rw-r--r--apps/eqs/Acoustic.cfg17
-rw-r--r--apps/eqs/Bass.cfg17
-rw-r--r--apps/eqs/Classical.cfg17
-rw-r--r--apps/eqs/Default.cfg17
-rw-r--r--apps/eqs/Disco.cfg17
-rw-r--r--apps/eqs/Electronic.cfg17
-rw-r--r--apps/eqs/Hip-Hop.cfg17
-rw-r--r--apps/eqs/Jazz.cfg17
-rw-r--r--apps/eqs/Lounge.cfg17
-rw-r--r--apps/eqs/Pop.cfg17
-rw-r--r--apps/eqs/R&B.cfg17
-rw-r--r--apps/eqs/Rock.cfg17
-rw-r--r--apps/eqs/Vocal.cfg17
-rw-r--r--apps/fracmul.h2
-rw-r--r--apps/metadata.c641
-rw-r--r--apps/metadata.h353
-rw-r--r--apps/metadata/a52.c103
-rw-r--r--apps/metadata/adx.c124
-rw-r--r--apps/metadata/aiff.c108
-rw-r--r--apps/metadata/ape.c182
-rw-r--r--apps/metadata/asap.c254
-rw-r--r--apps/metadata/asf.c591
-rw-r--r--apps/metadata/au.c105
-rw-r--r--apps/metadata/ay.c148
-rw-r--r--apps/metadata/flac.c127
-rw-r--r--apps/metadata/gbs.c65
-rw-r--r--apps/metadata/hes.c39
-rw-r--r--apps/metadata/id3tags.c1199
-rw-r--r--apps/metadata/kss.c53
-rw-r--r--apps/metadata/metadata_common.c374
-rw-r--r--apps/metadata/metadata_common.h69
-rw-r--r--apps/metadata/metadata_parsers.h59
-rw-r--r--apps/metadata/mod.c103
-rw-r--r--apps/metadata/monkeys.c97
-rw-r--r--apps/metadata/mp3.c193
-rw-r--r--apps/metadata/mp4.c842
-rw-r--r--apps/metadata/mpc.c220
-rw-r--r--apps/metadata/nsf.c278
-rw-r--r--apps/metadata/ogg.c215
-rw-r--r--apps/metadata/oma.c189
-rw-r--r--apps/metadata/rm.c464
-rw-r--r--apps/metadata/sgc.c67
-rw-r--r--apps/metadata/sid.c89
-rw-r--r--apps/metadata/smaf.c470
-rw-r--r--apps/metadata/spc.c130
-rw-r--r--apps/metadata/tta.c123
-rw-r--r--apps/metadata/vgm.c195
-rw-r--r--apps/metadata/vorbis.c381
-rw-r--r--apps/metadata/vox.c49
-rw-r--r--apps/metadata/wave.c432
-rw-r--r--apps/metadata/wavpack.c160
-rw-r--r--apps/mp3data.c849
-rw-r--r--apps/mp3data.h89
-rw-r--r--apps/plugins/lrcplayer.c1
-rw-r--r--apps/replaygain.c222
-rw-r--r--apps/replaygain.h34
-rw-r--r--apps/tdspeed.c450
-rw-r--r--apps/tdspeed.h49
71 files changed, 1 insertions, 15234 deletions
diff --git a/apps/SOURCES b/apps/SOURCES
index e1990217ca..45eb0768a3 100644
--- a/apps/SOURCES
+++ b/apps/SOURCES
@@ -26,7 +26,6 @@ menus/audiohw_eq_menu.c
menus/eq_menu.c
buffering.c
voice_thread.c
-replaygain.c
#else /* !SWCODEC */
mpeg.c
#endif
@@ -42,7 +41,6 @@ menus/sound_menu.c
menus/time_menu.c
#endif
misc.c
-mp3data.c
onplay.c
playlist.c
playlist_catalog.c
@@ -168,29 +166,13 @@ pcmbuf.c
codec_thread.c
playback.c
codecs.c
-dsp.c
-compressor.c
#ifndef HAVE_HARDWARE_BEEP
beep.c
#endif
-#ifdef HAVE_PITCHSCREEN
-tdspeed.c
-#endif
#ifdef HAVE_RECORDING
enc_config.c
recorder/pcm_record.c
#endif
-eq.c
-#if defined(CPU_COLDFIRE)
-dsp_cf.S
-eq_cf.S
-#elif defined(CPU_ARM)
-dsp_arm.S
-#if ARM_ARCH >= 6
-dsp_arm_v6.S
-#endif
-eq_arm.S
-#endif
#endif
#ifdef USB_ENABLE_HID
usb_keymaps.c
@@ -198,42 +180,6 @@ usb_keymaps.c
#ifndef USB_NONE
gui/usb_screen.c
#endif
-metadata.c
-metadata/id3tags.c
-metadata/mp3.c
-#if CONFIG_CODEC == SWCODEC
-metadata/metadata_common.c
-metadata/aiff.c
-metadata/ape.c
-metadata/asf.c
-metadata/adx.c
-metadata/flac.c
-metadata/monkeys.c
-metadata/mp4.c
-metadata/mpc.c
-metadata/ogg.c
-metadata/sid.c
-metadata/mod.c
-metadata/spc.c
-metadata/vorbis.c
-metadata/wave.c
-metadata/wavpack.c
-metadata/a52.c
-metadata/asap.c
-metadata/rm.c
-metadata/nsf.c
-metadata/oma.c
-metadata/smaf.c
-metadata/au.c
-metadata/vox.c
-metadata/tta.c
-metadata/ay.c
-metadata/gbs.c
-metadata/hes.c
-metadata/sgc.c
-metadata/vgm.c
-metadata/kss.c
-#endif
#ifdef HAVE_TAGCACHE
tagcache.c
#endif
diff --git a/apps/compressor.c b/apps/compressor.c
deleted file mode 100644
index 3a8d52e4da..0000000000
--- a/apps/compressor.c
+++ /dev/null
@@ -1,363 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2009 Jeffrey Goode
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include "config.h"
-#include "fixedpoint.h"
-#include "fracmul.h"
-#include "settings.h"
-#include "dsp.h"
-#include "compressor.h"
-
-/* Define LOGF_ENABLE to enable logf output in this file */
-/*#define LOGF_ENABLE*/
-#include "logf.h"
-
-static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */
-static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */
-static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */
-static int32_t release_gain IBSS_ATTR; /* S7.24 format */
-
-#define UNITY (1L << 24) /* unity gain in S7.24 format */
-
-/** COMPRESSOR UPDATE
- * Called via the menu system to configure the compressor process */
-bool compressor_update(void)
-{
- static int curr_set[5];
- int new_set[5] = {
- global_settings.compressor_threshold,
- global_settings.compressor_makeup_gain,
- global_settings.compressor_ratio,
- global_settings.compressor_knee,
- global_settings.compressor_release_time};
-
- /* make menu values useful */
- int threshold = new_set[0];
- bool auto_gain = (new_set[1] == 1);
- const int comp_ratios[] = {2, 4, 6, 10, 0};
- int ratio = comp_ratios[new_set[2]];
- bool soft_knee = (new_set[3] == 1);
- int release = new_set[4] * NATIVE_FREQUENCY / 1000;
-
- bool changed = false;
- bool active = (threshold < 0);
-
- for (int i = 0; i < 5; i++)
- {
- if (curr_set[i] != new_set[i])
- {
- changed = true;
- curr_set[i] = new_set[i];
-
-#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
- switch (i)
- {
- case 0:
- logf(" Compressor Threshold: %d dB\tEnabled: %s",
- threshold, active ? "Yes" : "No");
- break;
- case 1:
- logf(" Compressor Makeup Gain: %s",
- auto_gain ? "Auto" : "Off");
- break;
- case 2:
- if (ratio)
- { logf(" Compressor Ratio: %d:1", ratio); }
- else
- { logf(" Compressor Ratio: Limit"); }
- break;
- case 3:
- logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard");
- break;
- case 4:
- logf(" Compressor Release: %d", release);
- break;
- }
-#endif
- }
- }
-
- if (changed && active)
- {
- /* configure variables for compressor operation */
- static const int32_t db[] = {
- /* positive db equivalents in S15.16 format */
- 0x000000, 0x241FA4, 0x1E1A5E, 0x1A94C8,
- 0x181518, 0x1624EA, 0x148F82, 0x1338BD,
- 0x120FD2, 0x1109EB, 0x101FA4, 0x0F4BB6,
- 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E,
- 0x0C0A8C, 0x0B83BE, 0x0B04A5, 0x0A8C6C,
- 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398,
- 0x0884F6, 0x082A30, 0x07D2FA, 0x077F0F,
- 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF,
- 0x060546, 0x05C0DA, 0x057E78, 0x053E03,
- 0x04FF5F, 0x04C273, 0x048726, 0x044D64,
- 0x041518, 0x03DE30, 0x03A89B, 0x037448,
- 0x03412A, 0x030F32, 0x02DE52, 0x02AE80,
- 0x027FB0, 0x0251D6, 0x0224EA, 0x01F8E2,
- 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC,
- 0x0128EB, 0x010190, 0x00DAE4, 0x00B4E1,
- 0x008F82, 0x006AC1, 0x004699, 0x002305};
-
- struct curve_point
- {
- int32_t db; /* S15.16 format */
- int32_t offset; /* S15.16 format */
- } db_curve[5];
-
- /** Set up the shape of the compression curve first as decibel
- values */
- /* db_curve[0] = bottom of knee
- [1] = threshold
- [2] = top of knee
- [3] = 0 db input
- [4] = ~+12db input (2 bits clipping overhead) */
-
- db_curve[1].db = threshold << 16;
- if (soft_knee)
- {
- /* bottom of knee is 3dB below the threshold for soft knee*/
- db_curve[0].db = db_curve[1].db - (3 << 16);
- /* top of knee is 3dB above the threshold for soft knee */
- db_curve[2].db = db_curve[1].db + (3 << 16);
- if (ratio)
- /* offset = -3db * (ratio - 1) / ratio */
- db_curve[2].offset = (int32_t)((long long)(-3 << 16)
- * (ratio - 1) / ratio);
- else
- /* offset = -3db for hard limit */
- db_curve[2].offset = (-3 << 16);
- }
- else
- {
- /* bottom of knee is at the threshold for hard knee */
- db_curve[0].db = threshold << 16;
- /* top of knee is at the threshold for hard knee */
- db_curve[2].db = threshold << 16;
- db_curve[2].offset = 0;
- }
-
- /* Calculate 0db and ~+12db offsets */
- db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
- if (ratio)
- {
- /* offset = threshold * (ratio - 1) / ratio */
- db_curve[3].offset = (int32_t)((long long)(threshold << 16)
- * (ratio - 1) / ratio);
- db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
- * (ratio - 1) / ratio) + db_curve[3].offset;
- }
- else
- {
- /* offset = threshold for hard limit */
- db_curve[3].offset = (threshold << 16);
- db_curve[4].offset = -db_curve[4].db + db_curve[3].offset;
- }
-
- /** Now set up the comp_curve table with compression offsets in the
- form of gain factors in S7.24 format */
- /* comp_curve[0] is 0 (-infinity db) input */
- comp_curve[0] = UNITY;
- /* comp_curve[1 to 63] are intermediate compression values
- corresponding to the 6 MSB of the input values of a non-clipped
- signal */
- for (int i = 1; i < 64; i++)
- {
- /* db constants are stored as positive numbers;
- make them negative here */
- int32_t this_db = -db[i];
-
- /* no compression below the knee */
- if (this_db <= db_curve[0].db)
- comp_curve[i] = UNITY;
-
- /* if soft knee and below top of knee,
- interpolate along soft knee slope */
- else if (soft_knee && (this_db <= db_curve[2].db))
- comp_curve[i] = fp_factor(fp_mul(
- ((this_db - db_curve[0].db) / 6),
- db_curve[2].offset, 16), 16) << 8;
-
- /* interpolate along ratio slope above the knee */
- else
- comp_curve[i] = fp_factor(fp_mul(
- fp_div((db_curve[1].db - this_db), db_curve[1].db, 16),
- db_curve[3].offset, 16), 16) << 8;
- }
- /* comp_curve[64] is the compression level of a maximum level,
- non-clipped signal */
- comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
-
- /* comp_curve[65] is the compression level of a maximum level,
- clipped signal */
- comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8;
-
-#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
- logf("\n *** Compression Offsets ***");
- /* some settings for display only, not used in calculations */
- db_curve[0].offset = 0;
- db_curve[1].offset = 0;
- db_curve[3].db = 0;
-
- for (int i = 0; i <= 4; i++)
- {
- logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i,
- (float)db_curve[i].db / (1 << 16),
- (float)db_curve[i].offset / (1 << 16));
- }
-
- logf("\nGain factors:");
- for (int i = 1; i <= 65; i++)
- {
- debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY);
- if (i % 4 == 0) debugf("\n");
- }
- debugf("\n");
-#endif
-
- /* if using auto peak, then makeup gain is max offset -
- .1dB headroom */
- comp_makeup_gain = auto_gain ?
- fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
- logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY);
-
- /* calculate per-sample gain change a rate of 10db over release time
- */
- comp_rel_slope = 0xAF0BB2 / release;
- logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
-
- release_gain = UNITY;
- }
-
- return active;
-}
-
-/** GET COMPRESSION GAIN
- * Returns the required gain factor in S7.24 format in order to compress the
- * sample in accordance with the compression curve. Always 1 or less.
- */
-static inline int32_t get_compression_gain(struct dsp_data *data,
- int32_t sample)
-{
- const int frac_bits_offset = data->frac_bits - 15;
-
- /* sample must be positive */
- if (sample < 0)
- sample = -(sample + 1);
-
- /* shift sample into 15 frac bit range */
- if (frac_bits_offset > 0)
- sample >>= frac_bits_offset;
- if (frac_bits_offset < 0)
- sample <<= -frac_bits_offset;
-
- /* normal case: sample isn't clipped */
- if (sample < (1 << 15))
- {
- /* index is 6 MSB, rem is 9 LSB */
- int index = sample >> 9;
- int32_t rem = (sample & 0x1FF) << 22;
-
- /* interpolate from the compression curve:
- higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */
- return comp_curve[index] - (FRACMUL(rem,
- (comp_curve[index] - comp_curve[index + 1])));
- }
- /* sample is somewhat clipped, up to 2 bits of overhead */
- if (sample < (1 << 17))
- {
- /* straight interpolation:
- higher gain - ((clipped portion of sample * 4/3
- / (1 << 31)) * (higher gain - lower gain)) */
- return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16,
- (comp_curve[64] - comp_curve[65])));
- }
-
- /* sample is too clipped, return invalid value */
- return -1;
-}
-
-/** COMPRESSOR PROCESS
- * Changes the gain of the samples according to the compressor curve
- */
-void compressor_process(int count, struct dsp_data *data, int32_t *buf[])
-{
- const int num_chan = data->num_channels;
- int32_t *in_buf[2] = {buf[0], buf[1]};
-
- while (count-- > 0)
- {
- int ch;
- /* use lowest (most compressed) gain factor of the output buffer
- sample pair for both samples (mono is also handled correctly here)
- */
- int32_t sample_gain = UNITY;
- for (ch = 0; ch < num_chan; ch++)
- {
- int32_t this_gain = get_compression_gain(data, *in_buf[ch]);
- if (this_gain < sample_gain)
- sample_gain = this_gain;
- }
-
- /* perform release slope; skip if no compression and no release slope
- */
- if ((sample_gain != UNITY) || (release_gain != UNITY))
- {
- /* if larger offset than previous slope, start new release slope
- */
- if ((sample_gain <= release_gain) && (sample_gain > 0))
- {
- release_gain = sample_gain;
- }
- else
- /* keep sloping towards unity gain (and ignore invalid value) */
- {
- release_gain += comp_rel_slope;
- if (release_gain > UNITY)
- {
- release_gain = UNITY;
- }
- }
- }
-
- /* total gain factor is the product of release gain and makeup gain,
- but avoid computation if possible */
- int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain :
- (comp_makeup_gain == UNITY) ? release_gain :
- FRACMUL_SHL(release_gain, comp_makeup_gain, 7));
-
- /* Implement the compressor: apply total gain factor (if any) to the
- output buffer sample pair/mono sample */
- if (total_gain != UNITY)
- {
- for (ch = 0; ch < num_chan; ch++)
- {
- *in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
- }
- }
- in_buf[0]++;
- in_buf[1]++;
- }
-}
-
-void compressor_reset(void)
-{
- release_gain = UNITY;
-}
diff --git a/apps/compressor.h b/apps/compressor.h
deleted file mode 100644
index 6154372e05..0000000000
--- a/apps/compressor.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2009 Jeffrey Goode
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef COMPRESSOR_H
-#define COMPRESSOR_H
-
-void compressor_process(int count, struct dsp_data *data, int32_t *buf[]);
-bool compressor_update(void);
-void compressor_reset(void);
-
-#endif /* COMPRESSOR_H */
diff --git a/apps/dsp.c b/apps/dsp.c
deleted file mode 100644
index 4da555747b..0000000000
--- a/apps/dsp.c
+++ /dev/null
@@ -1,1573 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Miika Pekkarinen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include "config.h"
-#include "system.h"
-#include <sound.h>
-#include "dsp.h"
-#include "dsp-util.h"
-#include "eq.h"
-#include "compressor.h"
-#include "kernel.h"
-#include "settings.h"
-#include "replaygain.h"
-#include "tdspeed.h"
-#include "core_alloc.h"
-#include "fixedpoint.h"
-#include "fracmul.h"
-
-/* Define LOGF_ENABLE to enable logf output in this file */
-/*#define LOGF_ENABLE*/
-#include "logf.h"
-
-/* 16-bit samples are scaled based on these constants. The shift should be
- * no more than 15.
- */
-#define WORD_SHIFT 12
-#define WORD_FRACBITS 27
-
-#define NATIVE_DEPTH 16
-#define SMALL_SAMPLE_BUF_COUNT 128 /* Per channel */
-#define DEFAULT_GAIN 0x01000000
-
-/* enums to index conversion properly with stereo mode and other settings */
-enum
-{
- SAMPLE_INPUT_LE_NATIVE_I_STEREO = STEREO_INTERLEAVED,
- SAMPLE_INPUT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED,
- SAMPLE_INPUT_LE_NATIVE_MONO = STEREO_MONO,
- SAMPLE_INPUT_GT_NATIVE_I_STEREO = STEREO_INTERLEAVED + STEREO_NUM_MODES,
- SAMPLE_INPUT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES,
- SAMPLE_INPUT_GT_NATIVE_MONO = STEREO_MONO + STEREO_NUM_MODES,
- SAMPLE_INPUT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES
-};
-
-enum
-{
- SAMPLE_OUTPUT_MONO = 0,
- SAMPLE_OUTPUT_STEREO,
- SAMPLE_OUTPUT_DITHERED_MONO,
- SAMPLE_OUTPUT_DITHERED_STEREO
-};
-
-/* No asm...yet */
-struct dither_data
-{
- long error[3]; /* 00h */
- long random; /* 0ch */
- /* 10h */
-};
-
-struct crossfeed_data
-{
- int32_t gain; /* 00h - Direct path gain */
- int32_t coefs[3]; /* 04h - Coefficients for the shelving filter */
- int32_t history[4]; /* 10h - Format is x[n - 1], y[n - 1] for both channels */
- int32_t delay[13][2]; /* 20h */
- int32_t *index; /* 88h - Current pointer into the delay line */
- /* 8ch */
-};
-
-/* Current setup is one lowshelf filters three peaking filters and one
- * highshelf filter. Varying the number of shelving filters make no sense,
- * but adding peaking filters is possible.
- */
-struct eq_state
-{
- char enabled[5]; /* 00h - Flags for active filters */
- struct eqfilter filters[5]; /* 08h - packing is 4? */
- /* 10ch */
-};
-
-/* Include header with defines which functions are implemented in assembly
- code for the target */
-#include <dsp_asm.h>
-
-/* Typedefs keep things much neater in this case */
-typedef void (*sample_input_fn_type)(int count, const char *src[],
- int32_t *dst[]);
-typedef int (*resample_fn_type)(int count, struct dsp_data *data,
- const int32_t *src[], int32_t *dst[]);
-typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst);
-
-/* Single-DSP channel processing in place */
-typedef void (*channels_process_fn_type)(int count, int32_t *buf[]);
-/* DSP local channel processing in place */
-typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
- int32_t *buf[]);
-
-/*
- ***************************************************************************/
-
-struct dsp_config
-{
- struct dsp_data data; /* Config members for use in external routines */
- long codec_frequency; /* Sample rate of data coming from the codec */
- long frequency; /* Effective sample rate after pitch shift (if any) */
- int sample_depth;
- int sample_bytes;
- int stereo_mode;
- int32_t tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */
-#ifdef HAVE_PITCHSCREEN
- bool tdspeed_active; /* Timestretch is in use */
-#endif
-#ifdef HAVE_SW_TONE_CONTROLS
- /* Filter struct for software bass/treble controls */
- struct eqfilter tone_filter;
-#endif
- /* Functions that change depending upon settings - NULL if stage is
- disabled */
- sample_input_fn_type input_samples;
- resample_fn_type resample;
- sample_output_fn_type output_samples;
- /* These will be NULL for the voice codec and is more economical that
- way */
- channels_process_dsp_fn_type apply_gain;
- channels_process_fn_type apply_crossfeed;
- channels_process_fn_type eq_process;
- channels_process_fn_type channels_process;
- channels_process_dsp_fn_type compressor_process;
-};
-
-/* General DSP config */
-static struct dsp_config dsp_conf[2] IBSS_ATTR; /* 0=A, 1=V */
-/* Dithering */
-static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */
-static long dither_mask IBSS_ATTR;
-static long dither_bias IBSS_ATTR;
-/* Crossfeed */
-struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */
-{
- .index = (int32_t *)crossfeed_data.delay
-};
-
-/* Equalizer */
-static struct eq_state eq_data; /* A */
-
-/* Software tone controls */
-#ifdef HAVE_SW_TONE_CONTROLS
-static int prescale; /* A/V */
-static int bass; /* A/V */
-static int treble; /* A/V */
-#endif
-
-/* Settings applicable to audio codec only */
-#ifdef HAVE_PITCHSCREEN
-static int32_t pitch_ratio = PITCH_SPEED_100;
-static int big_sample_locks;
-#endif
-static int channels_mode;
- long dsp_sw_gain;
- long dsp_sw_cross;
-static bool dither_enabled;
-static long eq_precut;
-static long track_gain;
-static bool new_gain;
-static long album_gain;
-static long track_peak;
-static long album_peak;
-static long replaygain;
-static bool crossfeed_enabled;
-
-#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
-#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
-
-/* The internal format is 32-bit samples, non-interleaved, stereo. This
- * format is similar to the raw output from several codecs, so the amount
- * of copying needed is minimized for that case.
- */
-
-#define RESAMPLE_RATIO 4 /* Enough for 11,025 Hz -> 44,100 Hz */
-#define SMALL_RESAMPLE_BUF_COUNT (SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO)
-#define BIG_SAMPLE_BUF_COUNT SMALL_RESAMPLE_BUF_COUNT
-#define BIG_RESAMPLE_BUF_COUNT (BIG_SAMPLE_BUF_COUNT * RESAMPLE_RATIO)
-
-static int32_t small_sample_buf[2][SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
-static int32_t small_resample_buf[2][SMALL_RESAMPLE_BUF_COUNT] IBSS_ATTR;
-
-#ifdef HAVE_PITCHSCREEN
-static int32_t (* big_sample_buf)[BIG_SAMPLE_BUF_COUNT] = NULL;
-static int32_t (* big_resample_buf)[BIG_RESAMPLE_BUF_COUNT] = NULL;
-#endif
-
-static int sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
-static int32_t *sample_buf[2] = { small_sample_buf[0], small_sample_buf[1] };
-static int resample_buf_count = SMALL_RESAMPLE_BUF_COUNT;
-static int32_t *resample_buf[2] = { small_resample_buf[0], small_resample_buf[1] };
-
-#ifdef HAVE_PITCHSCREEN
-int32_t sound_get_pitch(void)
-{
- return pitch_ratio;
-}
-
-void sound_set_pitch(int32_t percent)
-{
- pitch_ratio = percent;
- dsp_configure(&AUDIO_DSP, DSP_SWITCH_FREQUENCY,
- AUDIO_DSP.codec_frequency);
-}
-
-static void tdspeed_set_pointers( bool time_stretch_active )
-{
- if( time_stretch_active )
- {
- sample_buf_count = BIG_SAMPLE_BUF_COUNT;
- resample_buf_count = BIG_RESAMPLE_BUF_COUNT;
- sample_buf[0] = big_sample_buf[0];
- sample_buf[1] = big_sample_buf[1];
- resample_buf[0] = big_resample_buf[0];
- resample_buf[1] = big_resample_buf[1];
- }
- else
- {
- sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
- resample_buf_count = SMALL_RESAMPLE_BUF_COUNT;
- sample_buf[0] = small_sample_buf[0];
- sample_buf[1] = small_sample_buf[1];
- resample_buf[0] = small_resample_buf[0];
- resample_buf[1] = small_resample_buf[1];
- }
-}
-
-static void tdspeed_setup(struct dsp_config *dspc)
-{
- /* Assume timestretch will not be used */
- dspc->tdspeed_active = false;
-
- tdspeed_set_pointers( false );
-
- if (!dsp_timestretch_available())
- return; /* Timestretch not enabled or buffer not allocated */
-
- if (dspc->tdspeed_percent == 0)
- dspc->tdspeed_percent = PITCH_SPEED_100;
-
- if (!tdspeed_config(
- dspc->codec_frequency == 0 ? NATIVE_FREQUENCY : dspc->codec_frequency,
- dspc->stereo_mode != STEREO_MONO,
- dspc->tdspeed_percent))
- return; /* Timestretch not possible or needed with these parameters */
-
- /* Timestretch is to be used */
- dspc->tdspeed_active = true;
-
- tdspeed_set_pointers( true );
-}
-
-
-static int move_callback(int handle, void* current, void* new)
-{
- (void)handle;(void)current;
-
- if ( big_sample_locks > 0 )
- return BUFLIB_CB_CANNOT_MOVE;
-
- big_sample_buf = new;
-
- /* no allocation without timestretch enabled */
- tdspeed_set_pointers( true );
- return BUFLIB_CB_OK;
-}
-
-static void lock_sample_buf( bool lock )
-{
- if ( lock )
- big_sample_locks++;
- else
- big_sample_locks--;
-}
-
-static struct buflib_callbacks ops = {
- .move_callback = move_callback,
- .shrink_callback = NULL,
-};
-
-
-void dsp_timestretch_enable(bool enabled)
-{
- /* Hook to set up timestretch buffer on first call to settings_apply() */
- static int handle = -1;
- if (enabled)
- {
- if (big_sample_buf)
- return; /* already allocated and enabled */
-
- /* Set up timestretch buffers */
- big_sample_buf = &small_resample_buf[0];
- handle = core_alloc_ex("resample buf",
- 2 * BIG_RESAMPLE_BUF_COUNT * sizeof(int32_t),
- &ops);
- big_sample_locks = 0;
- enabled = handle >= 0;
-
- if (enabled)
- {
- /* success, now setup tdspeed */
- big_resample_buf = core_get_data(handle);
-
- tdspeed_init();
- tdspeed_setup(&AUDIO_DSP);
- }
- }
-
- if (!enabled)
- {
- dsp_set_timestretch(PITCH_SPEED_100);
- tdspeed_finish();
-
- if (handle >= 0)
- core_free(handle);
-
- handle = -1;
- big_sample_buf = NULL;
- }
-}
-
-void dsp_set_timestretch(int32_t percent)
-{
- AUDIO_DSP.tdspeed_percent = percent;
- tdspeed_setup(&AUDIO_DSP);
-}
-
-int32_t dsp_get_timestretch()
-{
- return AUDIO_DSP.tdspeed_percent;
-}
-
-bool dsp_timestretch_available()
-{
- return (global_settings.timestretch_enabled && big_sample_buf);
-}
-#endif /* HAVE_PITCHSCREEN */
-
-/* Convert count samples to the internal format, if needed. Updates src
- * to point past the samples "consumed" and dst is set to point to the
- * samples to consume. Note that for mono, dst[0] equals dst[1], as there
- * is no point in processing the same data twice.
- */
-
-/* convert count 16-bit mono to 32-bit mono */
-static void sample_input_lte_native_mono(
- int count, const char *src[], int32_t *dst[])
-{
- const int16_t *s = (int16_t *) src[0];
- const int16_t * const send = s + count;
- int32_t *d = dst[0] = dst[1] = sample_buf[0];
- int scale = WORD_SHIFT;
-
- while (s < send)
- {
- *d++ = *s++ << scale;
- }
-
- src[0] = (char *)s;
-}
-
-/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
-static void sample_input_lte_native_i_stereo(
- int count, const char *src[], int32_t *dst[])
-{
- const int32_t *s = (int32_t *) src[0];
- const int32_t * const send = s + count;
- int32_t *dl = dst[0] = sample_buf[0];
- int32_t *dr = dst[1] = sample_buf[1];
- int scale = WORD_SHIFT;
-
- while (s < send)
- {
- int32_t slr = *s++;
-#ifdef ROCKBOX_LITTLE_ENDIAN
- *dl++ = (slr >> 16) << scale;
- *dr++ = (int32_t)(int16_t)slr << scale;
-#else /* ROCKBOX_BIG_ENDIAN */
- *dl++ = (int32_t)(int16_t)slr << scale;
- *dr++ = (slr >> 16) << scale;
-#endif
- }
-
- src[0] = (char *)s;
-}
-
-/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
-static void sample_input_lte_native_ni_stereo(
- int count, const char *src[], int32_t *dst[])
-{
- const int16_t *sl = (int16_t *) src[0];
- const int16_t *sr = (int16_t *) src[1];
- const int16_t * const slend = sl + count;
- int32_t *dl = dst[0] = sample_buf[0];
- int32_t *dr = dst[1] = sample_buf[1];
- int scale = WORD_SHIFT;
-
- while (sl < slend)
- {
- *dl++ = *sl++ << scale;
- *dr++ = *sr++ << scale;
- }
-
- src[0] = (char *)sl;
- src[1] = (char *)sr;
-}
-
-/* convert count 32-bit mono to 32-bit mono */
-static void sample_input_gt_native_mono(
- int count, const char *src[], int32_t *dst[])
-{
- dst[0] = dst[1] = (int32_t *)src[0];
- src[0] = (char *)(dst[0] + count);
-}
-
-/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
-static void sample_input_gt_native_i_stereo(
- int count, const char *src[], int32_t *dst[])
-{
- const int32_t *s = (int32_t *)src[0];
- const int32_t * const send = s + 2*count;
- int32_t *dl = dst[0] = sample_buf[0];
- int32_t *dr = dst[1] = sample_buf[1];
-
- while (s < send)
- {
- *dl++ = *s++;
- *dr++ = *s++;
- }
-
- src[0] = (char *)send;
-}
-
-/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
-static void sample_input_gt_native_ni_stereo(
- int count, const char *src[], int32_t *dst[])
-{
- dst[0] = (int32_t *)src[0];
- dst[1] = (int32_t *)src[1];
- src[0] = (char *)(dst[0] + count);
- src[1] = (char *)(dst[1] + count);
-}
-
-/**
- * sample_input_new_format()
- *
- * set the to-native sample conversion function based on dsp sample parameters
- *
- * !DSPPARAMSYNC
- * needs syncing with changes to the following dsp parameters:
- * * dsp->stereo_mode (A/V)
- * * dsp->sample_depth (A/V)
- */
-static void sample_input_new_format(struct dsp_config *dsp)
-{
- static const sample_input_fn_type sample_input_functions[] =
- {
- [SAMPLE_INPUT_LE_NATIVE_I_STEREO] = sample_input_lte_native_i_stereo,
- [SAMPLE_INPUT_LE_NATIVE_NI_STEREO] = sample_input_lte_native_ni_stereo,
- [SAMPLE_INPUT_LE_NATIVE_MONO] = sample_input_lte_native_mono,
- [SAMPLE_INPUT_GT_NATIVE_I_STEREO] = sample_input_gt_native_i_stereo,
- [SAMPLE_INPUT_GT_NATIVE_NI_STEREO] = sample_input_gt_native_ni_stereo,
- [SAMPLE_INPUT_GT_NATIVE_MONO] = sample_input_gt_native_mono,
- };
-
- int convert = dsp->stereo_mode;
-
- if (dsp->sample_depth > NATIVE_DEPTH)
- convert += SAMPLE_INPUT_GT_NATIVE_1ST_INDEX;
-
- dsp->input_samples = sample_input_functions[convert];
-}
-
-
-#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
-/* write mono internal format to output format */
-static void sample_output_mono(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst)
-{
- const int32_t *s0 = src[0];
- const int scale = data->output_scale;
- const int dc_bias = 1 << (scale - 1);
-
- while (count-- > 0)
- {
- int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
- *dst++ = lr;
- *dst++ = lr;
- }
-}
-#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */
-
-/* write stereo internal format to output format */
-#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
-static void sample_output_stereo(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst)
-{
- const int32_t *s0 = src[0];
- const int32_t *s1 = src[1];
- const int scale = data->output_scale;
- const int dc_bias = 1 << (scale - 1);
-
- while (count-- > 0)
- {
- *dst++ = clip_sample_16((*s0++ + dc_bias) >> scale);
- *dst++ = clip_sample_16((*s1++ + dc_bias) >> scale);
- }
-}
-#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */
-
-/**
- * The "dither" code to convert the 24-bit samples produced by libmad was
- * taken from the coolplayer project - coolplayer.sourceforge.net
- *
- * This function handles mono and stereo outputs.
- */
-static void sample_output_dithered(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst)
-{
- const int32_t mask = dither_mask;
- const int32_t bias = dither_bias;
- const int scale = data->output_scale;
- const int32_t min = data->clip_min;
- const int32_t max = data->clip_max;
- const int32_t range = max - min;
- int ch;
- int16_t *d;
-
- for (ch = 0; ch < data->num_channels; ch++)
- {
- struct dither_data * const dither = &dither_data[ch];
- const int32_t *s = src[ch];
- int i;
-
- for (i = 0, d = &dst[ch]; i < count; i++, s++, d += 2)
- {
- int32_t output, sample;
- int32_t random;
-
- /* Noise shape and bias (for correct rounding later) */
- sample = *s;
- sample += dither->error[0] - dither->error[1] + dither->error[2];
- dither->error[2] = dither->error[1];
- dither->error[1] = dither->error[0]/2;
-
- output = sample + bias;
-
- /* Dither, highpass triangle PDF */
- random = dither->random*0x0019660dL + 0x3c6ef35fL;
- output += (random & mask) - (dither->random & mask);
- dither->random = random;
-
- /* Round sample to output range */
- output &= ~mask;
-
- /* Error feedback */
- dither->error[0] = sample - output;
-
- /* Clip */
- if ((uint32_t)(output - min) > (uint32_t)range)
- {
- int32_t c = min;
- if (output > min)
- c += range;
- output = c;
- }
-
- /* Quantize and store */
- *d = output >> scale;
- }
- }
-
- if (data->num_channels == 2)
- return;
-
- /* Have to duplicate left samples into the right channel since
- pcm buffer and hardware is interleaved stereo */
- d = &dst[0];
-
- while (count-- > 0)
- {
- int16_t s = *d++;
- *d++ = s;
- }
-}
-
-/**
- * sample_output_new_format()
- *
- * set the from-native to ouput sample conversion routine
- *
- * !DSPPARAMSYNC
- * needs syncing with changes to the following dsp parameters:
- * * dsp->stereo_mode (A/V)
- * * dither_enabled (A)
- */
-static void sample_output_new_format(struct dsp_config *dsp)
-{
- static const sample_output_fn_type sample_output_functions[] =
- {
- sample_output_mono,
- sample_output_stereo,
- sample_output_dithered,
- sample_output_dithered
- };
-
- int out = dsp->data.num_channels - 1;
-
- if (dsp == &AUDIO_DSP && dither_enabled)
- out += 2;
-
- dsp->output_samples = sample_output_functions[out];
-}
-
-/**
- * Linear interpolation resampling that introduces a one sample delay because
- * of our inability to look into the future at the end of a frame.
- */
-#ifndef DSP_HAVE_ASM_RESAMPLING
-static int dsp_downsample(int count, struct dsp_data *data,
- const int32_t *src[], int32_t *dst[])
-{
- int ch = data->num_channels - 1;
- uint32_t delta = data->resample_data.delta;
- uint32_t phase, pos;
- int32_t *d;
-
- /* Rolled channel loop actually showed slightly faster. */
- do
- {
- /* Just initialize things and not worry too much about the relatively
- * uncommon case of not being able to spit out a sample for the frame.
- */
- const int32_t *s = src[ch];
- int32_t last = data->resample_data.last_sample[ch];
-
- data->resample_data.last_sample[ch] = s[count - 1];
- d = dst[ch];
- phase = data->resample_data.phase;
- pos = phase >> 16;
-
- /* Do we need last sample of previous frame for interpolation? */
- if (pos > 0)
- last = s[pos - 1];
-
- while (pos < (uint32_t)count)
- {
- *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
- phase += delta;
- pos = phase >> 16;
- last = s[pos - 1];
- }
- }
- while (--ch >= 0);
-
- /* Wrap phase accumulator back to start of next frame. */
- data->resample_data.phase = phase - (count << 16);
- return d - dst[0];
-}
-
-static int dsp_upsample(int count, struct dsp_data *data,
- const int32_t *src[], int32_t *dst[])
-{
- int ch = data->num_channels - 1;
- uint32_t delta = data->resample_data.delta;
- uint32_t phase, pos;
- int32_t *d;
-
- /* Rolled channel loop actually showed slightly faster. */
- do
- {
- /* Should always be able to output a sample for a ratio up to RESAMPLE_RATIO */
- const int32_t *s = src[ch];
- int32_t last = data->resample_data.last_sample[ch];
-
- data->resample_data.last_sample[ch] = s[count - 1];
- d = dst[ch];
- phase = data->resample_data.phase;
- pos = phase >> 16;
-
- while (pos == 0)
- {
- *d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last);
- phase += delta;
- pos = phase >> 16;
- }
-
- while (pos < (uint32_t)count)
- {
- last = s[pos - 1];
- *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
- phase += delta;
- pos = phase >> 16;
- }
- }
- while (--ch >= 0);
-
- /* Wrap phase accumulator back to start of next frame. */
- data->resample_data.phase = phase & 0xffff;
- return d - dst[0];
-}
-#endif /* DSP_HAVE_ASM_RESAMPLING */
-
-static void resampler_new_delta(struct dsp_config *dsp)
-{
- dsp->data.resample_data.delta = (unsigned long)
- dsp->frequency * 65536LL / NATIVE_FREQUENCY;
-
- if (dsp->frequency == NATIVE_FREQUENCY)
- {
- /* NOTE: If fully glitch-free transistions from no resampling to
- resampling are desired, last_sample history should be maintained
- even when not resampling. */
- dsp->resample = NULL;
- dsp->data.resample_data.phase = 0;
- dsp->data.resample_data.last_sample[0] = 0;
- dsp->data.resample_data.last_sample[1] = 0;
- }
- else if (dsp->frequency < NATIVE_FREQUENCY)
- dsp->resample = dsp_upsample;
- else
- dsp->resample = dsp_downsample;
-}
-
-/* Resample count stereo samples. Updates the src array, if resampling is
- * done, to refer to the resampled data. Returns number of stereo samples
- * for further processing.
- */
-static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
-{
- int32_t *dst[2] =
- {
- resample_buf[0],
- resample_buf[1]
- };
- lock_sample_buf( true );
- count = dsp->resample(count, &dsp->data, (const int32_t **)src, dst);
-
- src[0] = dst[0];
- src[1] = dst[dsp->data.num_channels - 1];
- lock_sample_buf( false );
- return count;
-}
-
-static void dither_init(struct dsp_config *dsp)
-{
- memset(dither_data, 0, sizeof (dither_data));
- dither_bias = (1L << (dsp->data.frac_bits - NATIVE_DEPTH));
- dither_mask = (1L << (dsp->data.frac_bits + 1 - NATIVE_DEPTH)) - 1;
-}
-
-void dsp_dither_enable(bool enable)
-{
- struct dsp_config *dsp = &AUDIO_DSP;
- dither_enabled = enable;
- sample_output_new_format(dsp);
-}
-
-/* Applies crossfeed to the stereo signal in src.
- * Crossfeed is a process where listening over speakers is simulated. This
- * is good for old hard panned stereo records, which might be quite fatiguing
- * to listen to on headphones with no crossfeed.
- */
-#ifndef DSP_HAVE_ASM_CROSSFEED
-static void apply_crossfeed(int count, int32_t *buf[])
-{
- int32_t *hist_l = &crossfeed_data.history[0];
- int32_t *hist_r = &crossfeed_data.history[2];
- int32_t *delay = &crossfeed_data.delay[0][0];
- int32_t *coefs = &crossfeed_data.coefs[0];
- int32_t gain = crossfeed_data.gain;
- int32_t *di = crossfeed_data.index;
-
- int32_t acc;
- int32_t left, right;
- int i;
-
- for (i = 0; i < count; i++)
- {
- left = buf[0][i];
- right = buf[1][i];
-
- /* Filter delayed sample from left speaker */
- acc = FRACMUL(*di, coefs[0]);
- acc += FRACMUL(hist_l[0], coefs[1]);
- acc += FRACMUL(hist_l[1], coefs[2]);
- /* Save filter history for left speaker */
- hist_l[1] = acc;
- hist_l[0] = *di;
- *di++ = left;
- /* Filter delayed sample from right speaker */
- acc = FRACMUL(*di, coefs[0]);
- acc += FRACMUL(hist_r[0], coefs[1]);
- acc += FRACMUL(hist_r[1], coefs[2]);
- /* Save filter history for right speaker */
- hist_r[1] = acc;
- hist_r[0] = *di;
- *di++ = right;
- /* Now add the attenuated direct sound and write to outputs */
- buf[0][i] = FRACMUL(left, gain) + hist_r[1];
- buf[1][i] = FRACMUL(right, gain) + hist_l[1];
-
- /* Wrap delay line index if bigger than delay line size */
- if (di >= delay + 13*2)
- di = delay;
- }
- /* Write back local copies of data we've modified */
- crossfeed_data.index = di;
-}
-#endif /* DSP_HAVE_ASM_CROSSFEED */
-
-/**
- * dsp_set_crossfeed(bool enable)
- *
- * !DSPPARAMSYNC
- * needs syncing with changes to the following dsp parameters:
- * * dsp->stereo_mode (A)
- */
-void dsp_set_crossfeed(bool enable)
-{
- crossfeed_enabled = enable;
- AUDIO_DSP.apply_crossfeed = (enable && AUDIO_DSP.data.num_channels > 1)
- ? apply_crossfeed : NULL;
-}
-
-void dsp_set_crossfeed_direct_gain(int gain)
-{
- crossfeed_data.gain = get_replaygain_int(gain * 10) << 7;
- /* If gain is negative, the calculation overflowed and we need to clamp */
- if (crossfeed_data.gain < 0)
- crossfeed_data.gain = 0x7fffffff;
-}
-
-/* Both gains should be below 0 dB */
-void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
-{
- int32_t *c = crossfeed_data.coefs;
- long scaler = get_replaygain_int(lf_gain * 10) << 7;
-
- cutoff = 0xffffffff/NATIVE_FREQUENCY*cutoff;
- hf_gain -= lf_gain;
- /* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
- * point instead of shelf midpoint. This is for compatibility with the old
- * crossfeed shelf filter and should be removed if crossfeed settings are
- * ever made incompatible for any other good reason.
- */
- cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
- filter_shelf_coefs(cutoff, hf_gain, false, c);
- /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
- * over 1 and can do this safely
- */
- c[0] = FRACMUL_SHL(c[0], scaler, 4);
- c[1] = FRACMUL_SHL(c[1], scaler, 4);
- c[2] <<= 4;
-}
-
-/* Apply a constant gain to the samples (e.g., for ReplayGain).
- * Note that this must be called before the resampler.
- */
-#ifndef DSP_HAVE_ASM_APPLY_GAIN
-static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
-{
- const int32_t gain = data->gain;
- int ch;
-
- for (ch = 0; ch < data->num_channels; ch++)
- {
- int32_t *d = buf[ch];
- int i;
-
- for (i = 0; i < count; i++)
- d[i] = FRACMUL_SHL(d[i], gain, 8);
- }
-}
-#endif /* DSP_HAVE_ASM_APPLY_GAIN */
-
-/* Combine all gains to a global gain. */
-static void set_gain(struct dsp_config *dsp)
-{
- /* gains are in S7.24 format */
- dsp->data.gain = DEFAULT_GAIN;
-
- /* Replay gain not relevant to voice */
- if (dsp == &AUDIO_DSP && replaygain)
- {
- dsp->data.gain = replaygain;
- }
-
- if (dsp->eq_process && eq_precut)
- {
- dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24);
- }
-
-#ifdef HAVE_SW_VOLUME_CONTROL
- if (global_settings.volume < SW_VOLUME_MAX ||
- global_settings.volume > SW_VOLUME_MIN)
- {
- int vol_gain = get_replaygain_int(global_settings.volume * 100);
- dsp->data.gain = (long) (((int64_t) dsp->data.gain * vol_gain) >> 24);
- }
-#endif
-
- if (dsp->data.gain == DEFAULT_GAIN)
- {
- dsp->data.gain = 0;
- }
- else
- {
- dsp->data.gain >>= 1; /* convert gain to S8.23 format */
- }
-
- dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
-}
-
-/**
- * Update the amount to cut the audio before applying the equalizer.
- *
- * @param precut to apply in decibels (multiplied by 10)
- */
-void dsp_set_eq_precut(int precut)
-{
- eq_precut = get_replaygain_int(precut * -10);
- set_gain(&AUDIO_DSP);
-}
-
-/**
- * Synchronize the equalizer filter coefficients with the global settings.
- *
- * @param band the equalizer band to synchronize
- */
-void dsp_set_eq_coefs(int band)
-{
- /* Adjust setting pointer to the band we actually want to change */
- struct eq_band_setting *setting = &global_settings.eq_band_settings[band];
-
- /* Convert user settings to format required by coef generator functions */
- unsigned long cutoff = 0xffffffff / NATIVE_FREQUENCY * setting->cutoff;
- unsigned long q = setting->q;
- int gain = setting->gain;
-
- if (q == 0)
- q = 1;
-
- /* NOTE: The coef functions assume the EMAC unit is in fractional mode,
- which it should be, since we're executed from the main thread. */
-
- /* Assume a band is disabled if the gain is zero */
- if (gain == 0)
- {
- eq_data.enabled[band] = 0;
- }
- else
- {
- if (band == 0)
- eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
- else if (band == 4)
- eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
- else
- eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
-
- eq_data.enabled[band] = 1;
- }
-}
-
-/* Apply EQ filters to those bands that have got it switched on. */
-static void eq_process(int count, int32_t *buf[])
-{
- static const int shifts[] =
- {
- EQ_SHELF_SHIFT, /* low shelf */
- EQ_PEAK_SHIFT, /* peaking */
- EQ_PEAK_SHIFT, /* peaking */
- EQ_PEAK_SHIFT, /* peaking */
- EQ_SHELF_SHIFT, /* high shelf */
- };
- unsigned int channels = AUDIO_DSP.data.num_channels;
- int i;
-
- /* filter configuration currently is 1 low shelf filter, 3 band peaking
- filters and 1 high shelf filter, in that order. we need to know this
- so we can choose the correct shift factor.
- */
- for (i = 0; i < 5; i++)
- {
- if (!eq_data.enabled[i])
- continue;
- eq_filter(buf, &eq_data.filters[i], count, channels, shifts[i]);
- }
-}
-
-/**
- * Use to enable the equalizer.
- *
- * @param enable true to enable the equalizer
- */
-void dsp_set_eq(bool enable)
-{
- AUDIO_DSP.eq_process = enable ? eq_process : NULL;
- set_gain(&AUDIO_DSP);
-}
-
-static void dsp_set_stereo_width(int value)
-{
- long width, straight, cross;
-
- width = value * 0x7fffff / 100;
-
- if (value <= 100)
- {
- straight = (0x7fffff + width) / 2;
- cross = straight - width;
- }
- else
- {
- /* straight = (1 + width) / (2 * width) */
- straight = ((int64_t)(0x7fffff + width) << 22) / width;
- cross = straight - 0x7fffff;
- }
-
- dsp_sw_gain = straight << 8;
- dsp_sw_cross = cross << 8;
-}
-
-/**
- * Implements the different channel configurations and stereo width.
- */
-
-/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
- * completeness. */
-#if 0
-static void channels_process_sound_chan_stereo(int count, int32_t *buf[])
-{
- /* The channels are each just themselves */
- (void)count; (void)buf;
-}
-#endif
-
-#ifndef DSP_HAVE_ASM_SOUND_CHAN_MONO
-static void channels_process_sound_chan_mono(int count, int32_t *buf[])
-{
- int32_t *sl = buf[0], *sr = buf[1];
-
- while (count-- > 0)
- {
- int32_t lr = *sl/2 + *sr/2;
- *sl++ = lr;
- *sr++ = lr;
- }
-}
-#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */
-
-#ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
-static void channels_process_sound_chan_custom(int count, int32_t *buf[])
-{
- const int32_t gain = dsp_sw_gain;
- const int32_t cross = dsp_sw_cross;
- int32_t *sl = buf[0], *sr = buf[1];
-
- while (count-- > 0)
- {
- int32_t l = *sl;
- int32_t r = *sr;
- *sl++ = FRACMUL(l, gain) + FRACMUL(r, cross);
- *sr++ = FRACMUL(r, gain) + FRACMUL(l, cross);
- }
-}
-#endif /* DSP_HAVE_ASM_SOUND_CHAN_CUSTOM */
-
-static void channels_process_sound_chan_mono_left(int count, int32_t *buf[])
-{
- /* Just copy over the other channel */
- memcpy(buf[1], buf[0], count * sizeof (*buf));
-}
-
-static void channels_process_sound_chan_mono_right(int count, int32_t *buf[])
-{
- /* Just copy over the other channel */
- memcpy(buf[0], buf[1], count * sizeof (*buf));
-}
-
-#ifndef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
-static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
-{
- int32_t *sl = buf[0], *sr = buf[1];
-
- while (count-- > 0)
- {
- int32_t ch = *sl/2 - *sr/2;
- *sl++ = ch;
- *sr++ = -ch;
- }
-}
-#endif /* DSP_HAVE_ASM_SOUND_CHAN_KARAOKE */
-
-static void dsp_set_channel_config(int value)
-{
- static const channels_process_fn_type channels_process_functions[] =
- {
- /* SOUND_CHAN_STEREO = All-purpose index for no channel processing */
- [SOUND_CHAN_STEREO] = NULL,
- [SOUND_CHAN_MONO] = channels_process_sound_chan_mono,
- [SOUND_CHAN_CUSTOM] = channels_process_sound_chan_custom,
- [SOUND_CHAN_MONO_LEFT] = channels_process_sound_chan_mono_left,
- [SOUND_CHAN_MONO_RIGHT] = channels_process_sound_chan_mono_right,
- [SOUND_CHAN_KARAOKE] = channels_process_sound_chan_karaoke,
- };
-
- if ((unsigned)value >= ARRAYLEN(channels_process_functions) ||
- AUDIO_DSP.stereo_mode == STEREO_MONO)
- {
- value = SOUND_CHAN_STEREO;
- }
-
- /* This doesn't apply to voice */
- channels_mode = value;
- AUDIO_DSP.channels_process = channels_process_functions[value];
-}
-
-#if CONFIG_CODEC == SWCODEC
-
-#ifdef HAVE_SW_TONE_CONTROLS
-static void set_tone_controls(void)
-{
- filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
- 0xffffffff/NATIVE_FREQUENCY*3500,
- bass, treble, -prescale,
- AUDIO_DSP.tone_filter.coefs);
- /* Sync the voice dsp coefficients */
- memcpy(&VOICE_DSP.tone_filter.coefs, AUDIO_DSP.tone_filter.coefs,
- sizeof (VOICE_DSP.tone_filter.coefs));
-}
-#endif
-
-/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
- * code directly.
- */
-int dsp_callback(int msg, intptr_t param)
-{
- switch (msg)
- {
-#ifdef HAVE_SW_TONE_CONTROLS
- case DSP_CALLBACK_SET_PRESCALE:
- prescale = param;
- set_tone_controls();
- break;
- /* prescaler is always set after calling any of these, so we wait with
- * calculating coefs until the above case is hit.
- */
- case DSP_CALLBACK_SET_BASS:
- bass = param;
- break;
- case DSP_CALLBACK_SET_TREBLE:
- treble = param;
- break;
-#ifdef HAVE_SW_VOLUME_CONTROL
- case DSP_CALLBACK_SET_SW_VOLUME:
- set_gain(&AUDIO_DSP);
- break;
-#endif
-#endif
- case DSP_CALLBACK_SET_CHANNEL_CONFIG:
- dsp_set_channel_config(param);
- break;
- case DSP_CALLBACK_SET_STEREO_WIDTH:
- dsp_set_stereo_width(param);
- break;
- default:
- break;
- }
- return 0;
-}
-#endif
-
-/* Process and convert src audio to dst based on the DSP configuration,
- * reading count number of audio samples. dst is assumed to be large
- * enough; use dsp_output_count() to get the required number. src is an
- * array of pointers; for mono and interleaved stereo, it contains one
- * pointer to the start of the audio data and the other is ignored; for
- * non-interleaved stereo, it contains two pointers, one for each audio
- * channel. Returns number of bytes written to dst.
- */
-int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
-{
- static int32_t *tmp[2]; /* tdspeed_doit() needs it static */
- static long last_yield;
- long tick;
- int written = 0;
-
-#if defined(CPU_COLDFIRE)
- /* set emac unit for dsp processing, and save old macsr, we're running in
- codec thread context at this point, so can't clobber it */
- unsigned long old_macsr = coldfire_get_macsr();
- coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
-#endif
-
- if (new_gain)
- dsp_set_replaygain(); /* Gain has changed */
-
- /* Perform at least one yield before starting */
- last_yield = current_tick;
- yield();
-
- /* Testing function pointers for NULL is preferred since the pointer
- will be preloaded to be used for the call if not. */
- while (count > 0)
- {
- int samples = MIN(sample_buf_count, count);
- count -= samples;
-
- dsp->input_samples(samples, src, tmp);
-
-#ifdef HAVE_PITCHSCREEN
- if (dsp->tdspeed_active)
- samples = tdspeed_doit(tmp, samples);
-#endif
-
- int chunk_offset = 0;
- while (samples > 0)
- {
- int32_t *t2[2];
- t2[0] = tmp[0]+chunk_offset;
- t2[1] = tmp[1]+chunk_offset;
-
- int chunk = MIN(sample_buf_count, samples);
- chunk_offset += chunk;
- samples -= chunk;
-
- if (dsp->apply_gain)
- dsp->apply_gain(chunk, &dsp->data, t2);
-
- if (dsp->resample && (chunk = resample(dsp, chunk, t2)) <= 0)
- break; /* I'm pretty sure we're downsampling here */
-
- if (dsp->apply_crossfeed)
- dsp->apply_crossfeed(chunk, t2);
-
- if (dsp->eq_process)
- dsp->eq_process(chunk, t2);
-
-#ifdef HAVE_SW_TONE_CONTROLS
- if ((bass | treble) != 0)
- eq_filter(t2, &dsp->tone_filter, chunk,
- dsp->data.num_channels, FILTER_BISHELF_SHIFT);
-#endif
-
- if (dsp->channels_process)
- dsp->channels_process(chunk, t2);
-
- if (dsp->compressor_process)
- dsp->compressor_process(chunk, &dsp->data, t2);
-
- dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
-
- written += chunk;
- dst += chunk * sizeof (int16_t) * 2;
-
- /* yield at least once each tick */
- tick = current_tick;
- if (TIME_AFTER(tick, last_yield))
- {
- last_yield = tick;
- yield();
- }
- }
- }
-
-#if defined(CPU_COLDFIRE)
- /* set old macsr again */
- coldfire_set_macsr(old_macsr);
-#endif
- return written;
-}
-
-/* Given count number of input samples, calculate the maximum number of
- * samples of output data that would be generated (the calculation is not
- * entirely exact and rounds upwards to be on the safe side; during
- * resampling, the number of samples generated depends on the current state
- * of the resampler).
- */
-/* dsp_input_size MUST be called afterwards */
-int dsp_output_count(struct dsp_config *dsp, int count)
-{
-#ifdef HAVE_PITCHSCREEN
- if (dsp->tdspeed_active)
- count = tdspeed_est_output_size();
-#endif
- if (dsp->resample)
- {
- count = (int)(((unsigned long)count * NATIVE_FREQUENCY
- + (dsp->frequency - 1)) / dsp->frequency);
- }
-
- /* Now we have the resampled sample count which must not exceed
- * resample_buf_count to avoid resample buffer overflow. One
- * must call dsp_input_count() to get the correct input sample
- * count.
- */
- if (count > resample_buf_count)
- count = resample_buf_count;
-
- return count;
-}
-
-/* Given count output samples, calculate number of input samples
- * that would be consumed in order to fill the output buffer.
- */
-int dsp_input_count(struct dsp_config *dsp, int count)
-{
- /* count is now the number of resampled input samples. Convert to
- original input samples. */
- if (dsp->resample)
- {
- /* Use the real resampling delta =
- * dsp->frequency * 65536 / NATIVE_FREQUENCY, and
- * round towards zero to avoid buffer overflows. */
- count = (int)(((unsigned long)count *
- dsp->data.resample_data.delta) >> 16);
- }
-
-#ifdef HAVE_PITCHSCREEN
- if (dsp->tdspeed_active)
- count = tdspeed_est_input_size(count);
-#endif
-
- return count;
-}
-
-static void dsp_set_gain_var(long *var, long value)
-{
- *var = value;
- new_gain = true;
-}
-
-static void dsp_update_functions(struct dsp_config *dsp)
-{
- sample_input_new_format(dsp);
- sample_output_new_format(dsp);
- if (dsp == &AUDIO_DSP)
- dsp_set_crossfeed(crossfeed_enabled);
-}
-
-intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
-{
- switch (setting)
- {
- case DSP_MYDSP:
- switch (value)
- {
- case CODEC_IDX_AUDIO:
- return (intptr_t)&AUDIO_DSP;
- case CODEC_IDX_VOICE:
- return (intptr_t)&VOICE_DSP;
- default:
- return (intptr_t)NULL;
- }
-
- case DSP_SET_FREQUENCY:
- memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data));
- /* Fall through!!! */
- case DSP_SWITCH_FREQUENCY:
- dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value;
- /* Account for playback speed adjustment when setting dsp->frequency
- if we're called from the main audio thread. Voice UI thread should
- not need this feature.
- */
-#ifdef HAVE_PITCHSCREEN
- if (dsp == &AUDIO_DSP)
- dsp->frequency = pitch_ratio * dsp->codec_frequency / PITCH_SPEED_100;
- else
-#endif
- dsp->frequency = dsp->codec_frequency;
-
- resampler_new_delta(dsp);
-#ifdef HAVE_PITCHSCREEN
- tdspeed_setup(dsp);
-#endif
- break;
-
- case DSP_SET_SAMPLE_DEPTH:
- dsp->sample_depth = value;
-
- if (dsp->sample_depth <= NATIVE_DEPTH)
- {
- dsp->data.frac_bits = WORD_FRACBITS;
- dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */
- dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
- dsp->data.clip_min = -((1 << WORD_FRACBITS));
- }
- else
- {
- dsp->data.frac_bits = value;
- dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */
- dsp->data.clip_max = (1 << value) - 1;
- dsp->data.clip_min = -(1 << value);
- }
-
- dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
- sample_input_new_format(dsp);
- dither_init(dsp);
- break;
-
- case DSP_SET_STEREO_MODE:
- dsp->stereo_mode = value;
- dsp->data.num_channels = value == STEREO_MONO ? 1 : 2;
- dsp_update_functions(dsp);
-#ifdef HAVE_PITCHSCREEN
- tdspeed_setup(dsp);
-#endif
- break;
-
- case DSP_RESET:
- dsp->stereo_mode = STEREO_NONINTERLEAVED;
- dsp->data.num_channels = 2;
- dsp->sample_depth = NATIVE_DEPTH;
- dsp->data.frac_bits = WORD_FRACBITS;
- dsp->sample_bytes = sizeof (int16_t);
- dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
- dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
- dsp->data.clip_min = -((1 << WORD_FRACBITS));
- dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
-
- if (dsp == &AUDIO_DSP)
- {
- track_gain = 0;
- album_gain = 0;
- track_peak = 0;
- album_peak = 0;
- new_gain = true;
- }
-
- dsp_update_functions(dsp);
- resampler_new_delta(dsp);
-#ifdef HAVE_PITCHSCREEN
- tdspeed_setup(dsp);
-#endif
- if (dsp == &AUDIO_DSP)
- compressor_reset();
- break;
-
- case DSP_FLUSH:
- memset(&dsp->data.resample_data, 0,
- sizeof (dsp->data.resample_data));
- resampler_new_delta(dsp);
- dither_init(dsp);
-#ifdef HAVE_PITCHSCREEN
- tdspeed_setup(dsp);
-#endif
- if (dsp == &AUDIO_DSP)
- compressor_reset();
- break;
-
- case DSP_SET_TRACK_GAIN:
- if (dsp == &AUDIO_DSP)
- dsp_set_gain_var(&track_gain, value);
- break;
-
- case DSP_SET_ALBUM_GAIN:
- if (dsp == &AUDIO_DSP)
- dsp_set_gain_var(&album_gain, value);
- break;
-
- case DSP_SET_TRACK_PEAK:
- if (dsp == &AUDIO_DSP)
- dsp_set_gain_var(&track_peak, value);
- break;
-
- case DSP_SET_ALBUM_PEAK:
- if (dsp == &AUDIO_DSP)
- dsp_set_gain_var(&album_peak, value);
- break;
-
- default:
- return 0;
- }
-
- return 1;
-}
-
-int get_replaygain_mode(bool have_track_gain, bool have_album_gain)
-{
- int type;
-
- bool track = ((global_settings.replaygain_type == REPLAYGAIN_TRACK)
- || ((global_settings.replaygain_type == REPLAYGAIN_SHUFFLE)
- && global_settings.playlist_shuffle));
-
- type = (!track && have_album_gain) ? REPLAYGAIN_ALBUM
- : have_track_gain ? REPLAYGAIN_TRACK : -1;
-
- return type;
-}
-
-void dsp_set_replaygain(void)
-{
- long gain = 0;
-
- new_gain = false;
-
- if ((global_settings.replaygain_type != REPLAYGAIN_OFF) ||
- global_settings.replaygain_noclip)
- {
- bool track_mode = get_replaygain_mode(track_gain != 0,
- album_gain != 0) == REPLAYGAIN_TRACK;
- long peak = (track_mode || !album_peak) ? track_peak : album_peak;
-
- if (global_settings.replaygain_type != REPLAYGAIN_OFF)
- {
- gain = (track_mode || !album_gain) ? track_gain : album_gain;
-
- if (global_settings.replaygain_preamp)
- {
- long preamp = get_replaygain_int(
- global_settings.replaygain_preamp * 10);
-
- gain = (long) (((int64_t) gain * preamp) >> 24);
- }
- }
-
- if (gain == 0)
- {
- /* So that noclip can work even with no gain information. */
- gain = DEFAULT_GAIN;
- }
-
- if (global_settings.replaygain_noclip && (peak != 0)
- && ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN))
- {
- gain = (((int64_t) DEFAULT_GAIN << 24) / peak);
- }
-
- if (gain == DEFAULT_GAIN)
- {
- /* Nothing to do, disable processing. */
- gain = 0;
- }
- }
-
- /* Store in S7.24 format to simplify calculations. */
- replaygain = gain;
- set_gain(&AUDIO_DSP);
-}
-
-/** SET COMPRESSOR
- * Called by the menu system to configure the compressor process */
-void dsp_set_compressor(void)
-{
- /* enable/disable the compressor */
- AUDIO_DSP.compressor_process = compressor_update() ?
- compressor_process : NULL;
-}
diff --git a/apps/dsp.h b/apps/dsp.h
deleted file mode 100644
index 2a00f649f8..0000000000
--- a/apps/dsp.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Miika Pekkarinen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _DSP_H
-#define _DSP_H
-
-#include <stdlib.h>
-#include <stdbool.h>
-
-#define NATIVE_FREQUENCY 44100
-
-enum
-{
- STEREO_INTERLEAVED = 0,
- STEREO_NONINTERLEAVED,
- STEREO_MONO,
- STEREO_NUM_MODES,
-};
-
-enum
-{
- CODEC_IDX_AUDIO = 0,
- CODEC_IDX_VOICE,
-};
-
-enum
-{
- DSP_MYDSP = 1,
- DSP_SET_FREQUENCY,
- DSP_SWITCH_FREQUENCY,
- DSP_SET_SAMPLE_DEPTH,
- DSP_SET_STEREO_MODE,
- DSP_RESET,
- DSP_FLUSH,
- DSP_SET_TRACK_GAIN,
- DSP_SET_ALBUM_GAIN,
- DSP_SET_TRACK_PEAK,
- DSP_SET_ALBUM_PEAK,
- DSP_CROSSFEED
-};
-
-
-/****************************************************************************
- * NOTE: Any assembly routines that use these structures must be updated
- * if current data members are moved or changed.
- */
-struct resample_data
-{
- uint32_t delta; /* 00h */
- uint32_t phase; /* 04h */
- int32_t last_sample[2]; /* 08h */
- /* 10h */
-};
-
-/* This is for passing needed data to external dsp routines. If another
- * dsp parameter needs to be passed, add to the end of the structure
- * and remove from dsp_config.
- * If another function type becomes assembly/external and requires dsp
- * config info, add a pointer paramter of type "struct dsp_data *".
- * If removing something from other than the end, reserve the spot or
- * else update every implementation for every target.
- * Be sure to add the offset of the new member for easy viewing as well. :)
- * It is the first member of dsp_config and all members can be accessesed
- * through the main aggregate but this is intended to make a safe haven
- * for these items whereas the c part can be rearranged at will. dsp_data
- * could even moved within dsp_config without disurbing the order.
- */
-struct dsp_data
-{
- int output_scale; /* 00h */
- int num_channels; /* 04h */
- struct resample_data resample_data; /* 08h */
- int32_t clip_min; /* 18h */
- int32_t clip_max; /* 1ch */
- int32_t gain; /* 20h - Note that this is in S8.23 format. */
- int frac_bits; /* 24h */
- /* 28h */
-};
-
-struct dsp_config;
-
-int dsp_process(struct dsp_config *dsp, char *dest,
- const char *src[], int count);
-int dsp_input_count(struct dsp_config *dsp, int count);
-int dsp_output_count(struct dsp_config *dsp, int count);
-intptr_t dsp_configure(struct dsp_config *dsp, int setting,
- intptr_t value);
-int get_replaygain_mode(bool have_track_gain, bool have_album_gain);
-void dsp_set_replaygain(void);
-void dsp_set_crossfeed(bool enable);
-void dsp_set_crossfeed_direct_gain(int gain);
-void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain,
- long cutoff);
-void dsp_set_eq(bool enable);
-void dsp_set_eq_precut(int precut);
-void dsp_set_eq_coefs(int band);
-void dsp_dither_enable(bool enable);
-void dsp_timestretch_enable(bool enable);
-bool dsp_timestretch_available(void);
-void sound_set_pitch(int32_t r);
-int32_t sound_get_pitch(void);
-void dsp_set_timestretch(int32_t percent);
-int32_t dsp_get_timestretch(void);
-int dsp_callback(int msg, intptr_t param);
-void dsp_set_compressor(void);
-
-#endif
diff --git a/apps/dsp_arm.S b/apps/dsp_arm.S
deleted file mode 100644
index 7e360749a3..0000000000
--- a/apps/dsp_arm.S
+++ /dev/null
@@ -1,561 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006-2007 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
- #include "config.h"
-
-/****************************************************************************
- * void channels_process_sound_chan_mono(int count, int32_t *buf[])
- */
-
-#include "config.h"
-
- .section .icode, "ax", %progbits
- .align 2
- .global channels_process_sound_chan_mono
- .type channels_process_sound_chan_mono, %function
-channels_process_sound_chan_mono:
- @ input: r0 = count, r1 = buf
- stmfd sp!, { r4, lr } @
- @
- ldmia r1, { r1, r2 } @ r1 = buf[0], r2 = buf[1]
- subs r0, r0, #1 @ odd: end at 0; even: end at -1
- beq .mono_singlesample @ Zero? Only one sample!
- @
-.monoloop: @
- ldmia r1, { r3, r4 } @ r3, r4 = Li0, Li1
- ldmia r2, { r12, r14 } @ r12, r14 = Ri0, Ri1
- mov r3, r3, asr #1 @ Mo0 = Li0 / 2 + Ri0 / 2
- mov r4, r4, asr #1 @ Mo1 = Li1 / 2 + Ri1 / 2
- add r12, r3, r12, asr #1 @
- add r14, r4, r14, asr #1 @
- subs r0, r0, #2 @
- stmia r1!, { r12, r14 } @ store Mo0, Mo1
- stmia r2!, { r12, r14 } @ store Mo0, Mo1
- bgt .monoloop @
- @
- ldmpc cond=lt, regs=r4 @ if count was even, we're done
- @
-.mono_singlesample: @
- ldr r3, [r1] @ r3 = Ls
- ldr r12, [r2] @ r12 = Rs
- mov r3, r3, asr #1 @ Mo = Ls / 2 + Rs / 2
- add r12, r3, r12, asr #1 @
- str r12, [r1] @ store Mo
- str r12, [r2] @ store Mo
- @
- ldmpc regs=r4 @
- .size channels_process_sound_chan_mono, \
- .-channels_process_sound_chan_mono
-
-/****************************************************************************
- * void channels_process_sound_chan_custom(int count, int32_t *buf[])
- */
- .section .icode, "ax", %progbits
- .align 2
- .global channels_process_sound_chan_custom
- .type channels_process_sound_chan_custom, %function
-channels_process_sound_chan_custom:
- stmfd sp!, { r4-r10, lr }
-
- ldr r3, =dsp_sw_gain
- ldr r4, =dsp_sw_cross
-
- ldmia r1, { r1, r2 } @ r1 = buf[0], r2 = buf[1]
- ldr r3, [r3] @ r3 = dsp_sw_gain
- ldr r4, [r4] @ r4 = dsp_sw_cross
-
- subs r0, r0, #1
- beq .custom_single_sample @ Zero? Only one sample!
-
-.custom_loop:
- ldmia r1, { r5, r6 } @ r5 = Li0, r6 = Li1
- ldmia r2, { r7, r8 } @ r7 = Ri0, r8 = Ri1
-
- subs r0, r0, #2
-
- smull r9, r10, r5, r3 @ Lc0 = Li0*gain
- smull r12, r14, r7, r3 @ Rc0 = Ri0*gain
- smlal r9, r10, r7, r4 @ Lc0 += Ri0*cross
- smlal r12, r14, r5, r4 @ Rc0 += Li0*cross
-
- mov r9, r9, lsr #31 @ Convert to s0.31
- mov r12, r12, lsr #31
- orr r5, r9, r10, asl #1
- orr r7, r12, r14, asl #1
-
- smull r9, r10, r6, r3 @ Lc1 = Li1*gain
- smull r12, r14, r8, r3 @ Rc1 = Ri1*gain
- smlal r9, r10, r8, r4 @ Lc1 += Ri1*cross
- smlal r12, r14, r6, r4 @ Rc1 += Li1*cross
-
- mov r9, r9, lsr #31 @ Convert to s0.31
- mov r12, r12, lsr #31
- orr r6, r9, r10, asl #1
- orr r8, r12, r14, asl #1
-
- stmia r1!, { r5, r6 } @ Store Lc0, Lc1
- stmia r2!, { r7, r8 } @ Store Rc0, Rc1
-
- bgt .custom_loop
-
- ldmpc cond=lt, regs=r4-r10 @ < 0? even count
-
-.custom_single_sample:
- ldr r5, [r1] @ handle odd sample
- ldr r7, [r2]
-
- smull r9, r10, r5, r3 @ Lc0 = Li0*gain
- smull r12, r14, r7, r3 @ Rc0 = Ri0*gain
- smlal r9, r10, r7, r4 @ Lc0 += Ri0*cross
- smlal r12, r14, r5, r4 @ Rc0 += Li0*cross
-
- mov r9, r9, lsr #31 @ Convert to s0.31
- mov r12, r12, lsr #31
- orr r5, r9, r10, asl #1
- orr r7, r12, r14, asl #1
-
- str r5, [r1] @ Store Lc0
- str r7, [r2] @ Store Rc0
-
- ldmpc regs=r4-r10
- .size channels_process_sound_chan_custom, \
- .-channels_process_sound_chan_custom
-
-/****************************************************************************
- * void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
- */
- .section .icode, "ax", %progbits
- .align 2
- .global channels_process_sound_chan_karaoke
- .type channels_process_sound_chan_karaoke, %function
-channels_process_sound_chan_karaoke:
- @ input: r0 = count, r1 = buf
- stmfd sp!, { r4, lr } @
- @
- ldmia r1, { r1, r2 } @ r1 = buf[0], r2 = buf[1]
- subs r0, r0, #1 @ odd: end at 0; even: end at -1
- beq .karaoke_singlesample @ Zero? Only one sample!
- @
-.karaokeloop: @
- ldmia r1, { r3, r4 } @ r3, r4 = Li0, Li1
- ldmia r2, { r12, r14 } @ r12, r14 = Ri0, Ri1
- mov r3, r3, asr #1 @ Lo0 = Li0 / 2 - Ri0 / 2
- mov r4, r4, asr #1 @ Lo1 = Li1 / 2 - Ri1 / 2
- sub r3, r3, r12, asr #1 @
- sub r4, r4, r14, asr #1 @
- rsb r12, r3, #0 @ Ro0 = -Lk0 = Rs0 / 2 - Ls0 / 2
- rsb r14, r4, #0 @ Ro1 = -Lk1 = Ri1 / 2 - Li1 / 2
- subs r0, r0, #2 @
- stmia r1!, { r3, r4 } @ store Lo0, Lo1
- stmia r2!, { r12, r14 } @ store Ro0, Ro1
- bgt .karaokeloop @
- @
- ldmpc cond=lt, regs=r4 @ if count was even, we're done
- @
-.karaoke_singlesample: @
- ldr r3, [r1] @ r3 = Li
- ldr r12, [r2] @ r12 = Ri
- mov r3, r3, asr #1 @ Lk = Li / 2 - Ri /2
- sub r3, r3, r12, asr #1 @
- rsb r12, r3, #0 @ Rk = -Lo = Ri / 2 - Li / 2
- str r3, [r1] @ store Lo
- str r12, [r2] @ store Ro
- @
- ldmpc regs=r4 @
- .size channels_process_sound_chan_karaoke, \
- .-channels_process_sound_chan_karaoke
-
-#if ARM_ARCH < 6
-/****************************************************************************
- * void sample_output_mono(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- */
- .section .icode, "ax", %progbits
- .align 2
- .global sample_output_mono
- .type sample_output_mono, %function
-sample_output_mono:
- @ input: r0 = count, r1 = data, r2 = src, r3 = dst
- stmfd sp!, { r4-r6, lr }
-
- ldr r1, [r1] @ lr = data->output_scale
- ldr r2, [r2] @ r2 = src[0]
-
- mov r4, #1
- mov r4, r4, lsl r1 @ r4 = 1 << (scale-1)
- mov r4, r4, lsr #1
- mvn r14, #0x8000 @ r14 = 0xffff7fff, needed for
- @ clipping and masking
- subs r0, r0, #1 @
- beq .som_singlesample @ Zero? Only one sample!
-
-.somloop:
- ldmia r2!, { r5, r6 }
- add r5, r5, r4 @ r6 = (r6 + 1<<(scale-1)) >> scale
- mov r5, r5, asr r1
- mov r12, r5, asr #15
- teq r12, r12, asr #31
- eorne r5, r14, r5, asr #31 @ Clip (-32768...+32767)
- add r6, r6, r4
- mov r6, r6, asr r1 @ r7 = (r7 + 1<<(scale-1)) >> scale
- mov r12, r6, asr #15
- teq r12, r12, asr #31
- eorne r6, r14, r6, asr #31 @ Clip (-32768...+32767)
-
- and r5, r5, r14, lsr #16
- and r6, r6, r14, lsr #16
- orr r5, r5, r5, lsl #16 @ pack first 2 halfwords into 1 word
- orr r6, r6, r6, lsl #16 @ pack last 2 halfwords into 1 word
- stmia r3!, { r5, r6 }
-
- subs r0, r0, #2
- bgt .somloop
-
- ldmpc cond=lt, regs=r4-r6 @ even 'count'? return
-
-.som_singlesample:
- ldr r5, [r2] @ do odd sample
- add r5, r5, r4
- mov r5, r5, asr r1
- mov r12, r5, asr #15
- teq r12, r12, asr #31
- eorne r5, r14, r5, asr #31
-
- and r5, r5, r14, lsr #16 @ pack 2 halfwords into 1 word
- orr r5, r5, r5, lsl #16
- str r5, [r3]
-
- ldmpc regs=r4-r6
- .size sample_output_mono, .-sample_output_mono
-
-/****************************************************************************
- * void sample_output_stereo(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- */
- .section .icode, "ax", %progbits
- .align 2
- .global sample_output_stereo
- .type sample_output_stereo, %function
-sample_output_stereo:
- @ input: r0 = count, r1 = data, r2 = src, r3 = dst
- stmfd sp!, { r4-r9, lr }
-
- ldr r1, [r1] @ r1 = data->output_scale
- ldmia r2, { r2, r5 } @ r2 = src[0], r5 = src[1]
-
- mov r4, #1
- mov r4, r4, lsl r1 @ r4 = 1 << (scale-1)
- mov r4, r4, lsr #1 @
-
- mvn r14, #0x8000 @ r14 = 0xffff7fff, needed for
- @ clipping and masking
- subs r0, r0, #1 @
- beq .sos_singlesample @ Zero? Only one sample!
-
-.sosloop:
- ldmia r2!, { r6, r7 } @ 2 left
- ldmia r5!, { r8, r9 } @ 2 right
-
- add r6, r6, r4 @ r6 = (r6 + 1<<(scale-1)) >> scale
- mov r6, r6, asr r1
- mov r12, r6, asr #15
- teq r12, r12, asr #31
- eorne r6, r14, r6, asr #31 @ Clip (-32768...+32767)
- add r7, r7, r4
- mov r7, r7, asr r1 @ r7 = (r7 + 1<<(scale-1)) >> scale
- mov r12, r7, asr #15
- teq r12, r12, asr #31
- eorne r7, r14, r7, asr #31 @ Clip (-32768...+32767)
-
- add r8, r8, r4 @ r8 = (r8 + 1<<(scale-1)) >> scale
- mov r8, r8, asr r1
- mov r12, r8, asr #15
- teq r12, r12, asr #31
- eorne r8, r14, r8, asr #31 @ Clip (-32768...+32767)
- add r9, r9, r4 @ r9 = (r9 + 1<<(scale-1)) >> scale
- mov r9, r9, asr r1
- mov r12, r9, asr #15
- teq r12, r12, asr #31
- eorne r9, r14, r9, asr #31 @ Clip (-32768...+32767)
-
- and r6, r6, r14, lsr #16 @ pack first 2 halfwords into 1 word
- orr r8, r6, r8, asl #16
- and r7, r7, r14, lsr #16 @ pack last 2 halfwords into 1 word
- orr r9, r7, r9, asl #16
-
- stmia r3!, { r8, r9 }
-
- subs r0, r0, #2
- bgt .sosloop
-
- ldmpc cond=lt, regs=r4-r9 @ even 'count'? return
-
-.sos_singlesample:
- ldr r6, [r2] @ left odd sample
- ldr r8, [r5] @ right odd sample
-
- add r6, r6, r4 @ r6 = (r7 + 1<<(scale-1)) >> scale
- mov r6, r6, asr r1
- mov r12, r6, asr #15
- teq r12, r12, asr #31
- eorne r6, r14, r6, asr #31 @ Clip (-32768...+32767)
- add r8, r8, r4 @ r8 = (r8 + 1<<(scale-1)) >> scale
- mov r8, r8, asr r1
- mov r12, r8, asr #15
- teq r12, r12, asr #31
- eorne r8, r14, r8, asr #31 @ Clip (-32768...+32767)
-
- and r6, r6, r14, lsr #16 @ pack 2 halfwords into 1 word
- orr r8, r6, r8, asl #16
-
- str r8, [r3]
-
- ldmpc regs=r4-r9
- .size sample_output_stereo, .-sample_output_stereo
-#endif /* ARM_ARCH < 6 */
-
-/****************************************************************************
- * void apply_crossfeed(int count, int32_t* src[])
- */
- .section .text
- .global apply_crossfeed
-apply_crossfeed:
- @ unfortunately, we ended up in a bit of a register squeeze here, and need
- @ to keep the count on the stack :/
- stmdb sp!, { r4-r11, lr } @ stack modified regs
- ldmia r1, { r2-r3 } @ r2 = src[0], r3 = src[1]
-
- ldr r1, =crossfeed_data
- ldmia r1!, { r4-r11 } @ load direct gain and filter data
- mov r12, r0 @ better to ldm delay + count later
- add r0, r1, #13*4*2 @ calculate end of delay
- stmdb sp!, { r0, r12 } @ stack end of delay adr and count
- ldr r0, [r1, #13*4*2] @ fetch current delay line address
-
- /* Register usage in loop:
- * r0 = &delay[index][0], r1 = accumulator high, r2 = src[0], r3 = src[1],
- * r4 = direct gain, r5-r7 = b0, b1, a1 (filter coefs),
- * r8-r11 = filter history, r12 = temp, r14 = accumulator low
- */
-.cfloop:
- smull r14, r1, r6, r8 @ acc = b1*dr[n - 1]
- smlal r14, r1, r7, r9 @ acc += a1*y_l[n - 1]
- ldr r8, [r0, #4] @ r8 = dr[n]
- smlal r14, r1, r5, r8 @ acc += b0*dr[n]
- mov r9, r1, lsl #1 @ fix format for filter history
- ldr r12, [r2] @ load left input
- smlal r14, r1, r4, r12 @ acc += gain*x_l[n]
- mov r1, r1, lsl #1 @ fix format
- str r1, [r2], #4 @ save result
-
- smull r14, r1, r6, r10 @ acc = b1*dl[n - 1]
- smlal r14, r1, r7, r11 @ acc += a1*y_r[n - 1]
- ldr r10, [r0] @ r10 = dl[n]
- str r12, [r0], #4 @ save left input to delay line
- smlal r14, r1, r5, r10 @ acc += b0*dl[n]
- mov r11, r1, lsl #1 @ fix format for filter history
- ldr r12, [r3] @ load right input
- smlal r14, r1, r4, r12 @ acc += gain*x_r[n]
- str r12, [r0], #4 @ save right input to delay line
- mov r1, r1, lsl #1 @ fix format
- ldmia sp, { r12, r14 } @ fetch delay line end addr and count from stack
- str r1, [r3], #4 @ save result
-
- cmp r0, r12 @ need to wrap to start of delay?
- subeq r0, r0, #13*4*2 @ wrap back delay line ptr to start
-
- subs r14, r14, #1 @ are we finished?
- strne r14, [sp, #4] @ nope, save count back to stack
- bne .cfloop
-
- @ save data back to struct
- ldr r12, =crossfeed_data + 4*4
- stmia r12, { r8-r11 } @ save filter history
- str r0, [r12, #30*4] @ save delay line index
- add sp, sp, #8 @ remove temp variables from stack
- ldmpc regs=r4-r11
- .size apply_crossfeed, .-apply_crossfeed
-
-/****************************************************************************
- * int dsp_downsample(int count, struct dsp_data *data,
- * in32_t *src[], int32_t *dst[])
- */
- .section .text
- .global dsp_downsample
-dsp_downsample:
- stmdb sp!, { r4-r11, lr } @ stack modified regs
- ldmib r1, { r5-r6 } @ r5 = num_channels,r6 = resample_data.delta
- sub r5, r5, #1 @ pre-decrement num_channels for use
- add r4, r1, #12 @ r4 = &resample_data.phase
- mov r12, #0xff
- orr r12, r12, #0xff00 @ r12 = 0xffff
-.dschannel_loop:
- ldr r1, [r4] @ r1 = resample_data.phase
- ldr r7, [r2, r5, lsl #2] @ r7 = s = src[ch - 1]
- ldr r8, [r3, r5, lsl #2] @ r8 = d = dst[ch - 1]
- add r9, r4, #4 @ r9 = &last_sample[0]
- ldr r10, [r9, r5, lsl #2] @ r10 = last_sample[ch - 1]
- sub r11, r0, #1
- ldr r14, [r7, r11, lsl #2] @ load last sample in s[] ...
- str r14, [r9, r5, lsl #2] @ and write as next frame's last_sample
- movs r9, r1, lsr #16 @ r9 = pos = phase >> 16
- ldreq r11, [r7] @ if pos = 0, load src[0] and jump into loop
- beq .dsuse_last_start
- cmp r9, r0 @ if pos >= count, we're already done
- bge .dsloop_skip
-
- @ Register usage in loop:
- @ r0 = count, r1 = phase, r4 = &resample_data.phase, r5 = cur_channel,
- @ r6 = delta, r7 = s, r8 = d, r9 = pos, r10 = s[pos - 1], r11 = s[pos]
-.dsloop:
- add r9, r7, r9, lsl #2 @ r9 = &s[pos]
- ldmda r9, { r10, r11 } @ r10 = s[pos - 1], r11 = s[pos]
-.dsuse_last_start:
- sub r11, r11, r10 @ r11 = diff = s[pos] - s[pos - 1]
- @ keep frac in lower bits to take advantage of multiplier early termination
- and r9, r1, r12 @ frac = phase & 0xffff
- smull r9, r14, r11, r9
- add r1, r1, r6 @ phase += delta
- add r10, r10, r9, lsr #16 @ r10 = out = s[pos - 1] + frac*diff
- add r10, r10, r14, lsl #16
- str r10, [r8], #4 @ *d++ = out
- mov r9, r1, lsr #16 @ pos = phase >> 16
- cmp r9, r0 @ pos < count?
- blt .dsloop @ yup, do more samples
-.dsloop_skip:
- subs r5, r5, #1
- bpl .dschannel_loop @ if (--ch) >= 0, do another channel
- sub r1, r1, r0, lsl #16 @ wrap phase back to start
- str r1, [r4] @ store back
- ldr r1, [r3] @ r1 = &dst[0]
- sub r8, r8, r1 @ dst - &dst[0]
- mov r0, r8, lsr #2 @ convert bytes->samples
- ldmpc regs=r4-r11 @ ... and we're out
- .size dsp_downsample, .-dsp_downsample
-
-/****************************************************************************
- * int dsp_upsample(int count, struct dsp_data *dsp,
- * in32_t *src[], int32_t *dst[])
- */
- .section .text
- .global dsp_upsample
-dsp_upsample:
- stmfd sp!, { r4-r11, lr } @ stack modified regs
- ldmib r1, { r5-r6 } @ r5 = num_channels,r6 = resample_data.delta
- sub r5, r5, #1 @ pre-decrement num_channels for use
- add r4, r1, #12 @ r4 = &resample_data.phase
- mov r6, r6, lsl #16 @ we'll use carry to detect pos increments
- stmfd sp!, { r0, r4 } @ stack count and &resample_data.phase
-.uschannel_loop:
- ldr r12, [r4] @ r12 = resample_data.phase
- ldr r7, [r2, r5, lsl #2] @ r7 = s = src[ch - 1]
- ldr r8, [r3, r5, lsl #2] @ r8 = d = dst[ch - 1]
- add r9, r4, #4 @ r9 = &last_sample[0]
- mov r1, r12, lsl #16 @ we'll use carry to detect pos increments
- sub r11, r0, #1
- ldr r14, [r7, r11, lsl #2] @ load last sample in s[] ...
- ldr r10, [r9, r5, lsl #2] @ r10 = last_sample[ch - 1]
- str r14, [r9, r5, lsl #2] @ and write as next frame's last_sample
- movs r14, r12, lsr #16 @ pos = resample_data.phase >> 16
- beq .usstart_0 @ pos = 0
- cmp r14, r0 @ if pos >= count, we're already done
- bge .usloop_skip
- add r7, r7, r14, lsl #2 @ r7 = &s[pos]
- ldr r10, [r7, #-4] @ r11 = s[pos - 1]
- b .usstart_0
-
- @ Register usage in loop:
- @ r0 = count, r1 = phase, r4 = &resample_data.phase, r5 = cur_channel,
- @ r6 = delta, r7 = s, r8 = d, r9 = diff, r10 = s[pos - 1], r11 = s[pos]
-.usloop_1:
- mov r10, r11 @ r10 = previous sample
-.usstart_0:
- ldr r11, [r7], #4 @ r11 = next sample
- mov r4, r1, lsr #16 @ r4 = frac = phase >> 16
- sub r9, r11, r10 @ r9 = diff = s[pos] - s[pos - 1]
-.usloop_0:
- smull r12, r14, r4, r9
- adds r1, r1, r6 @ phase += delta << 16
- mov r4, r1, lsr #16 @ r4 = frac = phase >> 16
- add r14, r10, r14, lsl #16
- add r14, r14, r12, lsr #16 @ r14 = out = s[pos - 1] + frac*diff
- str r14, [r8], #4 @ *d++ = out
- bcc .usloop_0 @ if carry is set, pos is incremented
- subs r0, r0, #1 @ if count > 0, do another sample
- bgt .usloop_1
-.usloop_skip:
- subs r5, r5, #1
- ldmfd sp, { r0, r4 } @ reload count and &resample_data.phase
- bpl .uschannel_loop @ if (--ch) >= 0, do another channel
- mov r1, r1, lsr #16 @ wrap phase back to start of next frame
- ldr r2, [r3] @ r1 = &dst[0]
- str r1, [r4] @ store phase
- sub r8, r8, r2 @ dst - &dst[0]
- mov r0, r8, lsr #2 @ convert bytes->samples
- add sp, sp, #8 @ adjust stack for temp variables
- ldmpc regs=r4-r11 @ ... and we're out
- .size dsp_upsample, .-dsp_upsample
-
-/****************************************************************************
- * void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
- */
- .section .icode, "ax", %progbits
- .align 2
- .global dsp_apply_gain
- .type dsp_apply_gain, %function
-dsp_apply_gain:
- @ input: r0 = count, r1 = data, r2 = buf[]
- stmfd sp!, { r4-r8, lr }
-
- ldr r3, [r1, #4] @ r3 = data->num_channels
- ldr r4, [r1, #32] @ r5 = data->gain
-
-.dag_outerloop:
- ldr r1, [r2], #4 @ r1 = buf[0] and increment index of buf[]
- subs r12, r0, #1 @ r12 = r0 = count - 1
- beq .dag_singlesample @ Zero? Only one sample!
-
-.dag_innerloop:
- ldmia r1, { r5, r6 } @ load r5, r6 from r1
- smull r7, r8, r5, r4 @ r7 = FRACMUL_SHL(r5, r4, 8)
- smull r14, r5, r6, r4 @ r14 = FRACMUL_SHL(r6, r4, 8)
- subs r12, r12, #2
- mov r7, r7, lsr #23
- mov r14, r14, lsr #23
- orr r7, r7, r8, asl #9
- orr r14, r14, r5, asl #9
- stmia r1!, { r7, r14 } @ save r7, r14 to [r1] and increment r1
- bgt .dag_innerloop @ end of inner loop
-
- blt .dag_evencount @ < 0? even count
-
-.dag_singlesample:
- ldr r5, [r1] @ handle odd sample
- smull r7, r8, r5, r4 @ r7 = FRACMUL_SHL(r5, r4, 8)
- mov r7, r7, lsr #23
- orr r7, r7, r8, asl #9
- str r7, [r1]
-
-.dag_evencount:
- subs r3, r3, #1
- bgt .dag_outerloop @ end of outer loop
-
- ldmpc regs=r4-r8
- .size dsp_apply_gain, .-dsp_apply_gain
diff --git a/apps/dsp_arm_v6.S b/apps/dsp_arm_v6.S
deleted file mode 100644
index 39949498ea..0000000000
--- a/apps/dsp_arm_v6.S
+++ /dev/null
@@ -1,127 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2010 Michael Sevakis
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-/****************************************************************************
- * void sample_output_mono(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- */
- .section .text, "ax", %progbits
- .align 2
- .global sample_output_mono
- .type sample_output_mono, %function
-sample_output_mono:
- @ input: r0 = count, r1 = data, r2 = src, r3 = dst
- stmfd sp!, { r4, lr } @
- @
- ldr r1, [r1] @ r1 = data->output_scale
- ldr r2, [r2] @ r2 = src[0]
- @
- mov r4, #1 @ r4 = 1 << (scale - 1)
- mov r4, r4, lsl r1 @
- subs r0, r0, #1 @ odd: end at 0; even: end at -1
- mov r4, r4, lsr #1 @
- beq 2f @ Zero? Only one sample!
- @
-1: @
- ldmia r2!, { r12, r14 } @ load Mi0, Mi1
- qadd r12, r12, r4 @ round, scale, saturate and
- qadd r14, r14, r4 @ pack Mi0 to So0, Mi1 to So1
- mov r12, r12, asr r1 @
- mov r14, r14, asr r1 @
- ssat r12, #16, r12 @
- ssat r14, #16, r14 @
- pkhbt r12, r12, r12, asl #16 @
- pkhbt r14, r14, r14, asl #16 @
- subs r0, r0, #2 @
- stmia r3!, { r12, r14 } @ store So0, So1
- bgt 1b @
- @
- ldmltfd sp!, { r4, pc } @ if count was even, we're done
- @
-2: @
- ldr r12, [r2] @ round, scale, saturate
- qadd r12, r12, r4 @ and pack Mi to So
- mov r12, r12, asr r1 @
- ssat r12, #16, r12 @
- pkhbt r12, r12, r12, asl #16 @
- str r12, [r3] @ store So
- @
- ldmfd sp!, { r4, pc } @
- .size sample_output_mono, .-sample_output_mono
-
-/****************************************************************************
- * void sample_output_stereo(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- */
- .section .text, "ax", %progbits
- .align 2
- .global sample_output_stereo
- .type sample_output_stereo, %function
-sample_output_stereo:
- @ input: r0 = count, r1 = data, r2 = src, r3 = dst
- stmfd sp!, { r4-r7, lr } @
- @
- ldr r1, [r1] @ r1 = data->output_scale
- ldmia r2, { r2, r4 } @ r2 = src[0], r4 = src[1]
- @
- mov r5, #1 @ r5 = 1 << (scale - 1)
- mov r5, r5, lsl r1 @
- subs r0, r0, #1 @ odd: end at 0; even: end at -1
- mov r5, r5, lsr #1 @
- beq 2f @ Zero? Only one sample!
- @
-1: @
- ldmia r2!, { r6, r7 } @ r6, r7 = Li0, Li1
- ldmia r4!, { r12, r14 } @ r12, r14 = Ri0, Ri1
- qadd r6, r6, r5 @ round, scale, saturate and pack
- qadd r7, r7, r5 @ Li0+Ri0 to So0, Li1+Ri1 to So1
- qadd r12, r12, r5 @
- qadd r14, r14, r5 @
- mov r6, r6, asr r1 @
- mov r7, r7, asr r1 @
- mov r12, r12, asr r1 @
- mov r14, r14, asr r1 @
- ssat r6, #16, r6 @
- ssat r12, #16, r12 @
- ssat r7, #16, r7 @
- ssat r14, #16, r14 @
- pkhbt r6, r6, r12, asl #16 @
- pkhbt r7, r7, r14, asl #16 @
- subs r0, r0, #2 @
- stmia r3!, { r6, r7 } @ store So0, So1
- bgt 1b @
- @
- ldmltfd sp!, { r4-r7, pc } @ if count was even, we're done
- @
-2: @
- ldr r6, [r2] @ r6 = Li
- ldr r12, [r4] @ r12 = Ri
- qadd r6, r6, r5 @ round, scale, saturate
- qadd r12, r12, r5 @ and pack Li+Ri to So
- mov r6, r6, asr r1 @
- mov r12, r12, asr r1 @
- ssat r6, #16, r6 @
- ssat r12, #16, r12 @
- pkhbt r6, r6, r12, asl #16 @
- str r6, [r3] @ store So
- @
- ldmfd sp!, { r4-r7, pc } @
- .size sample_output_stereo, .-sample_output_stereo
diff --git a/apps/dsp_asm.h b/apps/dsp_asm.h
deleted file mode 100644
index 7bf18370a3..0000000000
--- a/apps/dsp_asm.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <config.h>
-
-#ifndef _DSP_ASM_H
-#define _DSP_ASM_H
-
-/* Set the appropriate #defines based on CPU or whatever matters */
-#if defined(CPU_ARM)
-#define DSP_HAVE_ASM_APPLY_GAIN
-#define DSP_HAVE_ASM_RESAMPLING
-#define DSP_HAVE_ASM_CROSSFEED
-#define DSP_HAVE_ASM_SOUND_CHAN_MONO
-#define DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
-#define DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
-#define DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
-#define DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
-#elif defined (CPU_COLDFIRE)
-#define DSP_HAVE_ASM_APPLY_GAIN
-#define DSP_HAVE_ASM_RESAMPLING
-#define DSP_HAVE_ASM_CROSSFEED
-#define DSP_HAVE_ASM_SOUND_CHAN_MONO
-#define DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
-#define DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
-#define DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
-#define DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
-#endif /* CPU_COLDFIRE */
-
-/* Declare prototypes based upon what's #defined above */
-#ifdef DSP_HAVE_ASM_CROSSFEED
-void apply_crossfeed(int count, int32_t *buf[]);
-#endif
-
-#ifdef DSP_HAVE_ASM_APPLY_GAIN
-void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[]);
-#endif /* DSP_HAVE_ASM_APPLY_GAIN* */
-
-#ifdef DSP_HAVE_ASM_RESAMPLING
-int dsp_upsample(int count, struct dsp_data *data,
- const int32_t *src[], int32_t *dst[]);
-int dsp_downsample(int count, struct dsp_data *data,
- const int32_t *src[], int32_t *dst[]);
-#endif /* DSP_HAVE_ASM_RESAMPLING */
-
-#ifdef DSP_HAVE_ASM_SOUND_CHAN_MONO
-void channels_process_sound_chan_mono(int count, int32_t *buf[]);
-#endif
-
-#ifdef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
-void channels_process_sound_chan_custom(int count, int32_t *buf[]);
-#endif
-
-#ifdef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
-void channels_process_sound_chan_karaoke(int count, int32_t *buf[]);
-#endif
-
-#ifdef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
-void sample_output_stereo(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst);
-#endif
-
-#ifdef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
-void sample_output_mono(int count, struct dsp_data *data,
- const int32_t *src[], int16_t *dst);
-#endif
-
-#endif /* _DSP_ASM_H */
diff --git a/apps/dsp_cf.S b/apps/dsp_cf.S
deleted file mode 100644
index cda811a7d5..0000000000
--- a/apps/dsp_cf.S
+++ /dev/null
@@ -1,611 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006 Thom Johansen
- * Portions Copyright (C) 2007 Michael Sevakis
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-/****************************************************************************
- * void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
- */
- .section .text
- .align 2
- .global dsp_apply_gain
-dsp_apply_gain:
- lea.l -20(%sp), %sp | save registers
- movem.l %d2-%d4/%a2-%a3, (%sp) |
- movem.l 28(%sp), %a0-%a1 | %a0 = data,
- | %a1 = buf
- move.l 4(%a0), %d1 | %d1 = data->num_channels
- move.l 32(%a0), %a0 | %a0 = data->gain (in s8.23)
-10: | channel loop |
- move.l 24(%sp), %d0 | %d0 = count
- move.l -4(%a1, %d1.l*4), %a2 | %a2 = s = buf[ch-1]
- move.l %a2, %a3 | %a3 = d = s
- move.l (%a2)+, %d2 | %d2 = *s++,
- mac.l %a0, %d2, (%a2)+, %d2, %acc0 | %acc0 = S(n)*gain, load S(n+1)
- subq.l #1, %d0 | --count > 0 ? : effectively n++
- ble.b 30f | loop done | no? finish up
-20: | loop |
- move.l %accext01, %d4 | fetch S(n-1)[7:0]
- movclr.l %acc0, %d3 | fetch S(n-1)[40:8] in %d5[31:0]
- asl.l #8, %d3 | *s++ = (S(n-1)[40:8] << 8) | S(n-1)[7:0]
- mac.l %a0, %d2, (%a2)+, %d2, %acc0 | %acc0 = S(n)*gain, load S(n+1)
- move.b %d4, %d3 |
- move.l %d3, (%a3)+ |
- subq.l #1, %d0 | --count > 0 ? : effectively n++
- bgt.b 20b | loop | yes? do more samples
-30: | loop done |
- move.l %accext01, %d4 | fetch S(n-1)[7:0]
- movclr.l %acc0, %d3 | fetch S(n-1)[40:8] in %d5[31:0]
- asl.l #8, %d3 | *s = (S(n-1)[40:8] << 8) | S(n-1)[7:0]
- move.b %d4, %d3 |
- move.l %d3, (%a3) |
- subq.l #1, %d1 | next channel
- bgt.b 10b | channel loop |
- movem.l (%sp), %d2-%d4/%a2-%a3 | restore registers
- lea.l 20(%sp), %sp | cleanup stack
- rts |
- .size dsp_apply_gain,.-dsp_apply_gain
-
-/****************************************************************************
- * void apply_crossfeed(int count, int32_t *buf[])
- */
- .section .text
- .align 2
- .global apply_crossfeed
-apply_crossfeed:
- lea.l -44(%sp), %sp |
- movem.l %d2-%d7/%a2-%a6, (%sp) | save all regs
- movem.l 48(%sp), %d7/%a4 | %d7 = count, %a4 = src
- movem.l (%a4), %a4-%a5 | %a4 = src[0], %a5 = src[1]
- lea.l crossfeed_data, %a1 | %a1 = &crossfeed_data
- move.l (%a1)+, %d6 | %d6 = direct gain
- movem.l 12(%a1), %d0-%d3 | fetch filter history samples
- move.l 132(%a1), %a0 | fetch delay line address
- movem.l (%a1), %a1-%a3 | load filter coefs
- lea.l crossfeed_data+136, %a6 | %a6 = delay line wrap limit
- bra.b 20f | loop start | go to loop start point
- /* Register usage in loop:
- * %a0 = delay_p, %a1..%a3 = b0, b1, a1 (filter coefs),
- * %a4 = buf[0], %a5 = buf[1],
- * %a6 = delay line pointer wrap limit,
- * %d0..%d3 = history
- * %d4..%d5 = temp.
- * %d6 = direct gain,
- * %d7 = count
- */
-10: | loop |
- movclr.l %acc0, %d4 | write outputs
- move.l %d4, (%a4)+ | .
- movclr.l %acc1, %d5 | .
- move.l %d5, (%a5)+ | .
-20: | loop start |
- mac.l %a2, %d0, (%a0)+, %d0, %acc0 | %acc0 = b1*dl[n - 1], %d0 = dl[n]
- mac.l %a1, %d0 , %acc0 | %acc0 += b0*dl[n]
- mac.l %a3, %d1, (%a5), %d5, %acc0 | %acc0 += a1*y_r[n - 1], load R
- mac.l %a2, %d2, (%a0)+, %d2, %acc1 | %acc1 = b1*dr[n - 1], %d2 = dr[n]
- mac.l %a1, %d2 , %acc1 | %acc1 += b0*dr[n]
- mac.l %a3, %d3, (%a4), %d4, %acc1 | %acc1 += a1*y_l[n - 1], load L
- movem.l %d4-%d5, -8(%a0) | save left & right inputs to delay line
- move.l %acc0, %d3 | get filtered delayed left sample (y_l[n])
- move.l %acc1, %d1 | get filtered delayed right sample (y_r[n])
- mac.l %d6, %d4, %acc0 | %acc0 += gain*x_l[n]
- mac.l %d6, %d5, %acc1 | %acc1 += gain*x_r[n]
- cmp.l %a6, %a0 | wrap %a0 if passed end
- bhs.b 30f | wrap buffer |
- .word 0x51fb | tpf.l | trap the buffer wrap
-30: | wrap buffer | ...fwd taken branches more costly
- lea.l -104(%a0), %a0 | wrap it up
- subq.l #1, %d7 | --count > 0 ?
- bgt.b 10b | loop | yes? do more
- movclr.l %acc0, %d4 | write last outputs
- move.l %d4, (%a4) | .
- movclr.l %acc1, %d5 | .
- move.l %d5, (%a5) | .
- lea.l crossfeed_data+16, %a1 | save data back to struct
- movem.l %d0-%d3, (%a1) | ...history
- move.l %a0, 120(%a1) | ...delay_p
- movem.l (%sp), %d2-%d7/%a2-%a6 | restore all regs
- lea.l 44(%sp), %sp |
- rts |
- .size apply_crossfeed,.-apply_crossfeed
-
-/****************************************************************************
- * int dsp_downsample(int count, struct dsp_data *data,
- * in32_t *src[], int32_t *dst[])
- */
- .section .text
- .align 2
- .global dsp_downsample
-dsp_downsample:
- lea.l -40(%sp), %sp | save non-clobberables
- movem.l %d2-%d7/%a2-%a5, (%sp) |
- movem.l 44(%sp), %d2/%a0-%a2 | %d2 = count
- | %a0 = data
- | %a1 = src
- | %a2 = dst
- movem.l 4(%a0), %d3-%d4 | %d3 = ch = data->num_channels
- | %d4 = delta = data->resample_data.delta
- moveq.l #16, %d7 | %d7 = shift
-10: | channel loop |
- move.l 12(%a0), %d5 | %d5 = phase = data->resample_data.phase
- move.l -4(%a1, %d3.l*4), %a3 | %a3 = s = src[ch-1]
- move.l -4(%a2, %d3.l*4), %a4 | %a4 = d = dst[ch-1]
- lea.l 12(%a0, %d3.l*4), %a5 | %a5 = &data->resample_data.ast_sample[ch-1]
- move.l (%a5), %d0 | %d0 = last = data->resample_data.last_sample[ch-1]
- move.l -4(%a3, %d2.l*4), (%a5) | data->resample_data.last_sample[ch-1] = s[count-1]
- move.l %d5, %d6 | %d6 = pos = phase >> 16
- lsr.l %d7, %d6 |
- cmp.l %d2, %d6 | past end of samples?
- bge.b 40f | skip resample loop| yes? skip loop
- tst.l %d6 | need last sample of prev. frame?
- bne.b 20f | resample loop | no? start main loop
- move.l (%a3, %d6.l*4), %d1 | %d1 = s[pos]
- bra.b 30f | resample start last | start with last (last in %d0)
-20: | resample loop |
- lea.l -4(%a3, %d6.l*4), %a5 | load s[pos-1] and s[pos]
- movem.l (%a5), %d0-%d1 |
-30: | resample start last |
- sub.l %d0, %d1 | %d1 = diff = s[pos] - s[pos-1]
- move.l %d0, %acc0 | %acc0 = previous sample
- move.l %d5, %d0 | frac = (phase << 16) >> 1
- lsl.l %d7, %d0 |
- lsr.l #1, %d0 |
- mac.l %d0, %d1, %acc0 | %acc0 += frac * diff
- add.l %d4, %d5 | phase += delta
- move.l %d5, %d6 | pos = phase >> 16
- lsr.l %d7, %d6 |
- movclr.l %acc0, %d0 |
- move.l %d0, (%a4)+ | *d++ = %d0
- cmp.l %d2, %d6 | pos < count?
- blt.b 20b | resample loop | yes? continue resampling
-40: | skip resample loop |
- subq.l #1, %d3 | ch > 0?
- bgt.b 10b | channel loop | yes? process next channel
- lsl.l %d7, %d2 | wrap phase to start of next frame
- sub.l %d2, %d5 | data->resample_data.phase =
- move.l %d5, 12(%a0) | ... phase - (count << 16)
- move.l %a4, %d0 | return d - d[0]
- sub.l (%a2), %d0 |
- asr.l #2, %d0 | convert bytes->samples
- movem.l (%sp), %d2-%d7/%a2-%a5 | restore non-clobberables
- lea.l 40(%sp), %sp | cleanup stack
- rts | buh-bye
- .size dsp_downsample,.-dsp_downsample
-
-/****************************************************************************
- * int dsp_upsample(int count, struct dsp_data *dsp,
- * const int32_t *src[], int32_t *dst[])
- */
- .section .text
- .align 2
- .global dsp_upsample
-dsp_upsample:
- lea.l -40(%sp), %sp | save non-clobberables
- movem.l %d2-%d7/%a2-%a5, (%sp) |
- movem.l 44(%sp), %d2/%a0-%a2 | %d2 = count
- | %a0 = data
- | %a1 = src
- | %a2 = dst
- movem.l 4(%a0), %d3-%d4 | %d3 = ch = channels
- | %d4 = delta = data->resample_data.delta
- swap %d4 | swap delta to high word to use...
- | ...carries to increment position
-10: | channel loop |
- move.l 12(%a0), %d5 | %d5 = phase = data->resample_data.phase
- move.l -4(%a1, %d3.l*4), %a3 | %a3 = s = src[ch-1]
- lea.l 12(%a0, %d3.l*4), %a4 | %a4 = &data->resample_data.last_sample[ch-1]
- lea.l -4(%a3, %d2.l*4), %a5 | %a5 = src_end = &src[count-1]
- move.l (%a4), %d0 | %d0 = last = data->resample_data.last_sample[ch-1]
- move.l (%a5), (%a4) | data->resample_data.last_sample[ch-1] = s[count-1]
- move.l -4(%a2, %d3.l*4), %a4 | %a4 = d = dst[ch-1]
- move.l (%a3)+, %d1 | fetch first sample - might throw this...
- | ...away later but we'll be preincremented
- move.l %d1, %d6 | save sample value
- sub.l %d0, %d1 | %d1 = diff = s[0] - last
- swap %d5 | swap phase to high word to use
- | carries to increment position
- move.l %d5, %d7 | %d7 = pos = phase >> 16
- clr.w %d5 |
- eor.l %d5, %d7 | pos == 0?
- beq.b 40f | loop start | yes? start loop
- cmp.l %d2, %d7 | past end of samples?
- bge.b 50f | skip resample loop| yes? go to next channel and collect info
- lea.l (%a3, %d7.l*4), %a3 | %a3 = s = &s[pos+1]
- movem.l -8(%a3), %d0-%d1 | %d0 = s[pos-1], %d1 = s[pos]
- move.l %d1, %d6 | save sample value
- sub.l %d0, %d1 | %d1 = diff = s[pos] - s[pos-1]
- bra.b 40f | loop start |
-20: | next sample loop |
- move.l %d6, %d0 | move previous sample to %d0
- move.l (%a3)+, %d1 | fetch next sample
- move.l %d1, %d6 | save sample value
- sub.l %d0, %d1 | %d1 = diff = s[pos] - s[pos-1]
-30: | same sample loop |
- movclr.l %acc0, %d7 | %d7 = result
- move.l %d7, (%a4)+ | *d++ = %d7
-40: | loop start |
- lsr.l #1, %d5 | make phase into frac
- move.l %d0, %acc0 | %acc0 = s[pos-1]
- mac.l %d1, %d5, %acc0 | %acc0 = diff * frac
- lsl.l #1, %d5 | restore frac to phase
- add.l %d4, %d5 | phase += delta
- bcc.b 30b | same sample loop | load next values?
- cmp.l %a5, %a3 | src <= src_end?
- bls.b 20b | next sample loop | yes? continue resampling
- movclr.l %acc0, %d7 | %d7 = result
- move.l %d7, (%a4)+ | *d++ = %d7
-50: | skip resample loop |
- subq.l #1, %d3 | ch > 0?
- bgt.b 10b | channel loop | yes? process next channel
- swap %d5 | wrap phase to start of next frame
- move.l %d5, 12(%a0) | ...and save in data->resample_data.phase
- move.l %a4, %d0 | return d - d[0]
- sub.l (%a2), %d0 |
- movem.l (%sp), %d2-%d7/%a2-%a5 | restore non-clobberables
- asr.l #2, %d0 | convert bytes->samples
- lea.l 40(%sp), %sp | cleanup stack
- rts | buh-bye
- .size dsp_upsample,.-dsp_upsample
-
-/****************************************************************************
- * void channels_process_sound_chan_mono(int count, int32_t *buf[])
- *
- * Mix left and right channels 50/50 into a center channel.
- */
- .section .text
- .align 2
- .global channels_process_sound_chan_mono
-channels_process_sound_chan_mono:
- movem.l 4(%sp), %d0/%a0 | %d0 = count, %a0 = buf
- lea.l -20(%sp), %sp | save registers
- movem.l %d2-%d4/%a2-%a3, (%sp) |
- movem.l (%a0), %a0-%a1 | get channel pointers
- move.l %a0, %a2 | use separate dst pointers since read
- move.l %a1, %a3 | pointers run one ahead of write
- move.l #0x40000000, %d3 | %d3 = 0.5
- move.l (%a0)+, %d1 | prime the input registers
- move.l (%a1)+, %d2 |
- mac.l %d1, %d3, (%a0)+, %d1, %acc0 |
- mac.l %d2, %d3, (%a1)+, %d2, %acc0 |
- subq.l #1, %d0 |
- ble.s 20f | loop done |
-10: | loop |
- movclr.l %acc0, %d4 | L = R = l/2 + r/2
- mac.l %d1, %d3, (%a0)+, %d1, %acc0 |
- mac.l %d2, %d3, (%a1)+, %d2, %acc0 |
- move.l %d4, (%a2)+ | output to original buffer
- move.l %d4, (%a3)+ |
- subq.l #1, %d0 |
- bgt.s 10b | loop |
-20: | loop done |
- movclr.l %acc0, %d4 | output last sample
- move.l %d4, (%a2) |
- move.l %d4, (%a3) |
- movem.l (%sp), %d2-%d4/%a2-%a3 | restore registers
- lea.l 20(%sp), %sp | cleanup
- rts |
- .size channels_process_sound_chan_mono, \
- .-channels_process_sound_chan_mono
-
-/****************************************************************************
- * void channels_process_sound_chan_custom(int count, int32_t *buf[])
- *
- * Apply stereo width (narrowing/expanding) effect.
- */
- .section .text
- .align 2
- .global channels_process_sound_chan_custom
-channels_process_sound_chan_custom:
- movem.l 4(%sp), %d0/%a0 | %d0 = count, %a0 = buf
- lea.l -28(%sp), %sp | save registers
- movem.l %d2-%d6/%a2-%a3, (%sp) |
- movem.l (%a0), %a0-%a1 | get channel pointers
- move.l %a0, %a2 | use separate dst pointers since read
- move.l %a1, %a3 | pointers run one ahead of write
- move.l dsp_sw_gain, %d3 | load straight (mid) gain
- move.l dsp_sw_cross, %d4 | load cross (side) gain
- move.l (%a0)+, %d1 | prime the input registers
- move.l (%a1)+, %d2 |
- mac.l %d1, %d3 , %acc0 | L = l*gain + r*cross
- mac.l %d1, %d4, (%a0)+, %d1, %acc1 | R = r*gain + l*cross
- mac.l %d2, %d4 , %acc0 |
- mac.l %d2, %d3, (%a1)+, %d2, %acc1 |
- subq.l #1, %d0 |
- ble.b 20f | loop done |
-10: | loop |
- movclr.l %acc0, %d5 |
- movclr.l %acc1, %d6 |
- mac.l %d1, %d3 , %acc0 | L = l*gain + r*cross
- mac.l %d1, %d4, (%a0)+, %d1, %acc1 | R = r*gain + l*cross
- mac.l %d2, %d4 , %acc0 |
- mac.l %d2, %d3, (%a1)+, %d2, %acc1 |
- move.l %d5, (%a2)+ |
- move.l %d6, (%a3)+ |
- subq.l #1, %d0 |
- bgt.s 10b | loop |
-20: | loop done |
- movclr.l %acc0, %d5 | output last sample
- movclr.l %acc1, %d6 |
- move.l %d5, (%a2) |
- move.l %d6, (%a3) |
- movem.l (%sp), %d2-%d6/%a2-%a3 | restore registers
- lea.l 28(%sp), %sp | cleanup
- rts |
- .size channels_process_sound_chan_custom, \
- .-channels_process_sound_chan_custom
-
-/****************************************************************************
- * void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
- *
- * Separate channels into side channels.
- */
- .section .text
- .align 2
- .global channels_process_sound_chan_karaoke
-channels_process_sound_chan_karaoke:
- movem.l 4(%sp), %d0/%a0 | %d0 = count, %a0 = buf
- lea.l -20(%sp), %sp | save registers
- movem.l %d2-%d4/%a2-%a3, (%sp) |
- movem.l (%a0), %a0-%a1 | get channel src pointers
- move.l %a0, %a2 | use separate dst pointers since read
- move.l %a1, %a3 | pointers run one ahead of write
- move.l #0x40000000, %d3 | %d3 = 0.5
- move.l (%a0)+, %d1 | prime the input registers
- move.l (%a1)+, %d2 |
- mac.l %d1, %d3, (%a0)+, %d1, %acc0 | L = l/2 - r/2
- msac.l %d2, %d3, (%a1)+, %d2, %acc0 |
- subq.l #1, %d0 |
- ble.b 20f | loop done |
-10: | loop |
- movclr.l %acc0, %d4 |
- mac.l %d1, %d3, (%a0)+, %d1, %acc0 | L = l/2 - r/2
- msac.l %d2, %d3, (%a1)+, %d2, %acc0 |
- move.l %d4, (%a2)+ |
- neg.l %d4 | R = -L = -(l/2 - r/2) = r/2 - l/2
- move.l %d4, (%a3)+ |
- subq.l #1, %d0 |
- bgt.s 10b | loop |
-20: | loop done |
- movclr.l %acc0, %d4 | output last sample
- move.l %d4, (%a2) |
- neg.l %d4 | R = -L = -(l/2 - r/2) = r/2 - l/2
- move.l %d4, (%a3) |
- movem.l (%sp), %d2-%d4/%a2-%a3 | restore registers
- lea.l 20(%sp), %sp | cleanup
- rts |
- .size channels_process_sound_chan_karaoke, \
- .-channels_process_sound_chan_karaoke
-
-/****************************************************************************
- * void sample_output_stereo(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- *
- * Framework based on the ubiquitous Rockbox line transfer logic for
- * Coldfire CPUs.
- *
- * Does emac clamping and scaling (which proved faster than the usual
- * checks and branches - even single test clamping) and writes using
- * line burst transfers. Also better than writing a single L-R pair per
- * loop but a good deal more code.
- *
- * Attemping bursting during reads is rather futile since the source and
- * destination alignments rarely agree and too much complication will
- * slow us up. The parallel loads seem to do a bit better at least until
- * a pcm buffer can always give line aligned chunk and then aligning the
- * dest can then imply the source is aligned if the source buffers are.
- * For now longword alignment is assumed of both the source and dest.
- *
- */
- .section .text
- .align 2
- .global sample_output_stereo
-sample_output_stereo:
- lea.l -48(%sp), %sp | save registers
- move.l %macsr, %d1 | do it now as at many lines will
- movem.l %d1-%d7/%a2-%a6, (%sp) | be the far more common condition
- move.l #0x80, %macsr | put emac unit in signed int mode
- movem.l 52(%sp), %a0-%a2/%a4 |
- lea.l (%a4, %a0.l*4), %a0 | %a0 = end address
- move.l (%a1), %d1 | %a1 = multiplier: (1 << (16 - scale))
- sub.l #16, %d1 |
- neg.l %d1 |
- moveq.l #1, %d0 |
- asl.l %d1, %d0 |
- move.l %d0, %a1 |
- move.l #0x8000, %a6 | %a6 = rounding term
- movem.l (%a2), %a2-%a3 | get L/R channel pointers
- moveq.l #28, %d0 | %d0 = second line bound
- add.l %a4, %d0 |
- and.l #0xfffffff0, %d0 |
- cmp.l %a0, %d0 | at least a full line?
- bhi.w 40f | long loop 1 start | no? do as trailing longwords
- sub.l #16, %d0 | %d1 = first line bound
- cmp.l %a4, %d0 | any leading longwords?
- bls.b 20f | line loop start | no? start line loop
-10: | long loop 0 |
- move.l (%a2)+, %d1 | read longword from L and R
- move.l %a6, %acc0 |
- move.l %acc0, %acc1 |
- mac.l %d1, %a1, (%a3)+, %d2, %acc0 | shift L to high word
- mac.l %d2, %a1, %acc1 | shift R to high word
- movclr.l %acc0, %d1 | get possibly saturated results
- movclr.l %acc1, %d2 |
- swap %d2 | move R to low word
- move.w %d2, %d1 | interleave MS 16 bits of each
- move.l %d1, (%a4)+ | ...and write both
- cmp.l %a4, %d0 |
- bhi.b 10b | long loop 0 |
-20: | line loop start |
- lea.l -12(%a0), %a5 | %a5 = at or just before last line bound
-30: | line loop |
- move.l (%a3)+, %d4 | get next 4 R samples and scale
- move.l %a6, %acc0 |
- move.l %acc0, %acc1 |
- move.l %acc1, %acc2 |
- move.l %acc2, %acc3 |
- mac.l %d4, %a1, (%a3)+, %d5, %acc0 | with saturation
- mac.l %d5, %a1, (%a3)+, %d6, %acc1 |
- mac.l %d6, %a1, (%a3)+, %d7, %acc2 |
- mac.l %d7, %a1, (%a2)+, %d0, %acc3 |
- lea.l 16(%a4), %a4 | increment dest here, mitigate stalls
- movclr.l %acc0, %d4 | obtain R results
- movclr.l %acc1, %d5 |
- movclr.l %acc2, %d6 |
- movclr.l %acc3, %d7 |
- move.l %a6, %acc0 |
- move.l %acc0, %acc1 |
- move.l %acc1, %acc2 |
- move.l %acc2, %acc3 |
- mac.l %d0, %a1, (%a2)+, %d1, %acc0 | get next 4 L samples and scale
- mac.l %d1, %a1, (%a2)+, %d2, %acc1 | with saturation
- mac.l %d2, %a1, (%a2)+, %d3, %acc2 |
- mac.l %d3, %a1 , %acc3 |
- swap %d4 | a) interleave most significant...
- swap %d5 |
- swap %d6 |
- swap %d7 |
- movclr.l %acc0, %d0 | obtain L results
- movclr.l %acc1, %d1 |
- movclr.l %acc2, %d2 |
- movclr.l %acc3, %d3 |
- move.w %d4, %d0 | a) ... 16 bits of L and R
- move.w %d5, %d1 |
- move.w %d6, %d2 |
- move.w %d7, %d3 |
- movem.l %d0-%d3, -16(%a4) | write four stereo samples
- cmp.l %a4, %a5 |
- bhi.b 30b | line loop |
-40: | long loop 1 start |
- cmp.l %a4, %a0 | any longwords left?
- bls.b 60f | output end | no? stop
-50: | long loop 1 |
- move.l (%a2)+, %d1 | handle trailing longwords
- move.l %a6, %acc0 |
- move.l %acc0, %acc1 |
- mac.l %d1, %a1, (%a3)+, %d2, %acc0 | the same way as leading ones
- mac.l %d2, %a1, %acc1 |
- movclr.l %acc0, %d1 |
- movclr.l %acc1, %d2 |
- swap %d2 |
- move.w %d2, %d1 |
- move.l %d1, (%a4)+ |
- cmp.l %a4, %a0 |
- bhi.b 50b | long loop 1
-60: | output end |
- movem.l (%sp), %d1-%d7/%a2-%a6 | restore registers
- move.l %d1, %macsr |
- lea.l 48(%sp), %sp | cleanup
- rts |
- .size sample_output_stereo, .-sample_output_stereo
-
-/****************************************************************************
- * void sample_output_mono(int count, struct dsp_data *data,
- * const int32_t *src[], int16_t *dst)
- *
- * Same treatment as sample_output_stereo but for one channel.
- */
- .section .text
- .align 2
- .global sample_output_mono
-sample_output_mono:
- lea.l -32(%sp), %sp | save registers
- move.l %macsr, %d1 | do it now as at many lines will
- movem.l %d1-%d5/%a2-%a4, (%sp) | be the far more common condition
- move.l #0x80, %macsr | put emac unit in signed int mode
- movem.l 36(%sp), %a0-%a3 |
- lea.l (%a3, %a0.l*4), %a0 | %a0 = end address
- move.l (%a1), %d1 | %d5 = multiplier: (1 << (16 - scale))
- sub.l #16, %d1 |
- neg.l %d1 |
- moveq.l #1, %d5 |
- asl.l %d1, %d5 |
- move.l #0x8000, %a4 | %a4 = rounding term
- movem.l (%a2), %a2 | get source channel pointer
- moveq.l #28, %d0 | %d0 = second line bound
- add.l %a3, %d0 |
- and.l #0xfffffff0, %d0 |
- cmp.l %a0, %d0 | at least a full line?
- bhi.w 40f | long loop 1 start | no? do as trailing longwords
- sub.l #16, %d0 | %d1 = first line bound
- cmp.l %a3, %d0 | any leading longwords?
- bls.b 20f | line loop start | no? start line loop
-10: | long loop 0 |
- move.l (%a2)+, %d1 | read longword from L and R
- move.l %a4, %acc0 |
- mac.l %d1, %d5, %acc0 | shift L to high word
- movclr.l %acc0, %d1 | get possibly saturated results
- move.l %d1, %d2 |
- swap %d2 | move R to low word
- move.w %d2, %d1 | duplicate single channel into
- move.l %d1, (%a3)+ | L and R
- cmp.l %a3, %d0 |
- bhi.b 10b | long loop 0 |
-20: | line loop start |
- lea.l -12(%a0), %a1 | %a1 = at or just before last line bound
-30: | line loop |
- move.l (%a2)+, %d0 | get next 4 L samples and scale
- move.l %a4, %acc0 |
- move.l %acc0, %acc1 |
- move.l %acc1, %acc2 |
- move.l %acc2, %acc3 |
- mac.l %d0, %d5, (%a2)+, %d1, %acc0 | with saturation
- mac.l %d1, %d5, (%a2)+, %d2, %acc1 |
- mac.l %d2, %d5, (%a2)+, %d3, %acc2 |
- mac.l %d3, %d5 , %acc3 |
- lea.l 16(%a3), %a3 | increment dest here, mitigate stalls
- movclr.l %acc0, %d0 | obtain results
- movclr.l %acc1, %d1 |
- movclr.l %acc2, %d2 |
- movclr.l %acc3, %d3 |
- move.l %d0, %d4 | duplicate single channel
- swap %d4 | into L and R
- move.w %d4, %d0 |
- move.l %d1, %d4 |
- swap %d4 |
- move.w %d4, %d1 |
- move.l %d2, %d4 |
- swap %d4 |
- move.w %d4, %d2 |
- move.l %d3, %d4 |
- swap %d4 |
- move.w %d4, %d3 |
- movem.l %d0-%d3, -16(%a3) | write four stereo samples
- cmp.l %a3, %a1 |
- bhi.b 30b | line loop |
-40: | long loop 1 start |
- cmp.l %a3, %a0 | any longwords left?
- bls.b 60f | output end | no? stop
-50: | loop loop 1 |
- move.l (%a2)+, %d1 | handle trailing longwords
- move.l %a4, %acc0 |
- mac.l %d1, %d5, %acc0 | the same way as leading ones
- movclr.l %acc0, %d1 |
- move.l %d1, %d2 |
- swap %d2 |
- move.w %d2, %d1 |
- move.l %d1, (%a3)+ |
- cmp.l %a3, %a0 |
- bhi.b 50b | long loop 1 |
-60: | output end |
- movem.l (%sp), %d1-%d5/%a2-%a4 | restore registers
- move.l %d1, %macsr |
- lea.l 32(%sp), %sp | cleanup
- rts |
- .size sample_output_mono, .-sample_output_mono
diff --git a/apps/eq.c b/apps/eq.c
deleted file mode 100644
index 122a46a4c5..0000000000
--- a/apps/eq.c
+++ /dev/null
@@ -1,268 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006-2007 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <inttypes.h>
-#include "config.h"
-#include "fixedpoint.h"
-#include "fracmul.h"
-#include "eq.h"
-#include "replaygain.h"
-
-/**
- * Calculate first order shelving filter. Filter is not directly usable by the
- * eq_filter() function.
- * @param cutoff shelf midpoint frequency. See eq_pk_coefs for format.
- * @param A decibel value multiplied by ten, describing gain/attenuation of
- * shelf. Max value is 24 dB.
- * @param low true for low-shelf filter, false for high-shelf filter.
- * @param c pointer to coefficient storage. Coefficients are s4.27 format.
- */
-void filter_shelf_coefs(unsigned long cutoff, long A, bool low, int32_t *c)
-{
- long sin, cos;
- int32_t b0, b1, a0, a1; /* s3.28 */
- const long g = get_replaygain_int(A*5) << 4; /* 10^(db/40), s3.28 */
-
- sin = fp_sincos(cutoff/2, &cos);
- if (low) {
- const int32_t sin_div_g = fp_div(sin, g, 25);
- const int32_t sin_g = FRACMUL(sin, g);
- cos >>= 3;
- b0 = sin_g + cos; /* 0.25 .. 4.10 */
- b1 = sin_g - cos; /* -1 .. 3.98 */
- a0 = sin_div_g + cos; /* 0.25 .. 4.10 */
- a1 = sin_div_g - cos; /* -1 .. 3.98 */
- } else {
- const int32_t cos_div_g = fp_div(cos, g, 25);
- const int32_t cos_g = FRACMUL(cos, g);
- sin >>= 3;
- b0 = sin + cos_g; /* 0.25 .. 4.10 */
- b1 = sin - cos_g; /* -3.98 .. 1 */
- a0 = sin + cos_div_g; /* 0.25 .. 4.10 */
- a1 = sin - cos_div_g; /* -3.98 .. 1 */
- }
-
- const int32_t rcp_a0 = fp_div(1, a0, 57); /* 0.24 .. 3.98, s2.29 */
- *c++ = FRACMUL_SHL(b0, rcp_a0, 1); /* 0.063 .. 15.85 */
- *c++ = FRACMUL_SHL(b1, rcp_a0, 1); /* -15.85 .. 15.85 */
- *c++ = -FRACMUL_SHL(a1, rcp_a0, 1); /* -1 .. 1 */
-}
-
-#ifdef HAVE_SW_TONE_CONTROLS
-/**
- * Calculate second order section filter consisting of one low-shelf and one
- * high-shelf section.
- * @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
- * @param cutoff_high high-shelf midpoint frequency.
- * @param A_low decibel value multiplied by ten, describing gain/attenuation of
- * low-shelf part. Max value is 24 dB.
- * @param A_high decibel value multiplied by ten, describing gain/attenuation of
- * high-shelf part. Max value is 24 dB.
- * @param A decibel value multiplied by ten, describing additional overall gain.
- * @param c pointer to coefficient storage. Coefficients are s4.27 format.
- */
-void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
- long A_low, long A_high, long A, int32_t *c)
-{
- const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
- int32_t c_ls[3], c_hs[3];
-
- filter_shelf_coefs(cutoff_low, A_low, true, c_ls);
- filter_shelf_coefs(cutoff_high, A_high, false, c_hs);
- c_ls[0] = FRACMUL(g, c_ls[0]);
- c_ls[1] = FRACMUL(g, c_ls[1]);
-
- /* now we cascade the two first order filters to one second order filter
- * which can be used by eq_filter(). these resulting coefficients have a
- * really wide numerical range, so we use a fixed point format which will
- * work for the selected cutoff frequencies (in dsp.c) only.
- */
- const int32_t b0 = c_ls[0], b1 = c_ls[1], b2 = c_hs[0], b3 = c_hs[1];
- const int32_t a0 = c_ls[2], a1 = c_hs[2];
- *c++ = FRACMUL_SHL(b0, b2, 4);
- *c++ = FRACMUL_SHL(b0, b3, 4) + FRACMUL_SHL(b1, b2, 4);
- *c++ = FRACMUL_SHL(b1, b3, 4);
- *c++ = a0 + a1;
- *c++ = -FRACMUL_SHL(a0, a1, 4);
-}
-#endif
-
-/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
- * Slightly faster calculation can be done by deriving forms which use tan()
- * instead of cos() and sin(), but the latter are far easier to use when doing
- * fixed point math, and performance is not a big point in the calculation part.
- * All the 'a' filter coefficients are negated so we can use only additions
- * in the filtering equation.
- */
-
-/**
- * Calculate second order section peaking filter coefficients.
- * @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
- * 0x80000000 represents the Nyquist frequency (samplerate/2).
- * @param Q Q factor value multiplied by ten. Lower bound is artificially set
- * at 0.5.
- * @param db decibel value multiplied by ten, describing gain/attenuation at
- * peak freq. Max value is 24 dB.
- * @param c pointer to coefficient storage. Coefficients are s3.28 format.
- */
-void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
-{
- long cs;
- const long one = 1 << 28; /* s3.28 */
- const long A = get_replaygain_int(db*5) << 5; /* 10^(db/40), s2.29 */
- const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
- int32_t a0, a1, a2; /* these are all s3.28 format */
- int32_t b0, b1, b2;
- const long alphadivA = fp_div(alpha, A, 27);
- const long alphaA = FRACMUL(alpha, A);
-
- /* possible numerical ranges are in comments by each coef */
- b0 = one + alphaA; /* [1 .. 5] */
- b1 = a1 = -2*(cs >> 3); /* [-2 .. 2] */
- b2 = one - alphaA; /* [-3 .. 1] */
- a0 = one + alphadivA; /* [1 .. 5] */
- a2 = one - alphadivA; /* [-3 .. 1] */
-
- /* range of this is roughly [0.2 .. 1], but we'll never hit 1 completely */
- const long rcp_a0 = fp_div(1, a0, 59); /* s0.31 */
- *c++ = FRACMUL(b0, rcp_a0); /* [0.25 .. 4] */
- *c++ = FRACMUL(b1, rcp_a0); /* [-2 .. 2] */
- *c++ = FRACMUL(b2, rcp_a0); /* [-2.4 .. 1] */
- *c++ = FRACMUL(-a1, rcp_a0); /* [-2 .. 2] */
- *c++ = FRACMUL(-a2, rcp_a0); /* [-0.6 .. 1] */
-}
-
-/**
- * Calculate coefficients for lowshelf filter. Parameters are as for
- * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
- */
-void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
-{
- long cs;
- const long one = 1 << 25; /* s6.25 */
- const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
- const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
- const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
- const long ap1 = (A >> 4) + one;
- const long am1 = (A >> 4) - one;
- const long ap1_cs = FRACMUL(ap1, cs);
- const long am1_cs = FRACMUL(am1, cs);
- const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
- int32_t a0, a1, a2; /* these are all s6.25 format */
- int32_t b0, b1, b2;
-
- /* [0.1 .. 40] */
- b0 = FRACMUL_SHL(A, ap1 - am1_cs + twosqrtalpha, 2);
- /* [-16 .. 63.4] */
- b1 = FRACMUL_SHL(A, am1 - ap1_cs, 3);
- /* [0 .. 31.7] */
- b2 = FRACMUL_SHL(A, ap1 - am1_cs - twosqrtalpha, 2);
- /* [0.5 .. 10] */
- a0 = ap1 + am1_cs + twosqrtalpha;
- /* [-16 .. 4] */
- a1 = -2*(am1 + ap1_cs);
- /* [0 .. 8] */
- a2 = ap1 + am1_cs - twosqrtalpha;
-
- /* [0.1 .. 1.99] */
- const long rcp_a0 = fp_div(1, a0, 55); /* s1.30 */
- *c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0.06 .. 15.9] */
- *c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-2 .. 31.7] */
- *c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 15.9] */
- *c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
- *c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
-}
-
-/**
- * Calculate coefficients for highshelf filter. Parameters are as for
- * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
- */
-void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
-{
- long cs;
- const long one = 1 << 25; /* s6.25 */
- const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
- const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
- const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
- const long ap1 = (A >> 4) + one;
- const long am1 = (A >> 4) - one;
- const long ap1_cs = FRACMUL(ap1, cs);
- const long am1_cs = FRACMUL(am1, cs);
- const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
- int32_t a0, a1, a2; /* these are all s6.25 format */
- int32_t b0, b1, b2;
-
- /* [0.1 .. 40] */
- b0 = FRACMUL_SHL(A, ap1 + am1_cs + twosqrtalpha, 2);
- /* [-63.5 .. 16] */
- b1 = -FRACMUL_SHL(A, am1 + ap1_cs, 3);
- /* [0 .. 32] */
- b2 = FRACMUL_SHL(A, ap1 + am1_cs - twosqrtalpha, 2);
- /* [0.5 .. 10] */
- a0 = ap1 - am1_cs + twosqrtalpha;
- /* [-4 .. 16] */
- a1 = 2*(am1 - ap1_cs);
- /* [0 .. 8] */
- a2 = ap1 - am1_cs - twosqrtalpha;
-
- /* [0.1 .. 1.99] */
- const long rcp_a0 = fp_div(1, a0, 55); /* s1.30 */
- *c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0 .. 16] */
- *c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-31.7 .. 2] */
- *c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 16] */
- *c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
- *c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
-}
-
-/* We realise the filters as a second order direct form 1 structure. Direct
- * form 1 was chosen because of better numerical properties for fixed point
- * implementations.
- */
-
-#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM))
-void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
- unsigned channels, unsigned shift)
-{
- unsigned c, i;
- long long acc;
-
- /* Direct form 1 filtering code.
- y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
- where y[] is output and x[] is input.
- */
-
- for (c = 0; c < channels; c++) {
- for (i = 0; i < num; i++) {
- acc = (long long) x[c][i] * f->coefs[0];
- acc += (long long) f->history[c][0] * f->coefs[1];
- acc += (long long) f->history[c][1] * f->coefs[2];
- acc += (long long) f->history[c][2] * f->coefs[3];
- acc += (long long) f->history[c][3] * f->coefs[4];
- f->history[c][1] = f->history[c][0];
- f->history[c][0] = x[c][i];
- f->history[c][3] = f->history[c][2];
- x[c][i] = (acc << shift) >> 32;
- f->history[c][2] = x[c][i];
- }
- }
-}
-#endif
-
diff --git a/apps/eq.h b/apps/eq.h
deleted file mode 100644
index a44e9153ac..0000000000
--- a/apps/eq.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006-2007 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _EQ_H
-#define _EQ_H
-
-#include <inttypes.h>
-#include <stdbool.h>
-
-/* These depend on the fixed point formats used by the different filter types
- and need to be changed when they change.
- */
-#define FILTER_BISHELF_SHIFT 5
-#define EQ_PEAK_SHIFT 4
-#define EQ_SHELF_SHIFT 6
-
-struct eqfilter {
- int32_t coefs[5]; /* Order is b0, b1, b2, a1, a2 */
- int32_t history[2][4];
-};
-
-void filter_shelf_coefs(unsigned long cutoff, long A, bool low, int32_t *c);
-void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
- long A_low, long A_high, long A, int32_t *c);
-void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);
-void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);
-void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);
-void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
- unsigned channels, unsigned shift);
-
-#endif
-
diff --git a/apps/eq_arm.S b/apps/eq_arm.S
deleted file mode 100644
index b0e1771e89..0000000000
--- a/apps/eq_arm.S
+++ /dev/null
@@ -1,89 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006-2007 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include "config.h"
-
-/* uncomment this to make filtering calculate lower bits after shifting.
- * without this, "shift" of the lower bits will be lost here.
- */
-/* #define HIGH_PRECISION */
-
-/*
- * void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
- * unsigned channels, unsigned shift)
- */
-#if CONFIG_CPU == PP5002
- .section .icode,"ax",%progbits
-#else
- .text
-#endif
- .global eq_filter
-eq_filter:
- ldr r12, [sp] @ get shift parameter
- stmdb sp!, { r0-r11, lr } @ save all params and clobbered regs
- ldmia r1!, { r4-r8 } @ load coefs
- mov r10, r1 @ loop prelude expects filter struct addr in r10
-
-.filterloop:
- ldr r9, [sp] @ get pointer to this channels data
- add r0, r9, #4
- str r0, [sp] @ save back pointer to next channels data
- ldr r9, [r9] @ r9 = x[]
- ldr r14, [sp, #8] @ r14 = numsamples
- ldmia r10, { r0-r3 } @ load history, r10 should be filter struct addr
- str r10, [sp, #4] @ save it for loop end
-
- /* r0-r3 = history, r4-r8 = coefs, r9 = x[], r10..r11 = accumulator,
- * r12 = shift amount, r14 = number of samples.
- */
-.loop:
- /* Direct form 1 filtering code.
- * y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
- * where y[] is output and x[] is input. This is performed out of order to
- * reuse registers, we're pretty short on regs.
- */
- smull r10, r11, r6, r1 @ acc = b2*x[i - 2]
- mov r1, r0 @ fix input history
- smlal r10, r11, r5, r0 @ acc += b1*x[i - 1]
- ldr r0, [r9] @ load input and fix history in same operation
- smlal r10, r11, r7, r2 @ acc += a1*y[i - 1]
- smlal r10, r11, r8, r3 @ acc += a2*y[i - 2]
- smlal r10, r11, r4, r0 @ acc += b0*x[i] /* avoid stall on arm9*/
- mov r3, r2 @ fix output history
- mov r2, r11, asl r12 @ get upper part of result and shift left
-#ifdef HIGH_PRECISION
- rsb r11, r12, #32 @ get shift amount for lower part
- orr r2, r2, r10, lsr r11 @ then mix in correctly shifted lower part
-#endif
- str r2, [r9], #4 @ save result
- subs r14, r14, #1 @ are we done with this channel?
- bne .loop
-
- ldr r10, [sp, #4] @ load filter struct pointer
- stmia r10!, { r0-r3 } @ save back history
- ldr r11, [sp, #12] @ load number of channels
- subs r11, r11, #1 @ all channels processed?
- strne r11, [sp, #12]
- bne .filterloop
-
- add sp, sp, #16 @ compensate for temp storage
- ldmpc regs=r4-r11
-
diff --git a/apps/eq_cf.S b/apps/eq_cf.S
deleted file mode 100644
index 30a28b9d99..0000000000
--- a/apps/eq_cf.S
+++ /dev/null
@@ -1,91 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006-2007 Thom Johansen
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-/* uncomment this to make filtering calculate lower bits after shifting.
- * without this, "shift" - 1 of the lower bits will be lost here.
- */
-/* #define HIGH_PRECISION */
-
-/*
- * void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
- * unsigned channels, unsigned shift)
- */
- .text
- .global eq_filter
-eq_filter:
- lea.l (-11*4, %sp), %sp
- movem.l %d2-%d7/%a2-%a6, (%sp) | save clobbered regs
- move.l (11*4+8, %sp), %a5 | fetch filter structure address
- move.l (11*4+20, %sp), %d7 | load shift count
- subq.l #1, %d7 | EMAC gives us one free shift
-#ifdef HIGH_PRECISION
- moveq.l #8, %d6
- sub.l %d7, %d6 | shift for lower part of accumulator
-#endif
- movem.l (%a5), %a0-%a4 | load coefs
- lea.l (5*4, %a5), %a5 | point to filter history
-
-.filterloop:
- move.l (11*4+4, %sp), %a6 | load input channel pointer
- addq.l #4, (11*4+4, %sp) | point x to next channel
- move.l (%a6), %a6
- move.l (11*4+12, %sp), %d5 | number of samples
- movem.l (%a5), %d0-%d3 | load filter history
-
- /* d0-d3 = history, d4 = temp, d5 = sample count, d6 = lower shift amount,
- * d7 = upper shift amount, a0-a4 = coefs, a5 = history pointer, a6 = x[]
- */
-.loop:
- /* Direct form 1 filtering code. We assume DSP has put EMAC in frac mode.
- * y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
- * where y[] is output and x[] is input. This is performed out of order
- * to do parallel load of input value.
- */
- mac.l %a2, %d1, %acc0 | acc = b2*x[i - 2]
- move.l %d0, %d1 | fix input history
- mac.l %a1, %d0, (%a6), %d0, %acc0 | acc += b1*x[i - 1], x[i] -> d0
- mac.l %a0, %d0, %acc0 | acc += b0*x[i]
- mac.l %a3, %d2, %acc0 | acc += a1*y[i - 1]
- mac.l %a4, %d3, %acc0 | acc += a2*y[i - 2]
- move.l %d2, %d3 | fix output history
-#ifdef HIGH_PRECISION
- move.l %accext01, %d2 | fetch lower part of accumulator
- move.b %d2, %d4 | clear upper three bytes
- lsr.l %d6, %d4 | shift lower bits
-#endif
- movclr.l %acc0, %d2 | fetch upper part of result
- asl.l %d7, %d2 | restore fixed point format
-#ifdef HIGH_PRECISION
- or.l %d2, %d4 | combine lower and upper parts
-#endif
- move.l %d2, (%a6)+ | save result
- subq.l #1, %d5 | are we done with this channel?
- jne .loop
-
- movem.l %d0-%d3, (%a5) | save history back to struct
- lea.l (4*4, %a5), %a5 | point to next channel's history
- subq.l #1, (11*4+16, %sp) | have we processed both channels?
- jne .filterloop
-
- movem.l (%sp), %d2-%d7/%a2-%a6
- lea.l (11*4, %sp), %sp
- rts
-
diff --git a/apps/eqs/Acoustic.cfg b/apps/eqs/Acoustic.cfg
deleted file mode 100644
index 34b5ed8a2b..0000000000
--- a/apps/eqs/Acoustic.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 45
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 45
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 10
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 15
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 30
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 20
diff --git a/apps/eqs/Bass.cfg b/apps/eqs/Bass.cfg
deleted file mode 100644
index 2742459081..0000000000
--- a/apps/eqs/Bass.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 50
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 50
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 35
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 15
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 5
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: -5
diff --git a/apps/eqs/Classical.cfg b/apps/eqs/Classical.cfg
deleted file mode 100644
index bf2f9f9566..0000000000
--- a/apps/eqs/Classical.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 50
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 50
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 40
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: -20
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 10
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 20
diff --git a/apps/eqs/Default.cfg b/apps/eqs/Default.cfg
deleted file mode 100644
index d6f345fa9e..0000000000
--- a/apps/eqs/Default.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: off
-eq precut: 0
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 0
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 0
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 0
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 0
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 0
diff --git a/apps/eqs/Disco.cfg b/apps/eqs/Disco.cfg
deleted file mode 100644
index f894f26da1..0000000000
--- a/apps/eqs/Disco.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 45
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 30
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 10
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 45
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 25
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 10
diff --git a/apps/eqs/Electronic.cfg b/apps/eqs/Electronic.cfg
deleted file mode 100644
index e70c911272..0000000000
--- a/apps/eqs/Electronic.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 55
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 45
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 5
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 25
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 15
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 55
diff --git a/apps/eqs/Hip-Hop.cfg b/apps/eqs/Hip-Hop.cfg
deleted file mode 100644
index 2d38425dc4..0000000000
--- a/apps/eqs/Hip-Hop.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 65
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 65
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 25
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: -10
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 15
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 35
diff --git a/apps/eqs/Jazz.cfg b/apps/eqs/Jazz.cfg
deleted file mode 100644
index f576f9fcc1..0000000000
--- a/apps/eqs/Jazz.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 60
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 40
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 15
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: -25
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 5
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 60
diff --git a/apps/eqs/Lounge.cfg b/apps/eqs/Lounge.cfg
deleted file mode 100644
index 39ae23a7e7..0000000000
--- a/apps/eqs/Lounge.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 20
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: -25
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 5
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 20
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: -15
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 15
diff --git a/apps/eqs/Pop.cfg b/apps/eqs/Pop.cfg
deleted file mode 100644
index 1d8cefe173..0000000000
--- a/apps/eqs/Pop.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 50
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: -10
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 5
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 50
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 15
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: -10
diff --git a/apps/eqs/R&B.cfg b/apps/eqs/R&B.cfg
deleted file mode 100644
index a460b587f5..0000000000
--- a/apps/eqs/R&B.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 45
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 35
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 45
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 5
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 25
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 30
diff --git a/apps/eqs/Rock.cfg b/apps/eqs/Rock.cfg
deleted file mode 100644
index ec4f0356a8..0000000000
--- a/apps/eqs/Rock.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 45
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: 25
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 10
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 0
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 20
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 45
diff --git a/apps/eqs/Vocal.cfg b/apps/eqs/Vocal.cfg
deleted file mode 100644
index 1de754f07c..0000000000
--- a/apps/eqs/Vocal.cfg
+++ /dev/null
@@ -1,17 +0,0 @@
-eq enabled: on
-eq precut: 45
-eq band 0 cutoff: 60
-eq band 0 q: 7
-eq band 0 gain: -45
-eq band 1 cutoff: 200
-eq band 1 q: 10
-eq band 1 gain: 5
-eq band 2 cutoff: 800
-eq band 2 q: 10
-eq band 2 gain: 45
-eq band 3 cutoff: 4000
-eq band 3 q: 10
-eq band 3 gain: 20
-eq band 4 cutoff: 12000
-eq band 4 q: 7
-eq band 4 gain: 0
diff --git a/apps/fracmul.h b/apps/fracmul.h
index 6aaedaf3e6..47b85e59ef 100644
--- a/apps/fracmul.h
+++ b/apps/fracmul.h
@@ -4,7 +4,7 @@
#include <stdint.h>
#include "gcc_extensions.h"
-/** FRACTIONAL MULTIPLICATION - TAKEN FROM apps/dsp.h
+/** FRACTIONAL MULTIPLICATION
* Multiply two fixed point numbers with 31 fractional bits:
* FRACMUL(x, y)
*
diff --git a/apps/metadata.c b/apps/metadata.c
deleted file mode 100644
index 2a93c1880c..0000000000
--- a/apps/metadata.c
+++ /dev/null
@@ -1,641 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include "string-extra.h"
-
-#include "debug.h"
-#include "logf.h"
-#include "settings.h"
-#include "cuesheet.h"
-#include "metadata.h"
-
-#include "metadata/metadata_parsers.h"
-
-#if CONFIG_CODEC == SWCODEC
-
-/* For trailing tag stripping and base audio data types */
-#include "buffering.h"
-
-#include "metadata/metadata_common.h"
-
-static bool get_shn_metadata(int fd, struct mp3entry *id3)
-{
- /* TODO: read the id3v2 header if it exists */
- id3->vbr = true;
- id3->filesize = filesize(fd);
- return skip_id3v2(fd, id3);
-}
-
-static bool get_other_asap_metadata(int fd, struct mp3entry *id3)
-{
- id3->bitrate = 706;
- id3->frequency = 44100;
- id3->vbr = false;
- id3->filesize = filesize(fd);
- id3->genre_string = id3_get_num_genre(36);
- return true;
-}
-#endif /* CONFIG_CODEC == SWCODEC */
-bool write_metadata_log = false;
-
-const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
-{
- /* Unknown file format */
- [0 ... AFMT_NUM_CODECS-1] =
- AFMT_ENTRY("???", NULL, NULL, NULL, "\0" ),
-
- /* MPEG Audio layer 2 */
- [AFMT_MPA_L2] =
- AFMT_ENTRY("MP2", "mpa", NULL, get_mp3_metadata, "mpa\0mp2\0"),
-
-#if CONFIG_CODEC != SWCODEC
- /* MPEG Audio layer 3 on HWCODEC: .talk clips, no encoder */
- [AFMT_MPA_L3] =
- AFMT_ENTRY("MP3", "mpa", NULL, get_mp3_metadata, "mp3\0talk\0"),
-
-#else /* CONFIG_CODEC == SWCODEC */
- /* MPEG Audio layer 3 on SWCODEC */
- [AFMT_MPA_L3] =
- AFMT_ENTRY("MP3", "mpa", "mp3_enc", get_mp3_metadata, "mp3\0"),
-
- /* MPEG Audio layer 1 */
- [AFMT_MPA_L1] =
- AFMT_ENTRY("MP1", "mpa", NULL, get_mp3_metadata, "mp1\0"),
- /* Audio Interchange File Format */
- [AFMT_AIFF] =
- AFMT_ENTRY("AIFF", "aiff", "aiff_enc", get_aiff_metadata, "aiff\0aif\0"),
- /* Uncompressed PCM in a WAV file OR ATRAC3 stream in WAV file (.at3) */
- [AFMT_PCM_WAV] =
- AFMT_ENTRY("WAV", "wav", "wav_enc", get_wave_metadata, "wav\0at3\0"),
- /* Ogg Vorbis */
- [AFMT_OGG_VORBIS] =
- AFMT_ENTRY("Ogg", "vorbis", NULL, get_ogg_metadata, "ogg\0oga\0"),
- /* FLAC */
- [AFMT_FLAC] =
- AFMT_ENTRY("FLAC", "flac", NULL, get_flac_metadata, "flac\0"),
- /* Musepack SV7 */
- [AFMT_MPC_SV7] =
- AFMT_ENTRY("MPCv7", "mpc", NULL, get_musepack_metadata,"mpc\0"),
- /* A/52 (aka AC3) audio */
- [AFMT_A52] =
- AFMT_ENTRY("AC3", "a52", NULL, get_a52_metadata, "a52\0ac3\0"),
- /* WavPack */
- [AFMT_WAVPACK] =
- AFMT_ENTRY("WV","wavpack","wavpack_enc",get_wavpack_metadata,"wv\0"),
- /* Apple Lossless Audio Codec */
- [AFMT_MP4_ALAC] =
- AFMT_ENTRY("ALAC", "alac", NULL, get_mp4_metadata, "m4a\0m4b\0"),
- /* Advanced Audio Coding in M4A container */
- [AFMT_MP4_AAC] =
- AFMT_ENTRY("AAC", "aac", NULL, get_mp4_metadata, "mp4\0"),
- /* Shorten */
- [AFMT_SHN] =
- AFMT_ENTRY("SHN","shorten", NULL, get_shn_metadata, "shn\0"),
- /* SID File Format */
- [AFMT_SID] =
- AFMT_ENTRY("SID", "sid", NULL, get_sid_metadata, "sid\0"),
- /* ADX File Format */
- [AFMT_ADX] =
- AFMT_ENTRY("ADX", "adx", NULL, get_adx_metadata, "adx\0"),
- /* NESM (NES Sound Format) */
- [AFMT_NSF] =
- AFMT_ENTRY("NSF", "nsf", NULL, get_nsf_metadata, "nsf\0nsfe\0"),
- /* Speex File Format */
- [AFMT_SPEEX] =
- AFMT_ENTRY("Speex", "speex",NULL, get_ogg_metadata, "spx\0"),
- /* SPC700 Save State */
- [AFMT_SPC] =
- AFMT_ENTRY("SPC", "spc", NULL, get_spc_metadata, "spc\0"),
- /* APE (Monkey's Audio) */
- [AFMT_APE] =
- AFMT_ENTRY("APE", "ape", NULL, get_monkeys_metadata,"ape\0mac\0"),
- /* WMA (WMAV1/V2 in ASF) */
- [AFMT_WMA] =
- AFMT_ENTRY("WMA", "wma", NULL, get_asf_metadata,"wma\0wmv\0asf\0"),
- /* WMA Professional in ASF */
- [AFMT_WMAPRO] =
- AFMT_ENTRY("WMAPro","wmapro",NULL, NULL, "wma\0wmv\0asf\0"),
- /* Amiga MOD File */
- [AFMT_MOD] =
- AFMT_ENTRY("MOD", "mod", NULL, get_mod_metadata, "mod\0"),
- /* Atari SAP File */
- [AFMT_SAP] =
- AFMT_ENTRY("SAP", "asap", NULL, get_asap_metadata, "sap\0"),
- /* Cook in RM/RA */
- [AFMT_RM_COOK] =
- AFMT_ENTRY("Cook", "cook", NULL, get_rm_metadata,"rm\0ra\0rmvb\0"),
- /* AAC in RM/RA */
- [AFMT_RM_AAC] =
- AFMT_ENTRY("RAAC", "raac", NULL, NULL, "rm\0ra\0rmvb\0"),
- /* AC3 in RM/RA */
- [AFMT_RM_AC3] =
- AFMT_ENTRY("AC3", "a52_rm", NULL, NULL, "rm\0ra\0rmvb\0"),
- /* ATRAC3 in RM/RA */
- [AFMT_RM_ATRAC3] =
- AFMT_ENTRY("ATRAC3","atrac3_rm",NULL, NULL, "rm\0ra\0rmvb\0"),
- /* Atari CMC File */
- [AFMT_CMC] =
- AFMT_ENTRY("CMC", "asap", NULL, get_other_asap_metadata,"cmc\0"),
- /* Atari CM3 File */
- [AFMT_CM3] =
- AFMT_ENTRY("CM3", "asap", NULL, get_other_asap_metadata,"cm3\0"),
- /* Atari CMR File */
- [AFMT_CMR] =
- AFMT_ENTRY("CMR", "asap", NULL, get_other_asap_metadata,"cmr\0"),
- /* Atari CMS File */
- [AFMT_CMS] =
- AFMT_ENTRY("CMS", "asap", NULL, get_other_asap_metadata,"cms\0"),
- /* Atari DMC File */
- [AFMT_DMC] =
- AFMT_ENTRY("DMC", "asap", NULL, get_other_asap_metadata,"dmc\0"),
- /* Atari DLT File */
- [AFMT_DLT] =
- AFMT_ENTRY("DLT", "asap", NULL, get_other_asap_metadata,"dlt\0"),
- /* Atari MPT File */
- [AFMT_MPT] =
- AFMT_ENTRY("MPT", "asap", NULL, get_other_asap_metadata,"mpt\0"),
- /* Atari MPD File */
- [AFMT_MPD] =
- AFMT_ENTRY("MPD", "asap", NULL, get_other_asap_metadata,"mpd\0"),
- /* Atari RMT File */
- [AFMT_RMT] =
- AFMT_ENTRY("RMT", "asap", NULL, get_other_asap_metadata,"rmt\0"),
- /* Atari TMC File */
- [AFMT_TMC] =
- AFMT_ENTRY("TMC", "asap", NULL, get_other_asap_metadata,"tmc\0"),
- /* Atari TM8 File */
- [AFMT_TM8] =
- AFMT_ENTRY("TM8", "asap", NULL, get_other_asap_metadata,"tm8\0"),
- /* Atari TM2 File */
- [AFMT_TM2] =
- AFMT_ENTRY("TM2", "asap", NULL, get_other_asap_metadata,"tm2\0"),
- /* Atrac3 in Sony OMA Container */
- [AFMT_OMA_ATRAC3] =
- AFMT_ENTRY("ATRAC3","atrac3_oma",NULL, get_oma_metadata, "oma\0aa3\0"),
- /* SMAF (Synthetic music Mobile Application Format) */
- [AFMT_SMAF] =
- AFMT_ENTRY("SMAF", "smaf", NULL, get_smaf_metadata, "mmf\0"),
- /* Sun Audio file */
- [AFMT_AU] =
- AFMT_ENTRY("AU", "au", NULL, get_au_metadata, "au\0snd\0"),
- /* VOX (Dialogic telephony file formats) */
- [AFMT_VOX] =
- AFMT_ENTRY("VOX", "vox", NULL, get_vox_metadata, "vox\0"),
- /* Wave64 */
- [AFMT_WAVE64] =
- AFMT_ENTRY("WAVE64","wav64",NULL, get_wave64_metadata,"w64\0"),
- /* True Audio */
- [AFMT_TTA] =
- AFMT_ENTRY("TTA", "tta", NULL, get_tta_metadata, "tta\0"),
- /* WMA Voice in ASF */
- [AFMT_WMAVOICE] =
- AFMT_ENTRY("WMAVoice","wmavoice",NULL, NULL, "wma\0wmv\0"),
- /* Musepack SV8 */
- [AFMT_MPC_SV8] =
- AFMT_ENTRY("MPCv8", "mpc", NULL, get_musepack_metadata,"mpc\0"),
- /* Advanced Audio Coding High Efficiency in M4A container */
- [AFMT_MP4_AAC_HE] =
- AFMT_ENTRY("AAC-HE","aac", NULL, get_mp4_metadata, "mp4\0"),
- /* AY (ZX Spectrum, Amstrad CPC Sound Format) */
- [AFMT_AY] =
- AFMT_ENTRY("AY", "ay", NULL, get_ay_metadata, "ay\0"),
- /* GBS (Game Boy Sound Format) */
- [AFMT_GBS] =
- AFMT_ENTRY("GBS", "gbs", NULL, get_gbs_metadata, "gbs\0"),
- /* HES (Hudson Entertainment System Sound Format) */
- [AFMT_HES] =
- AFMT_ENTRY("HES", "hes", NULL, get_hes_metadata, "hes\0"),
- /* SGC (Sega Master System, Game Gear, Coleco Vision Sound Format) */
- [AFMT_SGC] =
- AFMT_ENTRY("SGC", "sgc", NULL, get_sgc_metadata, "sgc\0"),
- /* VGM (Video Game Music Format) */
- [AFMT_VGM] =
- AFMT_ENTRY("VGM", "vgm", NULL, get_vgm_metadata, "vgm\0vgz\0"),
- /* KSS (MSX computer KSS Music File) */
- [AFMT_KSS] =
- AFMT_ENTRY("KSS", "kss", NULL, get_kss_metadata, "kss\0"),
-#endif
-};
-
-#if CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING)
-/* get REC_FORMAT_* corresponding AFMT_* */
-const int rec_format_afmt[REC_NUM_FORMATS] =
-{
- /* give AFMT_UNKNOWN by default */
- [0 ... REC_NUM_FORMATS-1] = AFMT_UNKNOWN,
- /* add new entries below this line */
- [REC_FORMAT_AIFF] = AFMT_AIFF,
- [REC_FORMAT_MPA_L3] = AFMT_MPA_L3,
- [REC_FORMAT_WAVPACK] = AFMT_WAVPACK,
- [REC_FORMAT_PCM_WAV] = AFMT_PCM_WAV,
-};
-
-#if 0 /* Currently unused, left for reference and future use */
-/* get AFMT_* corresponding REC_FORMAT_* */
-const int afmt_rec_format[AFMT_NUM_CODECS] =
-{
- /* give -1 by default */
- [0 ... AFMT_NUM_CODECS-1] = -1,
- /* add new entries below this line */
- [AFMT_AIFF] = REC_FORMAT_AIFF,
- [AFMT_MPA_L3] = REC_FORMAT_MPA_L3,
- [AFMT_WAVPACK] = REC_FORMAT_WAVPACK,
- [AFMT_PCM_WAV] = REC_FORMAT_PCM_WAV,
-};
-#endif
-#endif /* CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING) */
-
-#if CONFIG_CODEC == SWCODEC
-/* Get the canonical AFMT type */
-int get_audio_base_codec_type(int type)
-{
- int base_type = type;
- switch (type) {
- case AFMT_MPA_L1:
- case AFMT_MPA_L2:
- case AFMT_MPA_L3:
- base_type = AFMT_MPA_L3;
- break;
- case AFMT_MPC_SV7:
- case AFMT_MPC_SV8:
- base_type = AFMT_MPC_SV7;
- break;
- case AFMT_MP4_AAC:
- case AFMT_MP4_AAC_HE:
- base_type = AFMT_MP4_AAC;
- break;
- case AFMT_SAP:
- case AFMT_CMC:
- case AFMT_CM3:
- case AFMT_CMR:
- case AFMT_CMS:
- case AFMT_DMC:
- case AFMT_DLT:
- case AFMT_MPT:
- case AFMT_MPD:
- case AFMT_RMT:
- case AFMT_TMC:
- case AFMT_TM8:
- case AFMT_TM2:
- base_type = AFMT_SAP;
- break;
- default:
- break;
- }
-
- return base_type;
-}
-
-/* Get the basic audio type */
-enum data_type get_audio_base_data_type(int afmt)
-{
- if ((unsigned)afmt >= AFMT_NUM_CODECS)
- return TYPE_UNKNOWN;
-
- switch (get_audio_base_codec_type(afmt))
- {
- case AFMT_NSF:
- case AFMT_SPC:
- case AFMT_SID:
- case AFMT_MOD:
- case AFMT_SAP:
- case AFMT_AY:
- case AFMT_GBS:
- case AFMT_HES:
- case AFMT_SGC:
- case AFMT_VGM:
- case AFMT_KSS:
- /* Type must be allocated and loaded in its entirety onto
- the buffer */
- return TYPE_ATOMIC_AUDIO;
-
- default:
- /* Assume type may be loaded and discarded incrementally */
- return TYPE_PACKET_AUDIO;
-
- case AFMT_UNKNOWN:
- /* Have no idea at all */
- return TYPE_UNKNOWN;
- }
-}
-
-/* Is the format allowed to buffer starting at some offset other than 0
- or first frame only for resume purposes? */
-bool format_buffers_with_offset(int afmt)
-{
- switch (afmt)
- {
- case AFMT_MPA_L1:
- case AFMT_MPA_L2:
- case AFMT_MPA_L3:
- case AFMT_WAVPACK:
- /* Format may be loaded at the first needed frame */
- return true;
- default:
- /* Format must be loaded from the beginning of the file
- (does not imply 'atomic', while 'atomic' implies 'no offset') */
- return false;
- }
-}
-#endif /* CONFIG_CODEC == SWCODEC */
-
-
-/* Simple file type probing by looking at the filename extension. */
-unsigned int probe_file_format(const char *filename)
-{
- char *suffix;
- unsigned int i;
-
- suffix = strrchr(filename, '.');
-
- if (suffix == NULL)
- {
- return AFMT_UNKNOWN;
- }
-
- /* skip '.' */
- suffix++;
-
- for (i = 1; i < AFMT_NUM_CODECS; i++)
- {
- /* search extension list for type */
- const char *ext = audio_formats[i].ext_list;
-
- do
- {
- if (strcasecmp(suffix, ext) == 0)
- {
- return i;
- }
-
- ext += strlen(ext) + 1;
- }
- while (*ext != '\0');
- }
-
- return AFMT_UNKNOWN;
-}
-
-/* Note, that this returns false for successful, true for error! */
-bool mp3info(struct mp3entry *entry, const char *filename)
-{
- int fd;
- bool result;
-
- fd = open(filename, O_RDONLY);
- if (fd < 0)
- return true;
-
- result = !get_metadata(entry, fd, filename);
-
- close(fd);
-
- return result;
-}
-
-/* Get metadata for track - return false if parsing showed problems with the
- * file that would prevent playback.
- */
-bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
-{
- const struct afmt_entry *entry;
- int logfd = 0;
- DEBUGF("Read metadata for %s\n", trackname);
- if (write_metadata_log)
- {
- logfd = open("/metadata.log", O_WRONLY | O_APPEND | O_CREAT, 0666);
- if (logfd >= 0)
- {
- write(logfd, trackname, strlen(trackname));
- write(logfd, "\n", 1);
- close(logfd);
- }
- }
-
- /* Clear the mp3entry to avoid having bogus pointers appear */
- wipe_mp3entry(id3);
-
- /* Take our best guess at the codec type based on file extension */
- id3->codectype = probe_file_format(trackname);
-
- /* default values for embedded cuesheets */
- id3->has_embedded_cuesheet = false;
- id3->embedded_cuesheet.pos = 0;
-
- entry = &audio_formats[id3->codectype];
-
- /* Load codec specific track tag information and confirm the codec type. */
- if (!entry->parse_func)
- {
- DEBUGF("nothing to parse for %s (format %s)", trackname, entry->label);
- return false;
- }
-
- if (!entry->parse_func(fd, id3))
- {
- DEBUGF("parsing %s failed (format: %s)", trackname, entry->label);
- return false;
- }
-
- lseek(fd, 0, SEEK_SET);
- strlcpy(id3->path, trackname, sizeof(id3->path));
- /* We have successfully read the metadata from the file */
- return true;
-}
-
-#ifndef __PCTOOL__
-#if CONFIG_CODEC == SWCODEC
-void strip_tags(int handle_id)
-{
- static const unsigned char tag[] = "TAG";
- static const unsigned char apetag[] = "APETAGEX";
- size_t len, version;
- void *tail;
-
- if (bufgettail(handle_id, 128, &tail) != 128)
- return;
-
- if (memcmp(tail, tag, 3) == 0)
- {
- /* Skip id3v1 tag */
- logf("Cutting off ID3v1 tag");
- bufcuttail(handle_id, 128);
- }
-
- /* Get a new tail, as the old one may have been cut */
- if (bufgettail(handle_id, 32, &tail) != 32)
- return;
-
- /* Check for APE tag (look for the APE tag footer) */
- if (memcmp(tail, apetag, 8) != 0)
- return;
-
- /* Read the version and length from the footer */
- version = get_long_le(&((unsigned char *)tail)[8]);
- len = get_long_le(&((unsigned char *)tail)[12]);
- if (version == 2000)
- len += 32; /* APEv2 has a 32 byte header */
-
- /* Skip APE tag */
- logf("Cutting off APE tag (%ldB)", len);
- bufcuttail(handle_id, len);
-}
-#endif /* CONFIG_CODEC == SWCODEC */
-#endif /* ! __PCTOOL__ */
-
-#define MOVE_ENTRY(x) if (x) x += offset;
-
-void adjust_mp3entry(struct mp3entry *entry, void *dest, const void *orig)
-{
- long offset;
- if (orig > dest)
- offset = -((size_t)orig - (size_t)dest);
- else
- offset = ((size_t)dest - (size_t)orig);
-
- MOVE_ENTRY(entry->title)
- MOVE_ENTRY(entry->artist)
- MOVE_ENTRY(entry->album)
-
- if (entry->genre_string > (char*)orig &&
- entry->genre_string < (char*)orig + sizeof(struct mp3entry))
- /* Don't adjust that if it points to an entry of the "genres" array */
- entry->genre_string += offset;
-
- MOVE_ENTRY(entry->track_string)
- MOVE_ENTRY(entry->disc_string)
- MOVE_ENTRY(entry->year_string)
- MOVE_ENTRY(entry->composer)
- MOVE_ENTRY(entry->comment)
- MOVE_ENTRY(entry->albumartist)
- MOVE_ENTRY(entry->grouping)
- MOVE_ENTRY(entry->mb_track_id)
-}
-
-void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig)
-{
- memcpy(dest, orig, sizeof(struct mp3entry));
- adjust_mp3entry(dest, dest, orig);
-}
-
-/* A shortcut to simplify the common task of clearing the struct */
-void wipe_mp3entry(struct mp3entry *id3)
-{
- memset(id3, 0, sizeof (struct mp3entry));
-}
-
-#if CONFIG_CODEC == SWCODEC
-/* Glean what is possible from the filename alone - does not parse metadata */
-void fill_metadata_from_path(struct mp3entry *id3, const char *trackname)
-{
- char *p;
-
- /* Clear the mp3entry to avoid having bogus pointers appear */
- wipe_mp3entry(id3);
-
- /* Find the filename portion of the path */
- p = strrchr(trackname, '/');
- strlcpy(id3->id3v2buf, p ? ++p : id3->path, ID3V2_BUF_SIZE);
-
- /* Get the format from the extension and trim it off */
- p = strrchr(id3->id3v2buf, '.');
- if (p)
- {
- /* Might be wrong for container formats - should we bother? */
- id3->codectype = probe_file_format(p);
-
- if (id3->codectype != AFMT_UNKNOWN)
- *p = '\0';
- }
-
- /* Set the filename as the title */
- id3->title = id3->id3v2buf;
-
- /* Copy the path info */
- strlcpy(id3->path, trackname, sizeof (id3->path));
-}
-#endif /* CONFIG_CODEC == SWCODEC */
-
-#ifndef __PCTOOL__
-#ifdef HAVE_TAGCACHE
-#if CONFIG_CODEC == SWCODEC
-
-enum { AUTORESUMABLE_UNKNOWN = 0, AUTORESUMABLE_TRUE, AUTORESUMABLE_FALSE };
-
-bool autoresumable(struct mp3entry *id3)
-{
- char *endp, *path;
- size_t len;
- bool is_resumable;
-
- if (id3->autoresumable) /* result cached? */
- return id3->autoresumable == AUTORESUMABLE_TRUE;
-
- is_resumable = false;
-
- if (id3->path)
- {
- for (path = global_settings.autoresume_paths;
- *path; /* search terms left? */
- path++)
- {
- if (*path == ':') /* Skip empty search patterns */
- continue;
-
- /* FIXME: As soon as strcspn or strchrnul are made available in
- the core, the following can be made more efficient. */
- endp = strchr(path, ':');
- if (endp)
- len = endp - path;
- else
- len = strlen(path);
-
- /* Note: At this point, len is always > 0 */
-
- if (strncasecmp(id3->path, path, len) == 0)
- {
- /* Full directory-name matches only. Trailing '/' in
- search path OK. */
- if (id3->path[len] == '/' || id3->path[len - 1] == '/')
- {
- is_resumable = true;
- break;
- }
- }
- path += len - 1;
- }
- }
-
- /* cache result */
- id3->autoresumable =
- is_resumable ? AUTORESUMABLE_TRUE : AUTORESUMABLE_FALSE;
-
- logf("autoresumable: %s is%s resumable",
- id3->path, is_resumable ? "" : " not");
-
- return is_resumable;
-}
-
-#endif /* SWCODEC */
-#endif /* HAVE_TAGCACHE */
-#endif /* __PCTOOL__ */
diff --git a/apps/metadata.h b/apps/metadata.h
deleted file mode 100644
index 55e4d76f25..0000000000
--- a/apps/metadata.h
+++ /dev/null
@@ -1,353 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _METADATA_H
-#define _METADATA_H
-
-#include <stdbool.h>
-#include "config.h"
-#include "file.h"
-
-
-/* Audio file types. */
-/* NOTE: The values of the AFMT_* items are used for the %fc tag in the WPS
- - so new entries MUST be added to the end to maintain compatibility.
- */
-enum
-{
- AFMT_UNKNOWN = 0, /* Unknown file format */
-
- /* start formats */
- AFMT_MPA_L1, /* MPEG Audio layer 1 */
- AFMT_MPA_L2, /* MPEG Audio layer 2 */
- AFMT_MPA_L3, /* MPEG Audio layer 3 */
-
-#if CONFIG_CODEC == SWCODEC
- AFMT_AIFF, /* Audio Interchange File Format */
- AFMT_PCM_WAV, /* Uncompressed PCM in a WAV file */
- AFMT_OGG_VORBIS, /* Ogg Vorbis */
- AFMT_FLAC, /* FLAC */
- AFMT_MPC_SV7, /* Musepack SV7 */
- AFMT_A52, /* A/52 (aka AC3) audio */
- AFMT_WAVPACK, /* WavPack */
- AFMT_MP4_ALAC, /* Apple Lossless Audio Codec */
- AFMT_MP4_AAC, /* Advanced Audio Coding (AAC) in M4A container */
- AFMT_SHN, /* Shorten */
- AFMT_SID, /* SID File Format */
- AFMT_ADX, /* ADX File Format */
- AFMT_NSF, /* NESM (NES Sound Format) */
- AFMT_SPEEX, /* Ogg Speex speech */
- AFMT_SPC, /* SPC700 save state */
- AFMT_APE, /* Monkey's Audio (APE) */
- AFMT_WMA, /* WMAV1/V2 in ASF */
- AFMT_WMAPRO, /* WMA Professional in ASF */
- AFMT_MOD, /* Amiga MOD File Format */
- AFMT_SAP, /* Atari 8Bit SAP Format */
- AFMT_RM_COOK, /* Cook in RM/RA */
- AFMT_RM_AAC, /* AAC in RM/RA */
- AFMT_RM_AC3, /* AC3 in RM/RA */
- AFMT_RM_ATRAC3, /* ATRAC3 in RM/RA */
- AFMT_CMC, /* Atari 8bit cmc format */
- AFMT_CM3, /* Atari 8bit cm3 format */
- AFMT_CMR, /* Atari 8bit cmr format */
- AFMT_CMS, /* Atari 8bit cms format */
- AFMT_DMC, /* Atari 8bit dmc format */
- AFMT_DLT, /* Atari 8bit dlt format */
- AFMT_MPT, /* Atari 8bit mpt format */
- AFMT_MPD, /* Atari 8bit mpd format */
- AFMT_RMT, /* Atari 8bit rmt format */
- AFMT_TMC, /* Atari 8bit tmc format */
- AFMT_TM8, /* Atari 8bit tm8 format */
- AFMT_TM2, /* Atari 8bit tm2 format */
- AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */
- AFMT_SMAF, /* SMAF */
- AFMT_AU, /* Sun Audio file */
- AFMT_VOX, /* VOX */
- AFMT_WAVE64, /* Wave64 */
- AFMT_TTA, /* True Audio */
- AFMT_WMAVOICE, /* WMA Voice in ASF */
- AFMT_MPC_SV8, /* Musepack SV8 */
- AFMT_MP4_AAC_HE, /* Advanced Audio Coding (AAC-HE) in M4A container */
- AFMT_AY, /* AY (ZX Spectrum, Amstrad CPC Sound Format) */
- AFMT_GBS, /* GBS (Game Boy Sound Format) */
- AFMT_HES, /* HES (Hudson Entertainment System Sound Format) */
- AFMT_SGC, /* SGC (Sega Master System, Game Gear, Coleco Vision Sound Format) */
- AFMT_VGM, /* VGM (Video Game Music Format) */
- AFMT_KSS, /* KSS (MSX computer KSS Music File) */
-#endif
-
- /* add new formats at any index above this line to have a sensible order -
- specified array index inits are used */
- /* format arrays defined in id3.c */
-
- AFMT_NUM_CODECS,
-
-#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING)
- /* masks to decompose parts */
- CODEC_AFMT_MASK = 0x0fff,
- CODEC_TYPE_MASK = 0x7000,
-
- /* switch for specifying codec type when requesting a filename */
- CODEC_TYPE_DECODER = (0 << 12), /* default */
- CODEC_TYPE_ENCODER = (1 << 12),
-#endif /* CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) */
-};
-
-#if CONFIG_CODEC == SWCODEC
-#if (CONFIG_PLATFORM & PLATFORM_ANDROID)
-#define CODEC_EXTENSION "so"
-#define CODEC_PREFIX "lib"
-#else
-#define CODEC_EXTENSION "codec"
-#define CODEC_PREFIX ""
-#endif
-
-#ifdef HAVE_RECORDING
-enum rec_format_indexes
-{
- __REC_FORMAT_START_INDEX = -1,
-
- /* start formats */
-
- REC_FORMAT_PCM_WAV,
- REC_FORMAT_AIFF,
- REC_FORMAT_WAVPACK,
- REC_FORMAT_MPA_L3,
-
- /* add new formats at any index above this line to have a sensible order -
- specified array index inits are used
- REC_FORMAT_CFG_NUM_BITS should allocate enough bits to hold the range
- REC_FORMAT_CFG_VALUE_LIST should be in same order as indexes
- */
-
- REC_NUM_FORMATS,
-
- REC_FORMAT_DEFAULT = REC_FORMAT_PCM_WAV,
- REC_FORMAT_CFG_NUM_BITS = 2
-};
-
-#define REC_FORMAT_CFG_VAL_LIST "wave,aiff,wvpk,mpa3"
-
-/* get REC_FORMAT_* corresponding AFMT_* */
-extern const int rec_format_afmt[REC_NUM_FORMATS];
-/* get AFMT_* corresponding REC_FORMAT_* */
-/* unused: extern const int afmt_rec_format[AFMT_NUM_CODECS]; */
-
-#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
- { label, root_fname, enc_root_fname, func, ext_list }
-#else /* !HAVE_RECORDING */
-#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
- { label, root_fname, func, ext_list }
-#endif /* HAVE_RECORDING */
-
-#else /* !SWCODEC */
-
-#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
- { label, func, ext_list }
-#endif /* CONFIG_CODEC == SWCODEC */
-
-/** Database of audio formats **/
-/* record describing the audio format */
-struct mp3entry;
-struct afmt_entry
-{
- const char *label; /* format label */
-#if CONFIG_CODEC == SWCODEC
- const char *codec_root_fn; /* root codec filename (sans _enc and .codec) */
-#ifdef HAVE_RECORDING
- const char *codec_enc_root_fn; /* filename of encoder codec */
-#endif
-#endif
- bool (*parse_func)(int fd, struct mp3entry *id3); /* return true on success */
- const char *ext_list; /* NULL terminated extension
- list for type with the first as
- the default for recording */
-};
-
-/* database of labels and codecs. add formats per above enum */
-extern const struct afmt_entry audio_formats[AFMT_NUM_CODECS];
-
-#if MEMORYSIZE > 2
-#define ID3V2_BUF_SIZE 900
-#define ID3V2_MAX_ITEM_SIZE 240
-#else
-#define ID3V2_BUF_SIZE 300
-#define ID3V2_MAX_ITEM_SIZE 90
-#endif
-
-enum {
- ID3_VER_1_0 = 1,
- ID3_VER_1_1,
- ID3_VER_2_2,
- ID3_VER_2_3,
- ID3_VER_2_4
-};
-
-#ifdef HAVE_ALBUMART
-enum mp3_aa_type {
- AA_TYPE_UNSYNC = -1,
- AA_TYPE_UNKNOWN,
- AA_TYPE_BMP,
- AA_TYPE_PNG,
- AA_TYPE_JPG,
-};
-
-struct mp3_albumart {
- enum mp3_aa_type type;
- int size;
- off_t pos;
-};
-#endif
-
-enum character_encoding {
- CHAR_ENC_ISO_8859_1 = 1,
- CHAR_ENC_UTF_8,
- CHAR_ENC_UTF_16_LE,
- CHAR_ENC_UTF_16_BE,
-};
-
-/* cache embedded cuesheet details */
-struct embedded_cuesheet {
- int size;
- off_t pos;
- enum character_encoding encoding;
-};
-
-struct mp3entry {
- char path[MAX_PATH];
- char* title;
- char* artist;
- char* album;
- char* genre_string;
- char* disc_string;
- char* track_string;
- char* year_string;
- char* composer;
- char* comment;
- char* albumartist;
- char* grouping;
- int discnum;
- int tracknum;
- int layer;
- int year;
- unsigned char id3version;
- unsigned int codectype;
- unsigned int bitrate;
- unsigned long frequency;
- unsigned long id3v2len;
- unsigned long id3v1len;
- unsigned long first_frame_offset; /* Byte offset to first real MP3 frame.
- Used for skipping leading garbage to
- avoid gaps between tracks. */
- unsigned long filesize; /* without headers; in bytes */
- unsigned long length; /* song length in ms */
- unsigned long elapsed; /* ms played */
-
- int lead_trim; /* Number of samples to skip at the beginning */
- int tail_trim; /* Number of samples to remove from the end */
-
- /* Added for Vorbis, used by mp4 parser as well. */
- unsigned long samples; /* number of samples in track */
-
- /* MP3 stream specific info */
- unsigned long frame_count; /* number of frames in the file (if VBR) */
-
- /* Used for A52/AC3 */
- unsigned long bytesperframe; /* number of bytes per frame (if CBR) */
-
- /* Xing VBR fields */
- bool vbr;
- bool has_toc; /* True if there is a VBR header in the file */
- unsigned char toc[100]; /* table of contents */
-
- /* Added for ATRAC3 */
- unsigned int channels; /* Number of channels in the stream */
- unsigned int extradata_size; /* Size (in bytes) of the codec's extradata from the container */
-
- /* Added for AAC HE SBR */
- bool needs_upsampling_correction; /* flag used by aac codec */
-
- /* these following two fields are used for local buffering */
- char id3v2buf[ID3V2_BUF_SIZE];
- char id3v1buf[4][92];
-
- /* resume related */
- unsigned long offset; /* bytes played */
- int index; /* playlist index */
-
-#ifdef HAVE_TAGCACHE
- unsigned char autoresumable; /* caches result of autoresumable() */
-
- /* runtime database fields */
- long tagcache_idx; /* 0=invalid, otherwise idx+1 */
- int rating;
- int score;
- long playcount;
- long lastplayed;
- long playtime;
-#endif
-
- /* replaygain support */
-#if CONFIG_CODEC == SWCODEC
- long track_level; /* holds the level in dB * (1<<FP_BITS) */
- long album_level;
- long track_gain; /* s19.12 signed fixed point. 0 for no gain. */
- long album_gain;
- long track_peak; /* s19.12 signed fixed point. 0 for no peak. */
- long album_peak;
-#endif
-
-#ifdef HAVE_ALBUMART
- bool has_embedded_albumart;
- struct mp3_albumart albumart;
-#endif
-
- /* Cuesheet support */
- bool has_embedded_cuesheet;
- struct embedded_cuesheet embedded_cuesheet;
- struct cuesheet *cuesheet;
-
- /* Musicbrainz Track ID */
- char* mb_track_id;
-};
-
-unsigned int probe_file_format(const char *filename);
-bool get_metadata(struct mp3entry* id3, int fd, const char* trackname);
-bool mp3info(struct mp3entry *entry, const char *filename);
-void adjust_mp3entry(struct mp3entry *entry, void *dest, const void *orig);
-void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig);
-void wipe_mp3entry(struct mp3entry *id3);
-
-#if CONFIG_CODEC == SWCODEC
-void fill_metadata_from_path(struct mp3entry *id3, const char *trackname);
-int get_audio_base_codec_type(int type);
-void strip_tags(int handle_id);
-enum data_type get_audio_base_data_type(int afmt);
-bool format_buffers_with_offset(int afmt);
-#endif
-
-#ifdef HAVE_TAGCACHE
-bool autoresumable(struct mp3entry *id3);
-#endif
-
-#endif
-
-
diff --git a/apps/metadata/a52.c b/apps/metadata/a52.c
deleted file mode 100644
index a8aad3fa4f..0000000000
--- a/apps/metadata/a52.c
+++ /dev/null
@@ -1,103 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <stdio.h>
-#include "metadata.h"
-#include "logf.h"
-
-#include "metadata_parsers.h"
-
-static const unsigned short a52_bitrates[] =
-{
- 32, 40, 48, 56, 64, 80, 96, 112, 128, 160,
- 192, 224, 256, 320, 384, 448, 512, 576, 640
-};
-
-/* Only store frame sizes for 44.1KHz - others are simply multiples
- of the bitrate */
-static const unsigned short a52_441framesizes[] =
-{
- 69 * 2, 70 * 2, 87 * 2, 88 * 2, 104 * 2, 105 * 2, 121 * 2,
- 122 * 2, 139 * 2, 140 * 2, 174 * 2, 175 * 2, 208 * 2, 209 * 2,
- 243 * 2, 244 * 2, 278 * 2, 279 * 2, 348 * 2, 349 * 2, 417 * 2,
- 418 * 2, 487 * 2, 488 * 2, 557 * 2, 558 * 2, 696 * 2, 697 * 2,
- 835 * 2, 836 * 2, 975 * 2, 976 * 2, 1114 * 2, 1115 * 2, 1253 * 2,
- 1254 * 2, 1393 * 2, 1394 * 2
-};
-
-bool get_a52_metadata(int fd, struct mp3entry *id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- unsigned long totalsamples;
- int i;
-
- if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 5) < 5))
- {
- return false;
- }
-
- if ((buf[0] != 0x0b) || (buf[1] != 0x77))
- {
- logf("not an A52/AC3 file\n");
- return false;
- }
-
- i = buf[4] & 0x3e;
-
- if (i > 36)
- {
- logf("A52: Invalid frmsizecod: %d\n",i);
- return false;
- }
-
- id3->bitrate = a52_bitrates[i >> 1];
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- switch (buf[4] & 0xc0)
- {
- case 0x00:
- id3->frequency = 48000;
- id3->bytesperframe=id3->bitrate * 2 * 2;
- break;
-
- case 0x40:
- id3->frequency = 44100;
- id3->bytesperframe = a52_441framesizes[i];
- break;
-
- case 0x80:
- id3->frequency = 32000;
- id3->bytesperframe = id3->bitrate * 3 * 2;
- break;
-
- default:
- logf("A52: Invalid samplerate code: 0x%02x\n", buf[4] & 0xc0);
- return false;
- break;
- }
-
- /* One A52 frame contains 6 blocks, each containing 256 samples */
- totalsamples = id3->filesize / id3->bytesperframe * 6 * 256;
- id3->length = totalsamples / id3->frequency * 1000;
- return true;
-}
diff --git a/apps/metadata/adx.c b/apps/metadata/adx.c
deleted file mode 100644
index 7c341b4835..0000000000
--- a/apps/metadata/adx.c
+++ /dev/null
@@ -1,124 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "debug.h"
-
-bool get_adx_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char * buf = (unsigned char *)id3->path;
- int chanstart, channels;
- int looping = 0, start_adr = 0, end_adr = 0;
-
- /* try to get the basic header */
- if ((lseek(fd, 0, SEEK_SET) < 0)
- || (read(fd, buf, 0x38) < 0x38))
- {
- DEBUGF("lseek or read failed\n");
- return false;
- }
-
- /* ADX starts with 0x80 */
- if (buf[0] != 0x80) {
- DEBUGF("get_adx_metadata: wrong first byte %c\n",buf[0]);
- return false;
- }
-
- /* check for a reasonable offset */
- chanstart = ((buf[2] << 8) | buf[3]) + 4;
- if (chanstart > 4096) {
- DEBUGF("get_adx_metadata: bad chanstart %i\n", chanstart);
- return false;
- }
-
- /* check for a workable number of channels */
- channels = buf[7];
- if (channels != 1 && channels != 2) {
- DEBUGF("get_adx_metadata: bad channel count %i\n",channels);
- return false;
- }
-
- id3->frequency = get_long_be(&buf[8]);
- /* 32 samples per 18 bytes */
- id3->bitrate = id3->frequency * channels * 18 * 8 / 32 / 1000;
- id3->length = get_long_be(&buf[12]) / id3->frequency * 1000;
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- /* get loop info */
- if (!memcmp(buf+0x10,"\x01\xF4\x03",3)) {
- /* Soul Calibur 2 style (type 03) */
- DEBUGF("get_adx_metadata: type 03 found\n");
- /* check if header is too small for loop data */
- if (chanstart-6 < 0x2c) looping=0;
- else {
- looping = get_long_be(&buf[0x18]);
- end_adr = get_long_be(&buf[0x28]);
- start_adr = get_long_be(&buf[0x1c])/32*channels*18+chanstart;
- }
- } else if (!memcmp(buf+0x10,"\x01\xF4\x04",3)) {
- /* Standard (type 04) */
- DEBUGF("get_adx_metadata: type 04 found\n");
- /* check if header is too small for loop data */
- if (chanstart-6 < 0x38) looping=0;
- else {
- looping = get_long_be(&buf[0x24]);
- end_adr = get_long_be(&buf[0x34]);
- start_adr = get_long_be(&buf[0x28])/32*channels*18+chanstart;
- }
- } else {
- DEBUGF("get_adx_metadata: error, couldn't determine ADX type\n");
- return false;
- }
-
- /* is file using encryption */
- if (buf[0x13]==0x08) {
- DEBUGF("get_adx_metadata: error, encrypted ADX not supported\n");
- return false;
- }
-
- if (looping) {
- /* 2 loops, 10 second fade */
- id3->length = (start_adr-chanstart + 2*(end_adr-start_adr))
- *8 / id3->bitrate + 10000;
- }
-
- /* try to get the channel header */
- if ((lseek(fd, chanstart-6, SEEK_SET) < 0)
- || (read(fd, buf, 6) < 6))
- {
- return false;
- }
-
- /* check channel header */
- if (memcmp(buf, "(c)CRI", 6) != 0) return false;
-
- return true;
-}
diff --git a/apps/metadata/aiff.c b/apps/metadata/aiff.c
deleted file mode 100644
index 654f37cf98..0000000000
--- a/apps/metadata/aiff.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-
-#include "debug.h"
-
-/* compressionType: AIFC QuickTime IMA ADPCM */
-#define AIFC_FORMAT_QT_IMA_ADPCM "ima4"
-
-bool get_aiff_metadata(int fd, struct mp3entry* id3)
-{
- unsigned char buf[512];
- unsigned long numChannels = 0;
- unsigned long numSampleFrames = 0;
- unsigned long numbytes = 0;
- bool is_aifc = false;
-
- if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, &buf[0], 12) < 12) ||
- (memcmp(&buf[0], "FORM", 4) != 0) || (memcmp(&buf[8], "AIF", 3) != 0) ||
- (!(is_aifc = (buf[11] == 'C')) && buf[11] != 'F'))
- {
- return false;
- }
-
- while (read(fd, &buf[0], 8) == 8)
- {
- size_t size = get_long_be(&buf[4]); /* chunkSize */
-
- if (memcmp(&buf[0], "SSND", 4) == 0)
- {
- numbytes = size - 8;
- break; /* assume COMM was already read */
- }
-
- /* odd chunk sizes must be padded */
- size += size & 1;
-
- if (size > sizeof(buf))
- {
- DEBUGF("AIFF \"%4.4s\" chunk too large (%zd > %zd)",
- (char*) &buf[0], size, sizeof(buf));
- }
-
- if (memcmp(&buf[0], "COMM", 4) == 0)
- {
- if (size > sizeof(buf) || read(fd, &buf[0], size) != (ssize_t)size)
- return false;
-
- numChannels = ((buf[0]<<8)|buf[1]);
-
- numSampleFrames = get_long_be(&buf[2]);
-
- /* sampleRate */
- id3->frequency = get_long_be(&buf[10]);
- id3->frequency >>= (16+14-buf[9]);
-
- /* save format infos */
- id3->bitrate = ((buf[6]<<8)|buf[7]) * numChannels * id3->frequency;
- id3->bitrate /= 1000;
-
- if (!is_aifc || memcmp(&buf[18], AIFC_FORMAT_QT_IMA_ADPCM, 4) != 0)
- id3->length = ((int64_t) numSampleFrames * 1000) / id3->frequency;
- else
- {
- /* QuickTime IMA ADPCM is 1block = 64 data for each channel */
- id3->length = ((int64_t) numSampleFrames * 64000LL) / id3->frequency;
- }
-
- id3->vbr = false; /* AIFF files are CBR */
- id3->filesize = filesize(fd);
- }
- else
- {
- /* skip chunk */
- if (lseek(fd, size, SEEK_CUR) < 0)
- return false;
- }
- }
-
- return numbytes && numChannels;
-}
diff --git a/apps/metadata/ape.c b/apps/metadata/ape.c
deleted file mode 100644
index 0bd2477431..0000000000
--- a/apps/metadata/ape.c
+++ /dev/null
@@ -1,182 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "structec.h"
-
-#define APETAG_HEADER_LENGTH 32
-#define APETAG_HEADER_FORMAT "8llll8"
-#define APETAG_ITEM_HEADER_FORMAT "ll"
-#define APETAG_ITEM_TYPE_MASK 3
-
-#ifdef HAVE_ALBUMART
-/* The AA header consists of the pseudo filename "Album Cover (Front).ext"
- * whereas ".ext" is the file extension. For now ".jpg" and ".png" are
- * supported by this APE metadata parser. Therefore the length is 22. */
-#define APETAG_AA_HEADER_LENGTH 22
-#endif
-
-struct apetag_header
-{
- char id[8];
- long version;
- long length;
- long item_count;
- long flags;
- char reserved[8];
-};
-
-struct apetag_item_header
-{
- long length;
- long flags;
-};
-
-/* Read the items in an APEV2 tag. Only looks for a tag at the end of a
- * file. Returns true if a tag was found and fully read, false otherwise.
- */
-bool read_ape_tags(int fd, struct mp3entry* id3)
-{
- struct apetag_header header;
-
- if ((lseek(fd, -APETAG_HEADER_LENGTH, SEEK_END) < 0)
- || (ecread(fd, &header, 1, APETAG_HEADER_FORMAT, IS_BIG_ENDIAN)
- != APETAG_HEADER_LENGTH)
- || (memcmp(header.id, "APETAGEX", sizeof(header.id))))
- {
- return false;
- }
-
- if ((header.version == 2000) && (header.item_count > 0)
- && (header.length > APETAG_HEADER_LENGTH))
- {
- char *buf = id3->id3v2buf;
- unsigned int buf_remaining = sizeof(id3->id3v2buf)
- + sizeof(id3->id3v1buf);
- unsigned int tag_remaining = header.length - APETAG_HEADER_LENGTH;
- int i;
-
- if (lseek(fd, -header.length, SEEK_END) < 0)
- {
- return false;
- }
-
- for (i = 0; i < header.item_count; i++)
- {
- struct apetag_item_header item;
- char name[TAG_NAME_LENGTH];
- char value[TAG_VALUE_LENGTH];
- long r;
-
- if (tag_remaining < sizeof(item))
- {
- break;
- }
-
- if (ecread(fd, &item, 1, APETAG_ITEM_HEADER_FORMAT, IS_BIG_ENDIAN)
- < (long) sizeof(item))
- {
- return false;
- }
-
- tag_remaining -= sizeof(item);
- r = read_string(fd, name, sizeof(name), 0, tag_remaining);
-
- if (r == -1)
- {
- return false;
- }
-
- tag_remaining -= r + item.length;
-
- if ((item.flags & APETAG_ITEM_TYPE_MASK) == 0)
- {
- long len;
-
- if (read_string(fd, value, sizeof(value), -1, item.length)
- != item.length)
- {
- return false;
- }
-
- len = parse_tag(name, value, id3, buf, buf_remaining,
- TAGTYPE_APE);
- buf += len;
- buf_remaining -= len;
- }
- else
- {
-#ifdef HAVE_ALBUMART
- if (strcasecmp(name, "cover art (front)") == 0)
- {
- /* Allow to read at least APETAG_AA_HEADER_LENGTH bytes. */
- r = read_string(fd, name, sizeof(name), 0, APETAG_AA_HEADER_LENGTH);
- if (r == -1)
- {
- return false;
- }
-
- /* Gather the album art format from the pseudo file name's ending. */
- strcpy(name, name + strlen(name) - 4);
- id3->albumart.type = AA_TYPE_UNKNOWN;
- if (strcasecmp(name, ".jpg") == 0)
- {
- id3->albumart.type = AA_TYPE_JPG;
- }
- else if (strcasecmp(name, ".png") == 0)
- {
- id3->albumart.type = AA_TYPE_PNG;
- }
-
- /* Set the album art size and position. */
- if (id3->albumart.type != AA_TYPE_UNKNOWN)
- {
- id3->albumart.pos = lseek(fd, 0, SEEK_CUR);
- id3->albumart.size = item.length - r;
- id3->has_embedded_albumart = true;
- }
-
- /* Seek back to this APE items begin. */
- if (lseek(fd, -r, SEEK_CUR) < 0)
- {
- return false;
- }
- }
-#endif
- /* Seek to the next APE item. */
- if (lseek(fd, item.length, SEEK_CUR) < 0)
- {
- return false;
- }
- }
- }
- }
-
- return true;
-}
diff --git a/apps/metadata/asap.c b/apps/metadata/asap.c
deleted file mode 100644
index 9e7f227031..0000000000
--- a/apps/metadata/asap.c
+++ /dev/null
@@ -1,254 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2008 Dominik Wenger
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-#include "debug.h"
-
-#define MAX_SONGS 32
-
-static bool parse_dec(int *retval, const char *p, int minval, int maxval)
-{
- int r = 0;
- do {
- char c = *p;
- if (c >= '0' && c <= '9')
- r = 10 * r + c - '0';
- else
- return false;
- if (r > maxval)
- return false;
- } while (*++p != '\0');
- if (r < minval)
- return false;
- *retval = r;
- return true;
-}
-
-static bool parse_text(char *retval, const char *p)
-{
- int i;
- if (*p != '"')
- return false;
- p++;
- if (p[0] == '<' && p[1] == '?' && p[2] == '>' && p[3] == '"')
- return true;
- i = 0;
- while (*p != '"') {
- if (i >= 127)
- return false;
- if (*p == '\0')
- return false;
- retval[i++] = *p++;
- }
- retval[i] = '\0';
- return true;
-}
-
-static int ASAP_ParseDuration(const char *s)
-{
- int r;
- if (*s < '0' || *s > '9')
- return -1;
- r = *s++ - '0';
- if (*s >= '0' && *s <= '9')
- r = 10 * r + *s++ - '0';
- if (*s == ':') {
- s++;
- if (*s < '0' || *s > '5')
- return -1;
- r = 60 * r + (*s++ - '0') * 10;
- if (*s < '0' || *s > '9')
- return -1;
- r += *s++ - '0';
- }
- r *= 1000;
- if (*s != '.')
- return r;
- s++;
- if (*s < '0' || *s > '9')
- return r;
- r += 100 * (*s++ - '0');
- if (*s < '0' || *s > '9')
- return r;
- r += 10 * (*s++ - '0');
- if (*s < '0' || *s > '9')
- return r;
- r += *s - '0';
- return r;
-}
-
-static bool read_asap_string(char* source, char** buf, char** buffer_end, char** dest)
-{
- if(parse_text(*buf,source) == false)
- return false;
-
- /* set dest pointer */
- *dest = *buf;
-
- /* move buf ptr */
- *buf += strlen(*buf)+1;
-
- /* check size */
- if(*buf >= *buffer_end)
- {
- DEBUGF("Buffer full\n");
- return false;
- }
- return true;
-}
-
-static bool parse_sap_header(int fd, struct mp3entry* id3, int file_len)
-{
- int module_index = 0;
- int sap_signature = -1;
- int duration_index = 0;
- unsigned char cur_char = 0;
- int i;
-
- /* set defaults */
- int numSongs = 1;
- int defSong = 0;
- int durations[MAX_SONGS];
- for (i = 0; i < MAX_SONGS; i++)
- durations[i] = -1;
-
- /* use id3v2 buffer for our strings */
- char* buffer = id3->id3v2buf;
- char* buffer_end = id3->id3v2buf + ID3V2_BUF_SIZE;
-
- /* parse file */
- while (1)
- {
- char line[256];
- char *p;
-
- if (module_index + 8 >= file_len)
- return false;
- /* read a char */
- read(fd,&cur_char,1);
- /* end of header */
- if (cur_char == 0xff)
- break;
-
- i = 0;
- while (cur_char != 0x0d)
- {
- line[i++] = cur_char;
- module_index++;
- if (module_index >= file_len || (unsigned)i >= sizeof(line) - 1)
- return false;
- /* read a char */
- read(fd,&cur_char,1);
- }
- if (++module_index >= file_len )
- return false;
- /* read a char */
- read(fd,&cur_char,1);
- if ( cur_char != 0x0a)
- return false;
-
- line[i] = '\0';
- for (p = line; *p != '\0'; p++) {
- if (*p == ' ') {
- *p++ = '\0';
- break;
- }
- }
-
- /* parse tags */
- if(strcmp(line, "SAP") == 0)
- sap_signature = 1;
- if (sap_signature == -1)
- return false;
- if (strcmp(line, "AUTHOR") == 0)
- {
- if(read_asap_string(p, &buffer, &buffer_end, &id3->artist) == false)
- return false;
- }
- else if(strcmp(line, "NAME") == 0)
- {
- if(read_asap_string(p, &buffer, &buffer_end, &id3->title) == false)
- return false;
- }
- else if(strcmp(line, "DATE") == 0)
- {
- if(read_asap_string(p, &buffer, &buffer_end, &id3->year_string) == false)
- return false;
- }
- else if (strcmp(line, "SONGS") == 0)
- {
- if (parse_dec(&numSongs, p, 1, MAX_SONGS) == false )
- return false;
- }
- else if (strcmp(line, "DEFSONG") == 0)
- {
- if (parse_dec(&defSong, p, 0, MAX_SONGS) == false)
- return false;
- }
- else if (strcmp(line, "TIME") == 0)
- {
- int durationTemp = ASAP_ParseDuration(p);
- if (durationTemp < 0 || duration_index >= MAX_SONGS)
- return false;
- durations[duration_index++] = durationTemp;
- }
- }
-
- /* set length: */
- int length = durations[defSong];
- if (length < 0)
- length = 180 * 1000;
- id3->length = length;
-
- lseek(fd, 0, SEEK_SET);
- return true;
-}
-
-
-bool get_asap_metadata(int fd, struct mp3entry* id3)
-{
-
- int filelength = filesize(fd);
-
- if(parse_sap_header(fd, id3, filelength) == false)
- {
- DEBUGF("parse sap header failed.\n");
- return false;
- }
-
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- id3->vbr = false;
- id3->filesize = filelength;
- id3->genre_string = id3_get_num_genre(36);
-
- return true;
-}
diff --git a/apps/metadata/asf.c b/apps/metadata/asf.c
deleted file mode 100644
index b815c09769..0000000000
--- a/apps/metadata/asf.c
+++ /dev/null
@@ -1,591 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- *
- * $Id$
- *
- * Copyright (C) 2007 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "metadata.h"
-#include "replaygain.h"
-#include "debug.h"
-#include "rbunicode.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "system.h"
-#include <codecs/libasf/asf.h>
-
-/* TODO: Just read the GUIDs into a 16-byte array, and use memcmp to compare */
-struct guid_s {
- uint32_t v1;
- uint16_t v2;
- uint16_t v3;
- uint8_t v4[8];
-};
-typedef struct guid_s guid_t;
-
-struct asf_object_s {
- guid_t guid;
- uint64_t size;
- uint64_t datalen;
-};
-typedef struct asf_object_s asf_object_t;
-
-static const guid_t asf_guid_null =
-{0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}};
-
-/* top level object guids */
-
-static const guid_t asf_guid_header =
-{0x75B22630, 0x668E, 0x11CF, {0xA6, 0xD9, 0x00, 0xAA, 0x00, 0x62, 0xCE, 0x6C}};
-
-static const guid_t asf_guid_data =
-{0x75B22636, 0x668E, 0x11CF, {0xA6, 0xD9, 0x00, 0xAA, 0x00, 0x62, 0xCE, 0x6C}};
-
-static const guid_t asf_guid_index =
-{0x33000890, 0xE5B1, 0x11CF, {0x89, 0xF4, 0x00, 0xA0, 0xC9, 0x03, 0x49, 0xCB}};
-
-/* header level object guids */
-
-static const guid_t asf_guid_file_properties =
-{0x8cabdca1, 0xa947, 0x11cf, {0x8E, 0xe4, 0x00, 0xC0, 0x0C, 0x20, 0x53, 0x65}};
-
-static const guid_t asf_guid_stream_properties =
-{0xB7DC0791, 0xA9B7, 0x11CF, {0x8E, 0xE6, 0x00, 0xC0, 0x0C, 0x20, 0x53, 0x65}};
-
-static const guid_t asf_guid_content_description =
-{0x75B22633, 0x668E, 0x11CF, {0xA6, 0xD9, 0x00, 0xAA, 0x00, 0x62, 0xCE, 0x6C}};
-
-static const guid_t asf_guid_extended_content_description =
-{0xD2D0A440, 0xE307, 0x11D2, {0x97, 0xF0, 0x00, 0xA0, 0xC9, 0x5E, 0xA8, 0x50}};
-
-static const guid_t asf_guid_content_encryption =
-{0x2211b3fb, 0xbd23, 0x11d2, {0xb4, 0xb7, 0x00, 0xa0, 0xc9, 0x55, 0xfc, 0x6e}};
-
-static const guid_t asf_guid_extended_content_encryption =
-{0x298ae614, 0x2622, 0x4c17, {0xb9, 0x35, 0xda, 0xe0, 0x7e, 0xe9, 0x28, 0x9c}};
-
-/* stream type guids */
-
-static const guid_t asf_guid_stream_type_audio =
-{0xF8699E40, 0x5B4D, 0x11CF, {0xA8, 0xFD, 0x00, 0x80, 0x5F, 0x5C, 0x44, 0x2B}};
-
-static int asf_guid_match(const guid_t *guid1, const guid_t *guid2)
-{
- if((guid1->v1 != guid2->v1) ||
- (guid1->v2 != guid2->v2) ||
- (guid1->v3 != guid2->v3) ||
- (memcmp(guid1->v4, guid2->v4, 8))) {
- return 0;
- }
-
- return 1;
-}
-
-/* Read the 16 byte GUID from a file */
-static void asf_readGUID(int fd, guid_t* guid)
-{
- read_uint32le(fd, &guid->v1);
- read_uint16le(fd, &guid->v2);
- read_uint16le(fd, &guid->v3);
- read(fd, guid->v4, 8);
-}
-
-static void asf_read_object_header(asf_object_t *obj, int fd)
-{
- asf_readGUID(fd, &obj->guid);
- read_uint64le(fd, &obj->size);
- obj->datalen = 0;
-}
-
-/* Parse an integer from the extended content object - we always
- convert to an int, regardless of native format.
-*/
-static int asf_intdecode(int fd, int type, int length)
-{
- uint16_t tmp16;
- uint32_t tmp32;
- uint64_t tmp64;
-
- if (type == 3) {
- read_uint32le(fd, &tmp32);
- lseek(fd,length - 4,SEEK_CUR);
- return (int)tmp32;
- } else if (type == 4) {
- read_uint64le(fd, &tmp64);
- lseek(fd,length - 8,SEEK_CUR);
- return (int)tmp64;
- } else if (type == 5) {
- read_uint16le(fd, &tmp16);
- lseek(fd,length - 2,SEEK_CUR);
- return (int)tmp16;
- }
-
- return 0;
-}
-
-/* Decode a LE utf16 string from a disk buffer into a fixed-sized
- utf8 buffer.
-*/
-
-static void asf_utf16LEdecode(int fd,
- uint16_t utf16bytes,
- unsigned char **utf8,
- int* utf8bytes
- )
-{
- unsigned long ucs;
- int n;
- unsigned char utf16buf[256];
- unsigned char* utf16 = utf16buf;
- unsigned char* newutf8;
-
- n = read(fd, utf16buf, MIN(sizeof(utf16buf), utf16bytes));
- utf16bytes -= n;
-
- while (n > 0) {
- /* Check for a surrogate pair */
- if (utf16[1] >= 0xD8 && utf16[1] < 0xE0) {
- if (n < 4) {
- /* Run out of utf16 bytes, read some more */
- utf16buf[0] = utf16[0];
- utf16buf[1] = utf16[1];
-
- n = read(fd, utf16buf + 2, MIN(sizeof(utf16buf)-2, utf16bytes));
- utf16 = utf16buf;
- utf16bytes -= n;
- n += 2;
- }
-
- if (n < 4) {
- /* Truncated utf16 string, abort */
- break;
- }
- ucs = 0x10000 + ((utf16[0] << 10) | ((utf16[1] - 0xD8) << 18)
- | utf16[2] | ((utf16[3] - 0xDC) << 8));
- utf16 += 4;
- n -= 4;
- } else {
- ucs = (utf16[0] | (utf16[1] << 8));
- utf16 += 2;
- n -= 2;
- }
-
- if (*utf8bytes > 6) {
- newutf8 = utf8encode(ucs, *utf8);
- *utf8bytes -= (newutf8 - *utf8);
- *utf8 += (newutf8 - *utf8);
- }
-
- /* We have run out of utf16 bytes, read more if available */
- if ((n == 0) && (utf16bytes > 0)) {
- n = read(fd, utf16buf, MIN(sizeof(utf16buf), utf16bytes));
- utf16 = utf16buf;
- utf16bytes -= n;
- }
- }
-
- *utf8[0] = 0;
- --*utf8bytes;
-
- if (utf16bytes > 0) {
- /* Skip any remaining bytes */
- lseek(fd, utf16bytes, SEEK_CUR);
- }
- return;
-}
-
-static int asf_parse_header(int fd, struct mp3entry* id3,
- asf_waveformatex_t* wfx)
-{
- asf_object_t current;
- asf_object_t header;
- uint64_t datalen;
- int i;
- int fileprop = 0;
- uint64_t play_duration;
- uint16_t flags;
- uint32_t subobjects;
- uint8_t utf8buf[512];
- int id3buf_remaining = sizeof(id3->id3v2buf) + sizeof(id3->id3v1buf);
- unsigned char* id3buf = (unsigned char*)id3->id3v2buf;
-
- asf_read_object_header((asf_object_t *) &header, fd);
-
- //DEBUGF("header.size=%d\n",(int)header.size);
- if (header.size < 30) {
- /* invalid size for header object */
- return ASF_ERROR_OBJECT_SIZE;
- }
-
- read_uint32le(fd, &subobjects);
-
- /* Two reserved bytes - do we need to read them? */
- lseek(fd, 2, SEEK_CUR);
-
- //DEBUGF("Read header - size=%d, subobjects=%d\n",(int)header.size, (int)subobjects);
-
- if (subobjects > 0) {
- header.datalen = header.size - 30;
-
- /* TODO: Check that we have datalen bytes left in the file */
- datalen = header.datalen;
-
- for (i=0; i<(int)subobjects; i++) {
- //DEBUGF("Parsing header object %d - datalen=%d\n",i,(int)datalen);
- if (datalen < 24) {
- //DEBUGF("not enough data for reading object\n");
- break;
- }
-
- asf_read_object_header(&current, fd);
-
- if (current.size > datalen || current.size < 24) {
- //DEBUGF("invalid object size - current.size=%d, datalen=%d\n",(int)current.size,(int)datalen);
- break;
- }
-
- if (asf_guid_match(&current.guid, &asf_guid_file_properties)) {
- if (current.size < 104)
- return ASF_ERROR_OBJECT_SIZE;
-
- if (fileprop) {
- /* multiple file properties objects not allowed */
- return ASF_ERROR_INVALID_OBJECT;
- }
-
- fileprop = 1;
-
- /* Get the number of logical packets - uint16_t at offset 31
- * (Big endian byte order) */
- lseek(fd, 31, SEEK_CUR);
- read_uint16be(fd, &wfx->numpackets);
-
- /* Now get the play duration - uint64_t at offset 40 */
- lseek(fd, 7, SEEK_CUR);
- read_uint64le(fd, &play_duration);
- id3->length = play_duration / 10000;
-
- //DEBUGF("****** length = %lums\n", id3->length);
-
- /* Read the packet size - uint32_t at offset 68 */
- lseek(fd, 20, SEEK_CUR);
- read_uint32le(fd, &wfx->packet_size);
-
- /* Skip bytes remaining in object */
- lseek(fd, current.size - 24 - 72, SEEK_CUR);
- } else if (asf_guid_match(&current.guid, &asf_guid_stream_properties)) {
- guid_t guid;
- uint32_t propdatalen;
-
- if (current.size < 78)
- return ASF_ERROR_OBJECT_SIZE;
-
-#if 0
- asf_byteio_getGUID(&guid, current->data);
- datalen = asf_byteio_getDWLE(current->data + 40);
- flags = asf_byteio_getWLE(current->data + 48);
-#endif
-
- asf_readGUID(fd, &guid);
-
- lseek(fd, 24, SEEK_CUR);
- read_uint32le(fd, &propdatalen);
- lseek(fd, 4, SEEK_CUR);
- read_uint16le(fd, &flags);
-
- if (!asf_guid_match(&guid, &asf_guid_stream_type_audio)) {
- //DEBUGF("Found stream properties for non audio stream, skipping\n");
- lseek(fd,current.size - 24 - 50,SEEK_CUR);
- } else if (wfx->audiostream == -1) {
- lseek(fd, 4, SEEK_CUR);
- //DEBUGF("Found stream properties for audio stream %d\n",flags&0x7f);
-
- if (propdatalen < 18) {
- return ASF_ERROR_INVALID_LENGTH;
- }
-
-#if 0
- if (asf_byteio_getWLE(data + 16) > datalen - 16) {
- return ASF_ERROR_INVALID_LENGTH;
- }
-#endif
- read_uint16le(fd, &wfx->codec_id);
- read_uint16le(fd, &wfx->channels);
- read_uint32le(fd, &wfx->rate);
- read_uint32le(fd, &wfx->bitrate);
- wfx->bitrate *= 8;
- read_uint16le(fd, &wfx->blockalign);
- read_uint16le(fd, &wfx->bitspersample);
- read_uint16le(fd, &wfx->datalen);
-
- /* Round bitrate to the nearest kbit */
- id3->bitrate = (wfx->bitrate + 500) / 1000;
- id3->frequency = wfx->rate;
-
- if (wfx->codec_id == ASF_CODEC_ID_WMAV1) {
- read(fd, wfx->data, 4);
- lseek(fd,current.size - 24 - 72 - 4,SEEK_CUR);
- wfx->audiostream = flags&0x7f;
- } else if (wfx->codec_id == ASF_CODEC_ID_WMAV2) {
- read(fd, wfx->data, 6);
- lseek(fd,current.size - 24 - 72 - 6,SEEK_CUR);
- wfx->audiostream = flags&0x7f;
- } else if (wfx->codec_id == ASF_CODEC_ID_WMAPRO) {
- /* wma pro decoder needs the extra-data */
- read(fd, wfx->data, wfx->datalen);
- lseek(fd,current.size - 24 - 72 - wfx->datalen,SEEK_CUR);
- wfx->audiostream = flags&0x7f;
- /* Correct codectype to redirect playback to the proper .codec */
- id3->codectype = AFMT_WMAPRO;
- } else if (wfx->codec_id == ASF_CODEC_ID_WMAVOICE) {
- read(fd, wfx->data, wfx->datalen);
- lseek(fd,current.size - 24 - 72 - wfx->datalen,SEEK_CUR);
- wfx->audiostream = flags&0x7f;
- id3->codectype = AFMT_WMAVOICE;
- } else {
- DEBUGF("Unsupported WMA codec (Lossless, Voice, etc)\n");
- lseek(fd,current.size - 24 - 72,SEEK_CUR);
- }
-
- }
- } else if (asf_guid_match(&current.guid, &asf_guid_content_description)) {
- /* Object contains five 16-bit string lengths, followed by the five strings:
- title, artist, copyright, description, rating
- */
- uint16_t strlength[5];
- int i;
-
- //DEBUGF("Found GUID_CONTENT_DESCRIPTION - size=%d\n",(int)(current.size - 24));
-
- /* Read the 5 string lengths - number of bytes included trailing zero */
- for (i=0; i<5; i++) {
- read_uint16le(fd, &strlength[i]);
- //DEBUGF("strlength = %u\n",strlength[i]);
- }
-
- if (strlength[0] > 0) { /* 0 - Title */
- id3->title = id3buf;
- asf_utf16LEdecode(fd, strlength[0], &id3buf, &id3buf_remaining);
- }
-
- if (strlength[1] > 0) { /* 1 - Artist */
- id3->artist = id3buf;
- asf_utf16LEdecode(fd, strlength[1], &id3buf, &id3buf_remaining);
- }
-
- lseek(fd, strlength[2], SEEK_CUR); /* 2 - copyright */
-
- if (strlength[3] > 0) { /* 3 - description */
- id3->comment = id3buf;
- asf_utf16LEdecode(fd, strlength[3], &id3buf, &id3buf_remaining);
- }
-
- lseek(fd, strlength[4], SEEK_CUR); /* 4 - rating */
- } else if (asf_guid_match(&current.guid, &asf_guid_extended_content_description)) {
- uint16_t count;
- int i;
- int bytesleft = current.size - 24;
- //DEBUGF("Found GUID_EXTENDED_CONTENT_DESCRIPTION\n");
-
- read_uint16le(fd, &count);
- bytesleft -= 2;
- //DEBUGF("extended metadata count = %u\n",count);
-
- for (i=0; i < count; i++) {
- uint16_t length, type;
- unsigned char* utf8 = utf8buf;
- int utf8length = 512;
-
- read_uint16le(fd, &length);
- asf_utf16LEdecode(fd, length, &utf8, &utf8length);
- bytesleft -= 2 + length;
-
- read_uint16le(fd, &type);
- read_uint16le(fd, &length);
-
- if (!strcmp("WM/TrackNumber",utf8buf)) {
- if (type == 0) {
- id3->track_string = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- id3->tracknum = atoi(id3->track_string);
- } else if ((type >=2) && (type <= 5)) {
- id3->tracknum = asf_intdecode(fd, type, length);
- } else {
- lseek(fd, length, SEEK_CUR);
- }
- } else if ((!strcmp("WM/Genre", utf8buf)) && (type == 0)) {
- id3->genre_string = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- } else if ((!strcmp("WM/AlbumTitle", utf8buf)) && (type == 0)) {
- id3->album = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- } else if ((!strcmp("WM/AlbumArtist", utf8buf)) && (type == 0)) {
- id3->albumartist = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- } else if ((!strcmp("WM/Composer", utf8buf)) && (type == 0)) {
- id3->composer = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- } else if (!strcmp("WM/Year", utf8buf)) {
- if (type == 0) {
- id3->year_string = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- id3->year = atoi(id3->year_string);
- } else if ((type >=2) && (type <= 5)) {
- id3->year = asf_intdecode(fd, type, length);
- } else {
- lseek(fd, length, SEEK_CUR);
- }
- } else if (!strncmp("replaygain_", utf8buf, 11)) {
- char *value = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
- parse_replaygain(utf8buf, value, id3);
- } else if (!strcmp("MusicBrainz/Track Id", utf8buf)) {
- id3->mb_track_id = id3buf;
- asf_utf16LEdecode(fd, length, &id3buf, &id3buf_remaining);
-#ifdef HAVE_ALBUMART
- } else if (!strcmp("WM/Picture", utf8buf)) {
- uint32_t datalength, strlength;
- /* Expected is either "01 00 xx xx 03 yy yy yy yy" or
- * "03 yy yy yy yy". xx is the size of the WM/Picture
- * container in bytes. yy equals the raw data length of
- * the embedded image. */
- lseek(fd, -4, SEEK_CUR);
- read(fd, &type, 1);
- if (type == 1) {
- lseek(fd, 3, SEEK_CUR);
- read(fd, &type, 1);
- /* In case the parsing will fail in the next step we
- * might at least be able to skip the whole section. */
- datalength = length - 1;
- }
- if (type == 3) {
- /* Read the raw data length of the embedded image. */
- read_uint32le(fd, &datalength);
-
- /* Reset utf8 buffer */
- utf8 = utf8buf;
- utf8length = 512;
-
- /* Gather the album art format, this string has a
- * double zero-termination. */
- asf_utf16LEdecode(fd, 32, &utf8, &utf8length);
- strlength = (strlen(utf8buf) + 2) * 2;
- lseek(fd, strlength-32, SEEK_CUR);
- if (!strcmp("image/jpeg", utf8buf)) {
- id3->albumart.type = AA_TYPE_JPG;
- } else if (!strcmp("image/png", utf8buf)) {
- id3->albumart.type = AA_TYPE_PNG;
- } else {
- id3->albumart.type = AA_TYPE_UNKNOWN;
- }
-
- /* Set the album art size and position. */
- if (id3->albumart.type != AA_TYPE_UNKNOWN) {
- id3->albumart.pos = lseek(fd, 0, SEEK_CUR);
- id3->albumart.size = datalength;
- id3->has_embedded_albumart = true;
- }
- }
-
- lseek(fd, datalength, SEEK_CUR);
-#endif
- } else {
- lseek(fd, length, SEEK_CUR);
- }
- bytesleft -= 4 + length;
- }
-
- lseek(fd, bytesleft, SEEK_CUR);
- } else if (asf_guid_match(&current.guid, &asf_guid_content_encryption)
- || asf_guid_match(&current.guid, &asf_guid_extended_content_encryption)) {
- //DEBUGF("File is encrypted\n");
- return ASF_ERROR_ENCRYPTED;
- } else {
- //DEBUGF("Skipping %d bytes of object\n",(int)(current.size - 24));
- lseek(fd,current.size - 24,SEEK_CUR);
- }
-
- //DEBUGF("Parsed object - size = %d\n",(int)current.size);
- datalen -= current.size;
- }
-
- if (i != (int)subobjects || datalen != 0) {
- //DEBUGF("header data doesn't match given subobject count\n");
- return ASF_ERROR_INVALID_VALUE;
- }
-
- //DEBUGF("%d subobjects read successfully\n", i);
- }
-
-#if 0
- tmp = asf_parse_header_validate(file, &header);
- if (tmp < 0) {
- /* header read ok but doesn't validate correctly */
- return tmp;
- }
-#endif
-
- //DEBUGF("header validated correctly\n");
-
- return 0;
-}
-
-bool get_asf_metadata(int fd, struct mp3entry* id3)
-{
- int res;
- asf_object_t obj;
- asf_waveformatex_t wfx;
-
- wfx.audiostream = -1;
-
- res = asf_parse_header(fd, id3, &wfx);
-
- if (res < 0) {
- DEBUGF("ASF: parsing error - %d\n",res);
- return false;
- }
-
- if (wfx.audiostream == -1) {
- DEBUGF("ASF: No WMA streams found\n");
- return false;
- }
-
- asf_read_object_header(&obj, fd);
-
- if (!asf_guid_match(&obj.guid, &asf_guid_data)) {
- DEBUGF("ASF: No data object found\n");
- return false;
- }
-
- /* Store the current file position - no need to parse the header
- again in the codec. The +26 skips the rest of the data object
- header.
- */
- id3->first_frame_offset = lseek(fd, 0, SEEK_CUR) + 26;
- id3->filesize = filesize(fd);
- /* We copy the wfx struct to the MP3 TOC field in the id3 struct so
- the codec doesn't need to parse the header object again */
- memcpy(id3->toc, &wfx, sizeof(wfx));
-
- return true;
-}
diff --git a/apps/metadata/au.c b/apps/metadata/au.c
deleted file mode 100644
index 94e7453644..0000000000
--- a/apps/metadata/au.c
+++ /dev/null
@@ -1,105 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2010 Yoshihisa Uchida
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-static const unsigned char bitspersamples[9] = {
- 0, /* encoding */
- 8, /* 1: G.711 MULAW */
- 8, /* 2: Linear PCM 8bit */
- 16, /* 3: Linear PCM 16bit */
- 24, /* 4: Linear PCM 24bit */
- 32, /* 5: Linear PCM 32bit */
- 32, /* 6: IEEE float 32bit */
- 64, /* 7: IEEE float 64bit */
- /* encoding 8 - 26 unsupported. */
- 8, /* 27: G.711 ALAW */
-};
-
-static inline unsigned char get_au_bitspersample(unsigned int encoding)
-{
- if (encoding < 8)
- return bitspersamples[encoding];
- else if (encoding == 27)
- return bitspersamples[8];
-
- return 0;
-}
-
-bool get_au_metadata(int fd, struct mp3entry* id3)
-{
- /* temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- unsigned long numbytes = 0;
- int offset;
-
- id3->vbr = false; /* All Sun audio files are CBR */
- id3->filesize = filesize(fd);
- id3->length = 0;
-
- lseek(fd, 0, SEEK_SET);
- if ((read(fd, buf, 24) < 24) || (memcmp(buf, ".snd", 4) != 0))
- {
- /*
- * no header
- *
- * frequency: 8000 Hz
- * bits per sample: 8 bit
- * channel: mono
- */
- numbytes = id3->filesize;
- id3->frequency = 8000;
- id3->bitrate = 8;
- }
- else
- {
- /* parse header */
-
- /* data offset */
- offset = get_long_be(buf + 4);
- if (offset < 24)
- {
- DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
- return false;
- }
- /* data size */
- numbytes = get_long_be(buf + 8);
- if (numbytes == (uint32_t)0xffffffff)
- numbytes = id3->filesize - offset;
-
- id3->frequency = get_long_be(buf + 16);
- id3->bitrate = get_au_bitspersample(get_long_be(buf + 12)) * get_long_be(buf + 20)
- * id3->frequency / 1000;
- }
-
- /* Calculate track length [ms] */
- if (id3->bitrate)
- id3->length = (numbytes << 3) / id3->bitrate;
-
- return true;
-}
diff --git a/apps/metadata/ay.c b/apps/metadata/ay.c
deleted file mode 100644
index 5d00264b3d..0000000000
--- a/apps/metadata/ay.c
+++ /dev/null
@@ -1,148 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-/* Taken from blargg's Game_Music_Emu library */
-
-typedef unsigned char byte;
-
-/* AY file header */
-enum { header_size = 0x14 };
-struct header_t
-{
- byte tag[8];
- byte vers;
- byte player;
- byte unused[2];
- byte author[2];
- byte comment[2];
- byte max_track;
- byte first_track;
- byte track_info[2];
-};
-
-struct file_t {
- struct header_t const* header;
- byte const* tracks;
- byte const* end; /* end of file data */
-};
-
-static int get_be16( const void *a )
-{
- return get_short_be( (void*) a );
-}
-
-/* Given pointer to 2-byte offset of data, returns pointer to data, or NULL if
- * offset is 0 or there is less than min_size bytes of data available. */
-static byte const* get_data( struct file_t const* file, byte const ptr [], int min_size )
-{
- int offset = (int16_t) get_be16( ptr );
- int pos = ptr - (byte const*) file->header;
- int size = file->end - (byte const*) file->header;
- int limit = size - min_size;
- if ( limit < 0 || !offset || (unsigned) (pos + offset) > (unsigned) limit )
- return NULL;
- return ptr + offset;
-}
-
-static const char *parse_header( byte const in [], int size, struct file_t* out )
-{
- if ( size < header_size )
- return "wrong file type";
-
- out->header = (struct header_t const*) in;
- out->end = in + size;
- struct header_t const* h = (struct header_t const*) in;
- if ( memcmp( h->tag, "ZXAYEMUL", 8 ) )
- return "wrong file type";
-
- out->tracks = get_data( out, h->track_info, (h->max_track + 1) * 4 );
- if ( !out->tracks )
- return "missing track data";
-
- return 0;
-}
-
-static void copy_ay_fields( struct file_t const* file, struct mp3entry* id3, int track )
-{
- int track_count = file->header->max_track + 1;
-
- /* calculate track length based on number of subtracks */
- if (track_count > 1) {
- id3->length = file->header->max_track * 1000;
- } else {
- byte const* track_info = get_data( file, file->tracks + track * 4 + 2, 6 );
- if (track_info)
- id3->length = get_be16( track_info + 4 ) * (1000 / 50); /* frames to msec */
- else id3->length = 120 * 1000;
- }
-
- if ( id3->length <= 0 )
- id3->length = 120 * 1000; /* 2 minutes */
-
- /* If meta info was found in the m3u skip next step */
- if (id3->title && id3->title[0]) return;
-
- /* If file has more than one track will
- use file name as title */
- char * tmp;
- if (track_count <= 1) {
- tmp = (char *) get_data( file, file->tracks + track * 4, 1 );
- if ( tmp ) id3->title = tmp;
- }
-
- /* Author */
- tmp = (char *) get_data( file, file->header->author, 1 );
- if (tmp) id3->artist = tmp;
-
- /* Comment */
- tmp = (char *) get_data( file, file->header->comment, 1 );
- if (tmp) id3->comment = tmp;
-}
-
-static bool parse_ay_header(int fd, struct mp3entry *id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->id3v2buf;
- struct file_t file;
- int read_bytes;
-
- lseek(fd, 0, SEEK_SET);
- if ((read_bytes = read(fd, buf, ID3V2_BUF_SIZE)) < header_size)
- return false;
-
- buf [ID3V2_BUF_SIZE] = '\0';
- if ( parse_header( buf, read_bytes, &file ) )
- return false;
-
- copy_ay_fields( &file, id3, 0 );
- return true;
-}
-
-bool get_ay_metadata(int fd, struct mp3entry* id3)
-{
- char ay_type[8];
- if ((lseek(fd, 0, SEEK_SET) < 0) ||
- read(fd, ay_type, 8) < 8)
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* Make sure this is a ZX Ay file */
- if (memcmp( ay_type, "ZXAYEMUL", 8 ) != 0)
- return false;
-
- return parse_ay_header(fd, id3);
-}
diff --git a/apps/metadata/flac.c b/apps/metadata/flac.c
deleted file mode 100644
index 29937173fd..0000000000
--- a/apps/metadata/flac.c
+++ /dev/null
@@ -1,127 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-bool get_flac_metadata(int fd, struct mp3entry* id3)
-{
- /* A simple parser to read vital metadata from a FLAC file - length,
- * frequency, bitrate etc. This code should either be moved to a
- * seperate file, or discarded in favour of the libFLAC code.
- * The FLAC stream specification can be found at
- * http://flac.sourceforge.net/format.html#stream
- */
-
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- bool last_metadata = false;
- bool rc = false;
-
- if (!skip_id3v2(fd, id3) || (read(fd, buf, 4) < 4))
- {
- return rc;
- }
-
- if (memcmp(buf, "fLaC", 4) != 0)
- {
- return rc;
- }
-
- while (!last_metadata)
- {
- unsigned long i;
- int type;
-
- if (read(fd, buf, 4) < 0)
- {
- return rc;
- }
-
- last_metadata = buf[0] & 0x80;
- type = buf[0] & 0x7f;
- /* The length of the block */
- i = (buf[1] << 16) | (buf[2] << 8) | buf[3];
-
- if (type == 0) /* 0 is the STREAMINFO block */
- {
- unsigned long totalsamples;
-
- if (i >= sizeof(id3->path) || read(fd, buf, i) < 0)
- {
- return rc;
- }
-
- id3->vbr = true; /* All FLAC files are VBR */
- id3->filesize = filesize(fd);
- id3->frequency = (buf[10] << 12) | (buf[11] << 4)
- | ((buf[12] & 0xf0) >> 4);
- rc = true; /* Got vital metadata */
-
- /* totalsamples is a 36-bit field, but we assume <= 32 bits are used */
- totalsamples = get_long_be(&buf[14]);
-
- if(totalsamples > 0)
- {
- /* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
- id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
- id3->bitrate = (id3->filesize * 8) / id3->length;
- }
- else if (totalsamples == 0)
- {
- id3->length = 0;
- id3->bitrate = 0;
- }
- else
- {
- logf("flac length invalid!");
- return false;
- }
-
- }
- else if (type == 4) /* 4 is the VORBIS_COMMENT block */
- {
- /* The next i bytes of the file contain the VORBIS COMMENTS. */
- if (read_vorbis_tags(fd, id3, i) == 0)
- {
- return rc;
- }
- }
- else if (!last_metadata)
- {
- /* Skip to next metadata block */
- if (lseek(fd, i, SEEK_CUR) < 0)
- {
- return rc;
- }
- }
- }
-
- return true;
-}
diff --git a/apps/metadata/gbs.c b/apps/metadata/gbs.c
deleted file mode 100644
index 68f2b2a393..0000000000
--- a/apps/metadata/gbs.c
+++ /dev/null
@@ -1,65 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-static bool parse_gbs_header(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- lseek(fd, 0, SEEK_SET);
- if (read(fd, buf, 112) < 112)
- return false;
-
- /* Calculate track length with number of subtracks */
- id3->length = buf[4] * 1000;
-
- /* If meta info was found in the m3u skip next step */
- if (id3->title && id3->title[0]) return true;
-
- char *p = id3->id3v2buf;
-
- /* Some metadata entries have 32 bytes length */
- /* Game */
- memcpy(p, &buf[16], 32); *(p + 33) = '\0';
- id3->title = p;
- p += strlen(p)+1;
-
- /* Artist */
- memcpy(p, &buf[48], 32); *(p + 33) = '\0';
- id3->artist = p;
- p += strlen(p)+1;
-
- /* Copyright */
- memcpy(p, &buf[80], 32); *(p + 33) = '\0';
- id3->album = p;
-
- return true;
-}
-
-bool get_gbs_metadata(int fd, struct mp3entry* id3)
-{
- char gbs_type[3];
- if ((lseek(fd, 0, SEEK_SET) < 0) ||
- (read(fd, gbs_type, 3) < 3))
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
- /* we only render 16 bits, 44.1KHz, Stereo */
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* Check for GBS magic */
- if (memcmp( gbs_type, "GBS", 3 ) != 0)
- return false;
-
- return parse_gbs_header(fd, id3);
-}
diff --git a/apps/metadata/hes.c b/apps/metadata/hes.c
deleted file mode 100644
index 6d99d523cb..0000000000
--- a/apps/metadata/hes.c
+++ /dev/null
@@ -1,39 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-#include "plugin.h"
-
-bool get_hes_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the id3v2 buffer part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->id3v2buf;
- int read_bytes;
-
- if ((lseek(fd, 0, SEEK_SET) < 0)
- || ((read_bytes = read(fd, buf, 4)) < 4))
- return false;
-
- /* Verify this is a HES file */
- if (memcmp(buf,"HESM",4) != 0)
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
- /* we only render 16 bits, 44.1KHz, Stereo */
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* Set default track count (length)*/
- id3->length = 255 * 1000;
-
- return true;
-}
-
diff --git a/apps/metadata/id3tags.c b/apps/metadata/id3tags.c
deleted file mode 100644
index 2dd1c662ed..0000000000
--- a/apps/metadata/id3tags.c
+++ /dev/null
@@ -1,1199 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2002 by Daniel Stenberg
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-/*
- * Parts of this code has been stolen from the Ample project and was written
- * by David H�deman. It has since been extended and enhanced pretty much by
- * all sorts of friendly Rockbox people.
- *
- */
-
- /* tagResolver and associated code copyright 2003 Thomas Paul Diffenbach
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <errno.h>
-#include <stdbool.h>
-#include <stddef.h>
-#include <ctype.h>
-#include "string-extra.h"
-#include "config.h"
-#include "file.h"
-#include "logf.h"
-#include "system.h"
-#include "replaygain.h"
-#include "rbunicode.h"
-
-#include "metadata.h"
-#include "mp3data.h"
-#if CONFIG_CODEC == SWCODEC
-#include "metadata_common.h"
-#endif
-#include "metadata_parsers.h"
-#include "misc.h"
-
-static unsigned long unsync(unsigned long b0,
- unsigned long b1,
- unsigned long b2,
- unsigned long b3)
-{
- return (((long)(b0 & 0x7F) << (3*7)) |
- ((long)(b1 & 0x7F) << (2*7)) |
- ((long)(b2 & 0x7F) << (1*7)) |
- ((long)(b3 & 0x7F) << (0*7)));
-}
-
-static const char* const genres[] = {
- "Blues", "Classic Rock", "Country", "Dance", "Disco", "Funk", "Grunge",
- "Hip-Hop", "Jazz", "Metal", "New Age", "Oldies", "Other", "Pop", "R&B",
- "Rap", "Reggae", "Rock", "Techno", "Industrial", "Alternative", "Ska",
- "Death Metal", "Pranks", "Soundtrack", "Euro-Techno", "Ambient", "Trip-Hop",
- "Vocal", "Jazz+Funk", "Fusion", "Trance", "Classical", "Instrumental",
- "Acid", "House", "Game", "Sound Clip", "Gospel", "Noise", "AlternRock",
- "Bass", "Soul", "Punk", "Space", "Meditative", "Instrumental Pop",
- "Instrumental Rock", "Ethnic", "Gothic", "Darkwave", "Techno-Industrial",
- "Electronic", "Pop-Folk", "Eurodance", "Dream", "Southern Rock", "Comedy",
- "Cult", "Gangsta", "Top 40", "Christian Rap", "Pop/Funk", "Jungle",
- "Native American", "Cabaret", "New Wave", "Psychadelic", "Rave",
- "Showtunes", "Trailer", "Lo-Fi", "Tribal", "Acid Punk", "Acid Jazz",
- "Polka", "Retro", "Musical", "Rock & Roll", "Hard Rock",
-
- /* winamp extensions */
- "Folk", "Folk-Rock", "National Folk", "Swing", "Fast Fusion", "Bebob",
- "Latin", "Revival", "Celtic", "Bluegrass", "Avantgarde", "Gothic Rock",
- "Progressive Rock", "Psychedelic Rock", "Symphonic Rock", "Slow Rock",
- "Big Band", "Chorus", "Easy Listening", "Acoustic", "Humour", "Speech",
- "Chanson", "Opera", "Chamber Music", "Sonata", "Symphony", "Booty Bass",
- "Primus", "Porn Groove", "Satire", "Slow Jam", "Club", "Tango", "Samba",
- "Folklore", "Ballad", "Power Ballad", "Rhythmic Soul", "Freestyle",
- "Duet", "Punk Rock", "Drum Solo", "A capella", "Euro-House", "Dance Hall",
- "Goa", "Drum & Bass", "Club-House", "Hardcore", "Terror", "Indie",
- "BritPop", "Negerpunk", "Polsk Punk", "Beat", "Christian Gangsta Rap",
- "Heavy Metal", "Black Metal", "Crossover", "Contemporary Christian",
- "Christian Rock", "Merengue", "Salsa", "Thrash Metal", "Anime", "Jpop",
- "Synthpop"
-};
-
-#if CONFIG_CODEC != SWCODEC
-static
-#endif
-char* id3_get_num_genre(unsigned int genre_num)
-{
- if (genre_num < ARRAYLEN(genres))
- return (char*)genres[genre_num];
- return NULL;
-}
-
-/*
- HOW TO ADD ADDITIONAL ID3 VERSION 2 TAGS
- Code and comments by Thomas Paul Diffenbach
-
- To add another ID3v2 Tag, do the following:
- 1. add a char* named for the tag to struct mp3entry in id3.h,
- (I (tpd) prefer to use char* rather than ints, even for what seems like
- numerical values, for cases where a number won't do, e.g.,
- YEAR: "circa 1765", "1790/1977" (composed/performed), "28 Feb 1969"
- TRACK: "1/12", "1 of 12", GENRE: "Freeform genre name"
- Text is more flexible, and as the main use of id3 data is to
- display it, converting it to an int just means reconverting to
- display it, at a runtime cost.)
-
- 2. If any special processing beyond copying the tag value from the Id3
- block to the struct mp3entry is rrequired (such as converting to an
- int), write a function to perform this special processing.
-
- This function's prototype must match that of
- typedef tagPostProcessFunc, that is it must be:
- int func( struct mp3entry*, char* tag, int bufferpos )
- the first argument is a pointer to the current mp3entry structure the
- second argument is a pointer to the null terminated string value of the
- tag found the third argument is the offset of the next free byte in the
- mp3entry's buffer your function should return the corrected offset; if
- you don't lengthen or shorten the tag string, you can return the third
- argument unchanged.
-
- Unless you have a good reason no to, make the function static.
- TO JUST COPY THE TAG NO SPECIAL PROCESSING FUNCTION IS NEEDED.
-
- 3. add one or more entries to the tagList array, using the format:
- char* ID3 Tag symbolic name -- see the ID3 specification for these,
- sizeof() that name minus 1,
- offsetof( struct mp3entry, variable_name_in_struct_mp3entry ),
- pointer to your special processing function or NULL
- if you need no special processing
- flag indicating if this tag is binary or textual
- Many ID3 symbolic names come in more than one form. You can add both
- forms, each referencing the same variable in struct mp3entry.
- If both forms are present, the last found will be used.
- Note that the offset can be zero, in which case no entry will be set
- in the mp3entry struct; the frame is still read into the buffer and
- the special processing function is called (several times, if there
- are several frames with the same name).
-
- 4. Alternately, use the TAG_LIST_ENTRY macro with
- ID3 tag symbolic name,
- variable in struct mp3entry,
- special processing function address
-
- 5. Add code to wps-display.c function get_tag to assign a printf-like
- format specifier for the tag */
-
-/* Structure for ID3 Tag extraction information */
-struct tag_resolver {
- const char* tag;
- int tag_length;
- size_t offset;
- int (*ppFunc)(struct mp3entry*, char* tag, int bufferpos);
- bool binary;
-};
-
-static bool global_ff_found;
-
-static int unsynchronize(char* tag, int len, bool *ff_found)
-{
- int i;
- unsigned char c;
- unsigned char *rp, *wp;
-
- wp = rp = (unsigned char *)tag;
-
- rp = (unsigned char *)tag;
- for(i = 0;i < len;i++) {
- /* Read the next byte and write it back, but don't increment the
- write pointer */
- c = *rp++;
- *wp = c;
- if(*ff_found) {
- /* Increment the write pointer if it isn't an unsynch pattern */
- if(c != 0)
- wp++;
- *ff_found = false;
- } else {
- if(c == 0xff)
- *ff_found = true;
- wp++;
- }
- }
- return (long)wp - (long)tag;
-}
-
-static int unsynchronize_frame(char* tag, int len)
-{
- bool ff_found = false;
-
- return unsynchronize(tag, len, &ff_found);
-}
-
-static int read_unsynched(int fd, void *buf, int len)
-{
- int i;
- int rc;
- int remaining = len;
- char *wp;
- char *rp;
-
- wp = buf;
-
- while(remaining) {
- rp = wp;
- rc = read(fd, rp, remaining);
- if(rc <= 0)
- return rc;
-
- i = unsynchronize(wp, remaining, &global_ff_found);
- remaining -= i;
- wp += i;
- }
-
- return len;
-}
-
-static int skip_unsynched(int fd, int len)
-{
- int rc;
- int remaining = len;
- int rlen;
- char buf[32];
-
- while(remaining) {
- rlen = MIN(sizeof(buf), (unsigned int)remaining);
- rc = read(fd, buf, rlen);
- if(rc <= 0)
- return rc;
-
- remaining -= unsynchronize(buf, rlen, &global_ff_found);
- }
-
- return len;
-}
-
-/* parse numeric value from string */
-static int parsetracknum( struct mp3entry* entry, char* tag, int bufferpos )
-{
- entry->tracknum = atoi( tag );
- return bufferpos;
-}
-
-/* parse numeric value from string */
-static int parsediscnum( struct mp3entry* entry, char* tag, int bufferpos )
-{
- entry->discnum = atoi( tag );
- return bufferpos;
-}
-
-/* parse numeric value from string */
-static int parseyearnum( struct mp3entry* entry, char* tag, int bufferpos )
-{
- entry->year = atoi( tag );
- return bufferpos;
-}
-
-/* parse numeric genre from string, version 2.2 and 2.3 */
-static int parsegenre( struct mp3entry* entry, char* tag, int bufferpos )
-{
- /* Use bufferpos to hold current position in entry->id3v2buf. */
- bufferpos = tag - entry->id3v2buf;
-
- if(entry->id3version >= ID3_VER_2_4) {
- /* In version 2.4 and up, there are no parentheses, and the genre frame
- is a list of strings, either numbers or text. */
-
- /* Is it a number? */
- if(isdigit(tag[0])) {
- entry->genre_string = id3_get_num_genre(atoi( tag ));
- return bufferpos;
- } else {
- entry->genre_string = tag;
- return bufferpos + strlen(tag) + 1;
- }
- } else {
- if( tag[0] == '(' && tag[1] != '(' ) {
- entry->genre_string = id3_get_num_genre(atoi( tag + 1 ));
- return bufferpos;
- }
- else {
- entry->genre_string = tag;
- return bufferpos + strlen(tag) + 1;
- }
- }
-}
-
-#ifdef HAVE_ALBUMART
-/* parse embed albumart */
-static int parsealbumart( struct mp3entry* entry, char* tag, int bufferpos )
-{
- entry->has_embedded_albumart = false;
-
- /* we currently don't support unsynchronizing albumart */
- if (entry->albumart.type == AA_TYPE_UNSYNC)
- return bufferpos;
-
- entry->albumart.type = AA_TYPE_UNKNOWN;
-
- char *start = tag;
- /* skip text encoding */
- tag += 1;
-
- if (memcmp(tag, "image/", 6) == 0)
- {
- /* ID3 v2.3+ */
- tag += 6;
- if (strcmp(tag, "jpeg") == 0)
- {
- entry->albumart.type = AA_TYPE_JPG;
- tag += 5;
- }
- else if (strcmp(tag, "png") == 0)
- {
- entry->albumart.type = AA_TYPE_PNG;
- tag += 4;
- }
- }
- else
- {
- /* ID3 v2.2 */
- if (memcmp(tag, "JPG", 3) == 0)
- entry->albumart.type = AA_TYPE_JPG;
- else if (memcmp(tag, "PNG", 3) == 0)
- entry->albumart.type = AA_TYPE_PNG;
- tag += 3;
- }
-
- if (entry->albumart.type != AA_TYPE_UNKNOWN)
- {
- /* skip picture type */
- tag += 1;
- /* skip description */
- tag = strchr(tag, '\0') + 1;
- /* fixup offset&size for image data */
- entry->albumart.pos += tag - start;
- entry->albumart.size -= tag - start;
- entry->has_embedded_albumart = true;
- }
- /* return bufferpos as we didn't store anything in id3v2buf */
- return bufferpos;
-}
-#endif
-
-/* parse user defined text, looking for album artist and replaygain
- * information.
- */
-static int parseuser( struct mp3entry* entry, char* tag, int bufferpos )
-{
- char* value = NULL;
- int desc_len = strlen(tag);
- int length = 0;
-
- if ((tag - entry->id3v2buf + desc_len + 2) < bufferpos) {
- /* At least part of the value was read, so we can safely try to
- * parse it */
- value = tag + desc_len + 1;
-
- if (!strcasecmp(tag, "ALBUM ARTIST")) {
- length = strlen(value) + 1;
- strlcpy(tag, value, length);
- entry->albumartist = tag;
-#if CONFIG_CODEC == SWCODEC
- } else {
- /* Call parse_replaygain(). */
- parse_replaygain(tag, value, entry);
-#endif
- }
- }
-
- return tag - entry->id3v2buf + length;
-}
-
-#if CONFIG_CODEC == SWCODEC
-/* parse RVA2 binary data and convert to replaygain information. */
-static int parserva2( struct mp3entry* entry, char* tag, int bufferpos)
-{
- int desc_len = strlen(tag);
- int start_pos = tag - entry->id3v2buf;
- int end_pos = start_pos + desc_len + 5;
- unsigned char* value = tag + desc_len + 1;
-
- /* Only parse RVA2 replaygain tags if tag version == 2.4 and channel
- * type is master volume.
- */
- if (entry->id3version == ID3_VER_2_4 && end_pos < bufferpos
- && *value++ == 1) {
- long gain = 0;
- long peak = 0;
- long peakbits;
- long peakbytes;
- bool album = false;
-
- /* The RVA2 specification is unclear on some things (id string and
- * peak volume), but this matches how Quod Libet use them.
- */
-
- gain = (int16_t) ((value[0] << 8) | value[1]);
- value += 2;
- peakbits = *value++;
- peakbytes = (peakbits + 7) / 8;
-
- /* Only use the topmost 24 bits for peak volume */
- if (peakbytes > 3) {
- peakbytes = 3;
- }
-
- /* Make sure the peak bits were read */
- if (end_pos + peakbytes < bufferpos) {
- long shift = ((8 - (peakbits & 7)) & 7) + (3 - peakbytes) * 8;
-
- for ( ; peakbytes; peakbytes--) {
- peak <<= 8;
- peak += *value++;
- }
-
- peak <<= shift;
-
- if (peakbits > 24) {
- peak += *value >> (8 - shift);
- }
- }
-
- if (strcasecmp(tag, "album") == 0) {
- album = true;
- } else if (strcasecmp(tag, "track") != 0) {
- /* Only accept non-track values if we don't have any previous
- * value.
- */
- if (entry->track_gain != 0) {
- return start_pos;
- }
- }
-
- parse_replaygain_int(album, gain, peak * 2, entry);
- }
-
- return start_pos;
-}
-#endif
-
-static int parsembtid( struct mp3entry* entry, char* tag, int bufferpos )
-{
- char* value = NULL;
- int desc_len = strlen(tag);
- /*DEBUGF("MBID len: %d\n", desc_len);*/
- /* Musicbrainz track IDs are always 36 chars long */
- const size_t mbtid_len = 36;
-
- if ((tag - entry->id3v2buf + desc_len + 2) < bufferpos)
- {
- value = tag + desc_len + 1;
-
- if (strcasecmp(tag, "http://musicbrainz.org") == 0)
- {
- if (mbtid_len == strlen(value))
- {
- entry->mb_track_id = value;
- return bufferpos + mbtid_len + 1;
- }
- }
- }
-
- return bufferpos;
-}
-
-static const struct tag_resolver taglist[] = {
- { "TPE1", 4, offsetof(struct mp3entry, artist), NULL, false },
- { "TP1", 3, offsetof(struct mp3entry, artist), NULL, false },
- { "TIT2", 4, offsetof(struct mp3entry, title), NULL, false },
- { "TT2", 3, offsetof(struct mp3entry, title), NULL, false },
- { "TALB", 4, offsetof(struct mp3entry, album), NULL, false },
- { "TAL", 3, offsetof(struct mp3entry, album), NULL, false },
- { "TRK", 3, offsetof(struct mp3entry, track_string), &parsetracknum, false },
- { "TPOS", 4, offsetof(struct mp3entry, disc_string), &parsediscnum, false },
- { "TPA", 3, offsetof(struct mp3entry, disc_string), &parsediscnum, false },
- { "TRCK", 4, offsetof(struct mp3entry, track_string), &parsetracknum, false },
- { "TDRC", 4, offsetof(struct mp3entry, year_string), &parseyearnum, false },
- { "TYER", 4, offsetof(struct mp3entry, year_string), &parseyearnum, false },
- { "TYE", 3, offsetof(struct mp3entry, year_string), &parseyearnum, false },
- { "TCOM", 4, offsetof(struct mp3entry, composer), NULL, false },
- { "TCM", 3, offsetof(struct mp3entry, composer), NULL, false },
- { "TPE2", 4, offsetof(struct mp3entry, albumartist), NULL, false },
- { "TP2", 3, offsetof(struct mp3entry, albumartist), NULL, false },
- { "TIT1", 4, offsetof(struct mp3entry, grouping), NULL, false },
- { "TT1", 3, offsetof(struct mp3entry, grouping), NULL, false },
- { "COMM", 4, offsetof(struct mp3entry, comment), NULL, false },
- { "COM", 3, offsetof(struct mp3entry, comment), NULL, false },
- { "TCON", 4, offsetof(struct mp3entry, genre_string), &parsegenre, false },
- { "TCO", 3, offsetof(struct mp3entry, genre_string), &parsegenre, false },
-#ifdef HAVE_ALBUMART
- { "APIC", 4, 0, &parsealbumart, true },
- { "PIC", 3, 0, &parsealbumart, true },
-#endif
- { "TXXX", 4, 0, &parseuser, false },
-#if CONFIG_CODEC == SWCODEC
- { "RVA2", 4, 0, &parserva2, true },
-#endif
- { "UFID", 4, 0, &parsembtid, false },
-};
-
-#define TAGLIST_SIZE ((int)ARRAYLEN(taglist))
-
-/* Get the length of an ID3 string in the given encoding. Returns the length
- * in bytes, including end nil, or -1 if the encoding is unknown.
- */
-static int unicode_len(char encoding, const void* string)
-{
- int len = 0;
-
- if (encoding == 0x01 || encoding == 0x02) {
- char first;
- const char *s = string;
- /* string might be unaligned, so using short* can crash on ARM and SH1 */
- do {
- first = *s++;
- } while ((first | *s++) != 0);
-
- len = s - (const char*) string;
- } else {
- len = strlen((char*) string) + 1;
- }
-
- return len;
-}
-
-/* Checks to see if the passed in string is a 16-bit wide Unicode v2
- string. If it is, we convert it to a UTF-8 string. If it's not unicode,
- we convert from the default codepage */
-static int unicode_munge(char* string, char* utf8buf, int *len) {
- long tmp;
- bool le = false;
- int i = 0;
- unsigned char *str = (unsigned char *)string;
- int templen = 0;
- unsigned char* utf8 = (unsigned char *)utf8buf;
-
- switch (str[0]) {
- case 0x00: /* Type 0x00 is ordinary ISO 8859-1 */
- str++;
- (*len)--;
- utf8 = iso_decode(str, utf8, -1, *len);
- *utf8 = 0;
- *len = (unsigned long)utf8 - (unsigned long)utf8buf;
- break;
-
- case 0x01: /* Unicode with or without BOM */
- case 0x02:
- (*len)--;
- str++;
-
- /* Handle frames with more than one string
- (needed for TXXX frames).*/
- do {
- tmp = bytes2int(0, 0, str[0], str[1]);
-
- /* Now check if there is a BOM
- (zero-width non-breaking space, 0xfeff)
- and if it is in little or big endian format */
- if(tmp == 0xfffe) { /* Little endian? */
- le = true;
- str += 2;
- (*len)-=2;
- } else if(tmp == 0xfeff) { /* Big endian? */
- str += 2;
- (*len)-=2;
- } else
- /* If there is no BOM (which is a specification violation),
- let's try to guess it. If one of the bytes is 0x00, it is
- probably the most significant one. */
- if(str[1] == 0)
- le = true;
-
- while ((i < *len) && (str[0] || str[1])) {
- if(le)
- utf8 = utf16LEdecode(str, utf8, 1);
- else
- utf8 = utf16BEdecode(str, utf8, 1);
-
- str+=2;
- i += 2;
- }
-
- *utf8++ = 0; /* Terminate the string */
- templen += (strlen(&utf8buf[templen]) + 1);
- str += 2;
- i+=2;
- } while(i < *len);
- *len = templen - 1;
- break;
-
- case 0x03: /* UTF-8 encoded string */
- for(i=0; i < *len; i++)
- utf8[i] = str[i+1];
- (*len)--;
- break;
-
- default: /* Plain old string */
- utf8 = iso_decode(str, utf8, -1, *len);
- *utf8 = 0;
- *len = (unsigned long)utf8 - (unsigned long)utf8buf;
- break;
- }
- return 0;
-}
-
-/*
- * Sets the title of an MP3 entry based on its ID3v1 tag.
- *
- * Arguments: file - the MP3 file to scen for a ID3v1 tag
- * entry - the entry to set the title in
- *
- * Returns: true if a title was found and created, else false
- */
-bool setid3v1title(int fd, struct mp3entry *entry)
-{
- unsigned char buffer[128];
- static const char offsets[] = {3, 33, 63, 97, 93, 125, 127};
- int i, j;
- unsigned char* utf8;
-
- if (-1 == lseek(fd, -128, SEEK_END))
- return false;
-
- if (read(fd, buffer, sizeof buffer) != sizeof buffer)
- return false;
-
- if (strncmp((char *)buffer, "TAG", 3))
- return false;
-
- entry->id3v1len = 128;
- entry->id3version = ID3_VER_1_0;
-
- for (i=0; i < (int)sizeof offsets; i++) {
- unsigned char* ptr = (unsigned char *)buffer + offsets[i];
-
- switch(i) {
- case 0:
- case 1:
- case 2:
- /* kill trailing space in strings */
- for (j=29; j && (ptr[j]==0 || ptr[j]==' '); j--)
- ptr[j] = 0;
- /* convert string to utf8 */
- utf8 = (unsigned char *)entry->id3v1buf[i];
- utf8 = iso_decode(ptr, utf8, -1, 30);
- /* make sure string is terminated */
- *utf8 = 0;
- break;
-
- case 3:
- /* kill trailing space in strings */
- for (j=27; j && (ptr[j]==0 || ptr[j]==' '); j--)
- ptr[j] = 0;
- /* convert string to utf8 */
- utf8 = (unsigned char *)entry->id3v1buf[3];
- utf8 = iso_decode(ptr, utf8, -1, 28);
- /* make sure string is terminated */
- *utf8 = 0;
- break;
-
- case 4:
- ptr[4] = 0;
- entry->year = atoi((char *)ptr);
- break;
-
- case 5:
- /* id3v1.1 uses last two bytes of comment field for track
- number: first must be 0 and second is track num */
- if (!ptr[0] && ptr[1]) {
- entry->tracknum = ptr[1];
- entry->id3version = ID3_VER_1_1;
- }
- break;
-
- case 6:
- /* genre */
- entry->genre_string = id3_get_num_genre(ptr[0]);
- break;
- }
- }
-
- entry->title = entry->id3v1buf[0];
- entry->artist = entry->id3v1buf[1];
- entry->album = entry->id3v1buf[2];
- entry->comment = entry->id3v1buf[3];
-
- return true;
-}
-
-
-/*
- * Sets the title of an MP3 entry based on its ID3v2 tag.
- *
- * Arguments: file - the MP3 file to scan for a ID3v2 tag
- * entry - the entry to set the title in
- *
- * Returns: true if a title was found and created, else false
- */
-void setid3v2title(int fd, struct mp3entry *entry)
-{
- int minframesize;
- int size;
- long bufferpos = 0, totframelen, framelen;
- char header[10];
- char tmp[4];
- unsigned char version;
- char *buffer = entry->id3v2buf;
- int bytesread = 0;
- int buffersize = sizeof(entry->id3v2buf);
- unsigned char global_flags;
- int flags;
- bool global_unsynch = false;
- bool unsynch = false;
- int i, j;
- int rc;
-#if CONFIG_CODEC == SWCODEC
- bool itunes_gapless = false;
-#endif
-
- global_ff_found = false;
-
- /* Bail out if the tag is shorter than 10 bytes */
- if(entry->id3v2len < 10)
- return;
-
- /* Read the ID3 tag version from the header */
- lseek(fd, 0, SEEK_SET);
- if(10 != read(fd, header, 10))
- return;
-
- /* Get the total ID3 tag size */
- size = entry->id3v2len - 10;
-
- version = header[3];
- switch ( version ) {
- case 2:
- version = ID3_VER_2_2;
- minframesize = 8;
- break;
-
- case 3:
- version = ID3_VER_2_3;
- minframesize = 12;
- break;
-
- case 4:
- version = ID3_VER_2_4;
- minframesize = 12;
- break;
-
- default:
- /* unsupported id3 version */
- return;
- }
- entry->id3version = version;
- entry->tracknum = entry->year = entry->discnum = 0;
- entry->title = entry->artist = entry->album = NULL; /* FIXME incomplete */
-
- global_flags = header[5];
-
- /* Skip the extended header if it is present */
- if(global_flags & 0x40) {
- if(version == ID3_VER_2_3) {
- if(10 != read(fd, header, 10))
- return;
- /* The 2.3 extended header size doesn't include the header size
- field itself. Also, it is not unsynched. */
- framelen =
- bytes2int(header[0], header[1], header[2], header[3]) + 4;
-
- /* Skip the rest of the header */
- lseek(fd, framelen - 10, SEEK_CUR);
- }
-
- if(version >= ID3_VER_2_4) {
- if(4 != read(fd, header, 4))
- return;
-
- /* The 2.4 extended header size does include the entire header,
- so here we can just skip it. This header is unsynched. */
- framelen = unsync(header[0], header[1],
- header[2], header[3]);
-
- lseek(fd, framelen - 4, SEEK_CUR);
- }
- }
-
- /* Is unsynchronization applied? */
- if(global_flags & 0x80) {
- global_unsynch = true;
- }
-
- /*
- * We must have at least minframesize bytes left for the
- * remaining frames to be interesting
- */
- while (size >= minframesize && bufferpos < buffersize - 1) {
- flags = 0;
-
- /* Read frame header and check length */
- if(version >= ID3_VER_2_3) {
- if(global_unsynch && version <= ID3_VER_2_3)
- rc = read_unsynched(fd, header, 10);
- else
- rc = read(fd, header, 10);
- if(rc != 10)
- return;
- /* Adjust for the 10 bytes we read */
- size -= 10;
-
- flags = bytes2int(0, 0, header[8], header[9]);
-
- if (version >= ID3_VER_2_4) {
- framelen = unsync(header[4], header[5],
- header[6], header[7]);
- } else {
- /* version .3 files don't use synchsafe ints for
- * size */
- framelen = bytes2int(header[4], header[5],
- header[6], header[7]);
- }
- } else {
- if(6 != read(fd, header, 6))
- return;
- /* Adjust for the 6 bytes we read */
- size -= 6;
-
- framelen = bytes2int(0, header[3], header[4], header[5]);
- }
-
- logf("framelen = %ld, flags = 0x%04x", framelen, flags);
- if(framelen == 0){
- if (header[0] == 0 && header[1] == 0 && header[2] == 0)
- return;
- else
- continue;
- }
-
- unsynch = false;
-
- if(flags)
- {
- if (version >= ID3_VER_2_4) {
- if(flags & 0x0040) { /* Grouping identity */
- lseek(fd, 1, SEEK_CUR); /* Skip 1 byte */
- framelen--;
- }
- } else {
- if(flags & 0x0020) { /* Grouping identity */
- lseek(fd, 1, SEEK_CUR); /* Skip 1 byte */
- framelen--;
- }
- }
-
- if(flags & 0x000c) /* Compression or encryption */
- {
- /* Skip it */
- size -= framelen;
- lseek(fd, framelen, SEEK_CUR);
- continue;
- }
-
- if(flags & 0x0002) /* Unsynchronization */
- unsynch = true;
-
- if (version >= ID3_VER_2_4) {
- if(flags & 0x0001) { /* Data length indicator */
- if(4 != read(fd, tmp, 4))
- return;
-
- /* We don't need the data length */
- framelen -= 4;
- }
- }
- }
-
- if (framelen == 0)
- continue;
-
- if (framelen < 0)
- return;
-
- /* Keep track of the remaining frame size */
- totframelen = framelen;
-
- /* If the frame is larger than the remaining buffer space we try
- to read as much as would fit in the buffer */
- if(framelen >= buffersize - bufferpos)
- framelen = buffersize - bufferpos - 1;
-
- /* Limit the maximum length of an id3 data item to ID3V2_MAX_ITEM_SIZE
- bytes. This reduces the chance that the available buffer is filled
- by single metadata items like large comments. */
- if (ID3V2_MAX_ITEM_SIZE < framelen)
- framelen = ID3V2_MAX_ITEM_SIZE;
-
- logf("id3v2 frame: %.4s", header);
-
- /* Check for certain frame headers
-
- 'size' is the amount of frame bytes remaining. We decrement it by
- the amount of bytes we read. If we fail to read as many bytes as
- we expect, we assume that we can't read from this file, and bail
- out.
-
- For each frame. we will iterate over the list of supported tags,
- and read the tag into entry's buffer. All tags will be kept as
- strings, for cases where a number won't do, e.g., YEAR: "circa
- 1765", "1790/1977" (composed/performed), "28 Feb 1969" TRACK:
- "1/12", "1 of 12", GENRE: "Freeform genre name" Text is more
- flexible, and as the main use of id3 data is to display it,
- converting it to an int just means reconverting to display it, at a
- runtime cost.
-
- For tags that the current code does convert to ints, a post
- processing function will be called via a pointer to function. */
-
- for (i=0; i<TAGLIST_SIZE; i++) {
- const struct tag_resolver* tr = &taglist[i];
- char** ptag = tr->offset ? (char**) (((char*)entry) + tr->offset)
- : NULL;
- char* tag;
-
- /* Only ID3_VER_2_2 uses frames with three-character names. */
- if (((version == ID3_VER_2_2) && (tr->tag_length != 3))
- || ((version > ID3_VER_2_2) && (tr->tag_length != 4))) {
- continue;
- }
-
- if( !memcmp( header, tr->tag, tr->tag_length ) ) {
-
- /* found a tag matching one in tagList, and not yet filled */
- tag = buffer + bufferpos;
-
- if(global_unsynch && version <= ID3_VER_2_3)
- bytesread = read_unsynched(fd, tag, framelen);
- else
- bytesread = read(fd, tag, framelen);
-
- if( bytesread != framelen )
- return;
-
- size -= bytesread;
-
- if(unsynch || (global_unsynch && version >= ID3_VER_2_4))
- bytesread = unsynchronize_frame(tag, bytesread);
-
- /* the COMM frame has a 3 char field to hold an ISO-639-1
- * language string and an optional short description;
- * remove them so unicode_munge can work correctly
- */
-
- if((tr->tag_length == 4 && !memcmp( header, "COMM", 4)) ||
- (tr->tag_length == 3 && !memcmp( header, "COM", 3))) {
- int offset;
- if(bytesread >= 8 && !strncmp(tag+4, "iTun", 4)) {
-#if CONFIG_CODEC == SWCODEC
- /* check for iTunes gapless information */
- if(bytesread >= 12 && !strncmp(tag+4, "iTunSMPB", 8))
- itunes_gapless = true;
- else
-#endif
- /* ignore other with iTunes tags */
- break;
- }
-
- offset = 3 + unicode_len(*tag, tag + 4);
- if(bytesread > offset) {
- bytesread -= offset;
- memmove(tag + 1, tag + 1 + offset, bytesread - 1);
- }
- }
-
- /* Attempt to parse Unicode string only if the tag contents
- aren't binary */
- if(!tr->binary) {
- /* UTF-8 could potentially be 3 times larger */
- /* so we need to create a new buffer */
- char utf8buf[(3 * bytesread) + 1];
-
- unicode_munge( tag, utf8buf, &bytesread );
-
- if(bytesread >= buffersize - bufferpos)
- bytesread = buffersize - bufferpos - 1;
-
- if ( /* Is it an embedded cuesheet? */
- (tr->tag_length == 4 && !memcmp(header, "TXXX", 4)) &&
- (bytesread >= 14 && !strncmp(utf8buf, "CUESHEET", 8))
- ) {
- unsigned char char_enc = 0;
- /* [enc type]+"CUESHEET\0" = 10 */
- unsigned char cuesheet_offset = 10;
- switch (tag[0]) {
- case 0x00:
- char_enc = CHAR_ENC_ISO_8859_1;
- break;
- case 0x01:
- tag++;
- if (!memcmp(tag,
- BOM_UTF_16_BE, BOM_UTF_16_SIZE)) {
- char_enc = CHAR_ENC_UTF_16_BE;
- } else if (!memcmp(tag,
- BOM_UTF_16_LE, BOM_UTF_16_SIZE)) {
- char_enc = CHAR_ENC_UTF_16_LE;
- }
- /* \1 + BOM(2) + C0U0E0S0H0E0E0T000 = 21 */
- cuesheet_offset = 21;
- break;
- case 0x02:
- char_enc = CHAR_ENC_UTF_16_BE;
- /* \2 + 0C0U0E0S0H0E0E0T00 = 19 */
- cuesheet_offset = 19;
- break;
- case 0x03:
- char_enc = CHAR_ENC_UTF_8;
- break;
- }
- if (char_enc > 0) {
- entry->has_embedded_cuesheet = true;
- entry->embedded_cuesheet.pos = lseek(fd, 0, SEEK_CUR)
- - framelen + cuesheet_offset;
- entry->embedded_cuesheet.size = totframelen
- - cuesheet_offset;
- entry->embedded_cuesheet.encoding = char_enc;
- }
- break;
- }
-
- for (j = 0; j < bytesread; j++)
- tag[j] = utf8buf[j];
-
- /* remove trailing spaces */
- while ( bytesread > 0 && isspace(tag[bytesread-1]))
- bytesread--;
- }
-
- if(bytesread == 0)
- /* Skip empty frames */
- break;
-
- tag[bytesread] = 0;
- bufferpos += bytesread + 1;
-
-#if CONFIG_CODEC == SWCODEC
- /* parse the tag if it contains iTunes gapless info */
- if (itunes_gapless)
- {
- itunes_gapless = false;
- entry->lead_trim = get_itunes_int32(tag, 1);
- entry->tail_trim = get_itunes_int32(tag, 2);
- }
-#endif
-
- /* Note that parser functions sometimes set *ptag to NULL, so
- * the "!*ptag" check here doesn't always have the desired
- * effect. Should the parser functions (parsegenre in
- * particular) be updated to handle the case of being called
- * multiple times, or should the "*ptag" check be removed?
- */
- if (ptag && !*ptag)
- *ptag = tag;
-
-#ifdef HAVE_ALBUMART
- /* albumart */
- if ((!entry->has_embedded_albumart) &&
- ((tr->tag_length == 4 && !memcmp( header, "APIC", 4)) ||
- (tr->tag_length == 3 && !memcmp( header, "PIC" , 3))))
- {
- if (unsynch || (global_unsynch && version <= ID3_VER_2_3))
- entry->albumart.type = AA_TYPE_UNSYNC;
- else
- {
- entry->albumart.pos = lseek(fd, 0, SEEK_CUR) - framelen;
- entry->albumart.size = totframelen;
- entry->albumart.type = AA_TYPE_UNKNOWN;
- }
- }
-#endif
- if( tr->ppFunc )
- bufferpos = tr->ppFunc(entry, tag, bufferpos);
- break;
- }
- }
-
- if( i == TAGLIST_SIZE ) {
- /* no tag in tagList was found, or it was a repeat.
- skip it using the total size */
-
- if(global_unsynch && version <= ID3_VER_2_3) {
- size -= skip_unsynched(fd, totframelen);
- } else {
- size -= totframelen;
- if( lseek(fd, totframelen, SEEK_CUR) == -1 )
- return;
- }
- } else {
- /* Seek to the next frame */
- if(framelen < totframelen)
- lseek(fd, totframelen - framelen, SEEK_CUR);
- }
- }
-}
-
-/*
- * Calculates the size of the ID3v2 tag.
- *
- * Arguments: file - the file to search for a tag.
- *
- * Returns: the size of the tag or 0 if none was found
- */
-int getid3v2len(int fd)
-{
- char buf[6];
- int offset;
-
- /* Make sure file has a ID3 tag */
- if((-1 == lseek(fd, 0, SEEK_SET)) ||
- (read(fd, buf, 6) != 6) ||
- (strncmp(buf, "ID3", strlen("ID3")) != 0))
- offset = 0;
-
- /* Now check what the ID3v2 size field says */
- else
- if(read(fd, buf, 4) != 4)
- offset = 0;
- else
- offset = unsync(buf[0], buf[1], buf[2], buf[3]) + 10;
-
- logf("ID3V2 Length: 0x%x", offset);
- return offset;
-}
-
-#ifdef DEBUG_STANDALONE
-
-char *secs2str(int ms)
-{
- static char buffer[32];
- int secs = ms/1000;
- ms %= 1000;
- snprintf(buffer, sizeof(buffer), "%d:%02d.%d", secs/60, secs%60, ms/100);
- return buffer;
-}
-
-int main(int argc, char **argv)
-{
- int i;
- for(i=1; i<argc; i++) {
- struct mp3entry mp3;
- mp3.album = "Bogus";
- if(mp3info(&mp3, argv[i], false)) {
- printf("Failed to get %s\n", argv[i]);
- return 0;
- }
-
- printf("****** File: %s\n"
- " Title: %s\n"
- " Artist: %s\n"
- " Album: %s\n"
- " Genre: %s (%d) \n"
- " Composer: %s\n"
- " Year: %s (%d)\n"
- " Track: %s (%d)\n"
- " Length: %s / %d s\n"
- " Bitrate: %d\n"
- " Frequency: %d\n",
- argv[i],
- mp3.title?mp3.title:"<blank>",
- mp3.artist?mp3.artist:"<blank>",
- mp3.album?mp3.album:"<blank>",
- mp3.genre_string?mp3.genre_string:"<blank>",
- mp3.genre,
- mp3.composer?mp3.composer:"<blank>",
- mp3.year_string?mp3.year_string:"<blank>",
- mp3.year,
- mp3.track_string?mp3.track_string:"<blank>",
- mp3.tracknum,
- secs2str(mp3.length),
- mp3.length/1000,
- mp3.bitrate,
- mp3.frequency);
- }
-
- return 0;
-}
-
-#endif
diff --git a/apps/metadata/kss.c b/apps/metadata/kss.c
deleted file mode 100644
index 2ae0cf50b0..0000000000
--- a/apps/metadata/kss.c
+++ /dev/null
@@ -1,53 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-static bool parse_kss_header(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
-
- lseek(fd, 0, SEEK_SET);
- if (read(fd, buf, 0x20) < 0x20)
- return false;
-
- /* calculate track length with number of tracks */
- id3->length = 0;
- if (buf[14] == 0x10) {
- id3->length = (get_short_le((void *)(buf + 26)) + 1) * 1000;
- }
-
- if (id3->length <= 0)
- id3->length = 255 * 1000; /* 255 tracks */
-
- return true;
-}
-
-
-bool get_kss_metadata(int fd, struct mp3entry* id3)
-{
- uint32_t kss_type;
- if ((lseek(fd, 0, SEEK_SET) < 0) ||
- read_uint32be(fd, &kss_type) != (int)sizeof(kss_type))
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
- /* we only render 16 bits, 44.1KHz, Stereo */
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* Make sure this is an SGC file */
- if (kss_type != FOURCC('K','S','C','C') && kss_type != FOURCC('K','S','S','X'))
- return false;
-
- return parse_kss_header(fd, id3);
-}
diff --git a/apps/metadata/metadata_common.c b/apps/metadata/metadata_common.c
deleted file mode 100644
index e861644025..0000000000
--- a/apps/metadata/metadata_common.c
+++ /dev/null
@@ -1,374 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include "string-extra.h"
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "replaygain.h"
-
-/* Read a string from the file. Read up to size bytes, or, if eos != -1,
- * until the eos character is found (eos is not stored in buf, unless it is
- * nil). Writes up to buf_size chars to buf, always terminating with a nil.
- * Returns number of chars read or -1 on read error.
- */
-long read_string(int fd, char* buf, long buf_size, int eos, long size)
-{
- long read_bytes = 0;
- char c;
-
- while (size != 0)
- {
- if (read(fd, &c, 1) != 1)
- {
- read_bytes = -1;
- break;
- }
-
- read_bytes++;
- size--;
-
- if ((eos != -1) && (eos == (unsigned char) c))
- {
- break;
- }
-
- if (buf_size > 1)
- {
- *buf++ = c;
- buf_size--;
- }
- }
-
- *buf = 0;
- return read_bytes;
-}
-/* Read an unsigned 8-bit integer from a file. */
-int read_uint8(int fd, uint8_t* buf)
-{
- size_t n;
-
- n = read(fd, (char*) buf, 1);
- return n;
-}
-
-#ifdef ROCKBOX_LITTLE_ENDIAN
-/* Read an unsigned 16-bit integer from a big-endian file. */
-int read_uint16be(int fd, uint16_t* buf)
-{
- size_t n;
-
- n = read(fd, (char*) buf, 2);
- *buf = betoh16(*buf);
- return n;
-}
-/* Read an unsigned 32-bit integer from a big-endian file. */
-int read_uint32be(int fd, uint32_t* buf)
-{
- size_t n;
-
- n = read(fd, (char*) buf, 4);
- *buf = betoh32(*buf);
- return n;
-}
-/* Read an unsigned 64-bit integer from a big-endian file. */
-int read_uint64be(int fd, uint64_t* buf)
-{
- size_t n;
- uint8_t data[8];
- int i;
-
- n = read(fd, data, 8);
-
- for (i=0, *buf=0; i<=7; i++) {
- *buf <<= 8;
- *buf |= data[i];
- }
- return n;
-}
-#else
-/* Read unsigned integers from a little-endian file. */
-int read_uint16le(int fd, uint16_t* buf)
-{
- size_t n;
-
- n = read(fd, (char*) buf, 2);
- *buf = letoh16(*buf);
- return n;
-}
-int read_uint32le(int fd, uint32_t* buf)
-{
- size_t n;
-
- n = read(fd, (char*) buf, 4);
- *buf = letoh32(*buf);
- return n;
-}
-int read_uint64le(int fd, uint64_t* buf)
-{
- size_t n;
- uint8_t data[8];
- int i;
-
- n = read(fd, data, 8);
-
- for (i=7, *buf=0; i>=0; i--) {
- *buf <<= 8;
- *buf |= data[i];
- }
-
- return n;
-}
-#endif
-
-/* Read an unaligned 64-bit little endian unsigned integer from buffer. */
-uint64_t get_uint64_le(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24) | ((uint64_t)p[4] << 32) |
- ((uint64_t)p[5] << 40) | ((uint64_t)p[6] << 48) | ((uint64_t)p[7] << 56);
-}
-
-/* Read an unaligned 32-bit little endian long from buffer. */
-uint32_t get_long_le(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24);
-}
-
-/* Read an unaligned 16-bit little endian short from buffer. */
-uint16_t get_short_le(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return p[0] | (p[1] << 8);
-}
-
-/* Read an unaligned 32-bit big endian long from buffer. */
-uint32_t get_long_be(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
-}
-
-/* Read an unaligned 16-bit little endian short from buffer. */
-uint16_t get_short_be(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return (p[0] << 8) | p[1];
-}
-
-/* Read an unaligned 32-bit little endian long from buffer. */
-int32_t get_slong(void* buf)
-{
- unsigned char* p = (unsigned char*) buf;
-
- return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24);
-}
-
-uint32_t get_itunes_int32(char* value, int count)
-{
- static const char hexdigits[] = "0123456789ABCDEF";
- const char* c;
- int r = 0;
-
- while (count-- > 0)
- {
- while (isspace(*value))
- {
- value++;
- }
-
- while (*value && !isspace(*value))
- {
- value++;
- }
- }
-
- while (isspace(*value))
- {
- value++;
- }
-
- while (*value && ((c = strchr(hexdigits, toupper(*value))) != NULL))
- {
- r = (r << 4) | (c - hexdigits);
- value++;
- }
-
- return r;
-}
-
-/* Skip an ID3v2 tag if it can be found. We assume the tag is located at the
- * start of the file, which should be true in all cases where we need to skip it.
- * Returns true if successfully skipped or not skipped, and false if
- * something went wrong while skipping.
- */
-bool skip_id3v2(int fd, struct mp3entry *id3)
-{
- char buf[4];
-
- read(fd, buf, 4);
- if (memcmp(buf, "ID3", 3) == 0)
- {
- /* We have found an ID3v2 tag at the start of the file - find its
- length and then skip it. */
- if ((id3->first_frame_offset = getid3v2len(fd)) == 0)
- return false;
-
- if ((lseek(fd, id3->first_frame_offset, SEEK_SET) < 0))
- return false;
-
- return true;
- } else {
- lseek(fd, 0, SEEK_SET);
- id3->first_frame_offset = 0;
- return true;
- }
-}
-
-/* Parse the tag (the name-value pair) and fill id3 and buffer accordingly.
- * String values to keep are written to buf. Returns number of bytes written
- * to buf (including end nil).
- */
-long parse_tag(const char* name, char* value, struct mp3entry* id3,
- char* buf, long buf_remaining, enum tagtype type)
-{
- long len = 0;
- char** p;
-
- if ((((strcasecmp(name, "track") == 0) && (type == TAGTYPE_APE)))
- || ((strcasecmp(name, "tracknumber") == 0) && (type == TAGTYPE_VORBIS)))
- {
- id3->tracknum = atoi(value);
- p = &(id3->track_string);
- }
- else if (strcasecmp(name, "discnumber") == 0 || strcasecmp(name, "disc") == 0)
- {
- id3->discnum = atoi(value);
- p = &(id3->disc_string);
- }
- else if (((strcasecmp(name, "year") == 0) && (type == TAGTYPE_APE))
- || ((strcasecmp(name, "date") == 0) && (type == TAGTYPE_VORBIS)))
- {
- /* Date's can be in any format in Vorbis. However most of them
- * are in ISO8601 format so if we try and parse the first part
- * of the tag as a number, we should get the year. If we get crap,
- * then act like we never parsed it.
- */
- id3->year = atoi(value);
- if (id3->year < 1900)
- { /* yeah, not likely */
- id3->year = 0;
- }
- p = &(id3->year_string);
- }
- else if (strcasecmp(name, "title") == 0)
- {
- p = &(id3->title);
- }
- else if (strcasecmp(name, "artist") == 0)
- {
- p = &(id3->artist);
- }
- else if (strcasecmp(name, "album") == 0)
- {
- p = &(id3->album);
- }
- else if (strcasecmp(name, "genre") == 0)
- {
- p = &(id3->genre_string);
- }
- else if (strcasecmp(name, "composer") == 0)
- {
- p = &(id3->composer);
- }
- else if (strcasecmp(name, "comment") == 0)
- {
- p = &(id3->comment);
- }
- else if (strcasecmp(name, "albumartist") == 0)
- {
- p = &(id3->albumartist);
- }
- else if (strcasecmp(name, "album artist") == 0)
- {
- p = &(id3->albumartist);
- }
- else if (strcasecmp(name, "ensemble") == 0)
- {
- p = &(id3->albumartist);
- }
- else if (strcasecmp(name, "grouping") == 0)
- {
- p = &(id3->grouping);
- }
- else if (strcasecmp(name, "content group") == 0)
- {
- p = &(id3->grouping);
- }
- else if (strcasecmp(name, "contentgroup") == 0)
- {
- p = &(id3->grouping);
- }
- else if (strcasecmp(name, "musicbrainz_trackid") == 0
- || strcasecmp(name, "http://musicbrainz.org") == 0 )
- {
- p = &(id3->mb_track_id);
- }
- else
- {
- parse_replaygain(name, value, id3);
- p = NULL;
- }
-
- /* Do not overwrite already available metadata. Especially when reading
- * tags with e.g. multiple genres / artists. This way only the first
- * of multiple entries is used, all following are dropped. */
- if (p!=NULL && *p==NULL)
- {
- len = strlen(value);
- len = MIN(len, buf_remaining - 1);
- len = MIN(len, ID3V2_MAX_ITEM_SIZE); /* Limit max. item size. */
-
- if (len > 0)
- {
- len++;
- strlcpy(buf, value, len);
- *p = buf;
- }
- else
- {
- len = 0;
- }
- }
-
- return len;
-}
diff --git a/apps/metadata/metadata_common.h b/apps/metadata/metadata_common.h
deleted file mode 100644
index db91729de4..0000000000
--- a/apps/metadata/metadata_common.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <inttypes.h>
-#include "metadata.h"
-
-#ifdef ROCKBOX_BIG_ENDIAN
-#define IS_BIG_ENDIAN 1
-#else
-#define IS_BIG_ENDIAN 0
-#endif
-
-#define TAG_NAME_LENGTH 32
-#define TAG_VALUE_LENGTH 128
-
-#define FOURCC(a,b,c,d) (((a)<<24) | ((b) << 16) | ((c) << 8) | (d))
-
-enum tagtype { TAGTYPE_APE = 1, TAGTYPE_VORBIS };
-
-bool read_ape_tags(int fd, struct mp3entry* id3);
-long read_vorbis_tags(int fd, struct mp3entry *id3,
- long tag_remaining);
-
-bool skip_id3v2(int fd, struct mp3entry *id3);
-long read_string(int fd, char* buf, long buf_size, int eos, long size);
-
-int read_uint8(int fd, uint8_t* buf);
-#ifdef ROCKBOX_BIG_ENDIAN
-#define read_uint16be(fd,buf) read((fd), (buf), 2)
-#define read_uint32be(fd,buf) read((fd), (buf), 4)
-#define read_uint64be(fd,buf) read((fd), (buf), 8)
-int read_uint16le(int fd, uint16_t* buf);
-int read_uint32le(int fd, uint32_t* buf);
-int read_uint64le(int fd, uint64_t* buf);
-#else
-int read_uint16be(int fd, uint16_t* buf);
-int read_uint32be(int fd, uint32_t* buf);
-int read_uint64be(int fd, uint64_t* buf);
-#define read_uint16le(fd,buf) read((fd), (buf), 2)
-#define read_uint32le(fd,buf) read((fd), (buf), 4)
-#define read_uint64le(fd,buf) read((fd), (buf), 8)
-#endif
-
-uint64_t get_uint64_le(void* buf);
-uint32_t get_long_le(void* buf);
-uint16_t get_short_le(void* buf);
-uint32_t get_long_be(void* buf);
-uint16_t get_short_be(void* buf);
-int32_t get_slong(void* buf);
-uint32_t get_itunes_int32(char* value, int count);
-long parse_tag(const char* name, char* value, struct mp3entry* id3,
- char* buf, long buf_remaining, enum tagtype type);
diff --git a/apps/metadata/metadata_parsers.h b/apps/metadata/metadata_parsers.h
deleted file mode 100644
index 304e393538..0000000000
--- a/apps/metadata/metadata_parsers.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#if CONFIG_CODEC == SWCODEC
-char* id3_get_num_genre(unsigned int genre_num);
-#endif
-int getid3v2len(int fd);
-bool setid3v1title(int fd, struct mp3entry *entry);
-void setid3v2title(int fd, struct mp3entry *entry);
-bool get_mp3_metadata(int fd, struct mp3entry* id3);
-#if CONFIG_CODEC == SWCODEC
-bool get_adx_metadata(int fd, struct mp3entry* id3);
-bool get_aiff_metadata(int fd, struct mp3entry* id3);
-bool get_flac_metadata(int fd, struct mp3entry* id3);
-bool get_mp4_metadata(int fd, struct mp3entry* id3);
-bool get_monkeys_metadata(int fd, struct mp3entry* id3);
-bool get_musepack_metadata(int fd, struct mp3entry *id3);
-bool get_sid_metadata(int fd, struct mp3entry* id3);
-bool get_mod_metadata(int fd, struct mp3entry* id3);
-bool get_spc_metadata(int fd, struct mp3entry* id3);
-bool get_ogg_metadata(int fd, struct mp3entry* id3);
-bool get_wave_metadata(int fd, struct mp3entry* id3);
-bool get_wavpack_metadata(int fd, struct mp3entry* id3);
-bool get_a52_metadata(int fd, struct mp3entry* id3);
-bool get_asf_metadata(int fd, struct mp3entry* id3);
-bool get_asap_metadata(int fd, struct mp3entry* id3);
-bool get_rm_metadata(int fd, struct mp3entry* id3);
-bool get_nsf_metadata(int fd, struct mp3entry* id3);
-bool get_oma_metadata(int fd, struct mp3entry* id3);
-bool get_smaf_metadata(int fd, struct mp3entry* id3);
-bool get_au_metadata(int fd, struct mp3entry* id3);
-bool get_vox_metadata(int fd, struct mp3entry* id3);
-bool get_wave64_metadata(int fd, struct mp3entry* id3);
-bool get_tta_metadata(int fd, struct mp3entry* id3);
-bool get_ay_metadata(int fd, struct mp3entry* id3);
-bool get_gbs_metadata(int fd, struct mp3entry* id3);
-bool get_hes_metadata(int fd, struct mp3entry* id3);
-bool get_sgc_metadata(int fd, struct mp3entry* id3);
-bool get_vgm_metadata(int fd, struct mp3entry* id3);
-bool get_kss_metadata(int fd, struct mp3entry* id3);
-#endif /* CONFIG_CODEC == SWCODEC */
diff --git a/apps/metadata/mod.c b/apps/metadata/mod.c
deleted file mode 100644
index de76823e91..0000000000
--- a/apps/metadata/mod.c
+++ /dev/null
@@ -1,103 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include <string-extra.h>
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-#define MODULEHEADERSIZE 0x438
-
-bool get_mod_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char *buf = id3->id3v2buf;
- unsigned char id[4];
- bool is_mod_file = false;
-
- /* Seek to file begin */
- if (lseek(fd, 0, SEEK_SET) < 0)
- return false;
- /* Use id3v2buf as buffer for the track name */
- if (read(fd, buf, sizeof(id3->id3v2buf)) < (ssize_t)sizeof(id3->id3v2buf))
- return false;
- /* Seek to MOD ID position */
- if (lseek(fd, MODULEHEADERSIZE, SEEK_SET) < 0)
- return false;
- /* Read MOD ID */
- if (read(fd, id, sizeof(id)) < (ssize_t)sizeof(id))
- return false;
-
- /* Mod type checking based on MikMod */
- /* Protracker and variants */
- if ((!memcmp(id, "M.K.", 4)) || (!memcmp(id, "M!K!", 4))) {
- is_mod_file = true;
- }
-
- /* Star Tracker */
- if (((!memcmp(id, "FLT", 3)) || (!memcmp(id, "EXO", 3))) &&
- (isdigit(id[3]))) {
- char numchn = id[3] - '0';
- if (numchn == 4 || numchn == 8)
- is_mod_file = true;
- }
-
- /* Oktalyzer (Amiga) */
- if (!memcmp(id, "OKTA", 4)) {
- is_mod_file = true;
- }
-
- /* Oktalyser (Atari) */
- if (!memcmp(id, "CD81", 4)) {
- is_mod_file = true;
- }
-
- /* Fasttracker */
- if ((!memcmp(id + 1, "CHN", 3)) && (isdigit(id[0]))) {
- is_mod_file = true;
- }
- /* Fasttracker or Taketracker */
- if (((!memcmp(id + 2, "CH", 2)) || (!memcmp(id + 2, "CN", 2)))
- && (isdigit(id[0])) && (isdigit(id[1]))) {
- is_mod_file = true;
- }
-
- /* Don't try to play if we can't find a known mod type
- * (there are mod files which have nothing to do with music) */
- if (!is_mod_file)
- return false;
-
- id3->title = id3->id3v2buf; /* Point title to previous read ID3 buffer. */
- id3->bitrate = filesize(fd)/1024; /* size in kb */
- id3->frequency = 44100;
- id3->length = 120*1000;
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- return true;
-}
-
diff --git a/apps/metadata/monkeys.c b/apps/metadata/monkeys.c
deleted file mode 100644
index 4aff1412aa..0000000000
--- a/apps/metadata/monkeys.c
+++ /dev/null
@@ -1,97 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2007 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-
-bool get_monkeys_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- unsigned char* header;
- bool rc = false;
- uint32_t descriptorlength;
- uint32_t totalsamples;
- uint32_t blocksperframe, finalframeblocks, totalframes;
- int fileversion;
-
- lseek(fd, 0, SEEK_SET);
-
- if (read(fd, buf, 4) < 4)
- {
- return rc;
- }
-
- if (memcmp(buf, "MAC ", 4) != 0)
- {
- return rc;
- }
-
- read(fd, buf + 4, MAX_PATH - 4);
-
- fileversion = get_short_le(buf+4);
- if (fileversion < 3970)
- {
- /* Not supported */
- return false;
- }
-
- if (fileversion >= 3980)
- {
- descriptorlength = get_long_le(buf+8);
-
- header = buf + descriptorlength;
-
- blocksperframe = get_long_le(header+4);
- finalframeblocks = get_long_le(header+8);
- totalframes = get_long_le(header+12);
- id3->frequency = get_long_le(header+20);
- }
- else
- {
- /* v3.95 and later files all have a fixed framesize */
- blocksperframe = 73728 * 4;
-
- finalframeblocks = get_long_le(buf+28);
- totalframes = get_long_le(buf+24);
- id3->frequency = get_long_le(buf+12);
- }
-
- id3->vbr = true; /* All APE files are VBR */
- id3->filesize = filesize(fd);
-
- totalsamples = finalframeblocks;
- if (totalframes > 1)
- totalsamples += blocksperframe * (totalframes-1);
-
- id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
- id3->bitrate = (id3->filesize * 8) / id3->length;
-
- read_ape_tags(fd, id3);
- return true;
-}
diff --git a/apps/metadata/mp3.c b/apps/metadata/mp3.c
deleted file mode 100644
index feb1a52f77..0000000000
--- a/apps/metadata/mp3.c
+++ /dev/null
@@ -1,193 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2002 by Daniel Stenberg
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-/*
- * Parts of this code has been stolen from the Ample project and was written
- * by David H�deman. It has since been extended and enhanced pretty much by
- * all sorts of friendly Rockbox people.
- *
- */
-
- /* tagResolver and associated code copyright 2003 Thomas Paul Diffenbach
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <stdbool.h>
-#include "string-extra.h"
-#include "config.h"
-#include "file.h"
-#include "logf.h"
-
-#include "system.h"
-#include "metadata.h"
-#include "mp3data.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-
-/*
- * Calculates the length (in milliseconds) of an MP3 file.
- *
- * Modified to only use integers.
- *
- * Arguments: file - the file to calculate the length upon
- * entry - the entry to update with the length
- *
- * Returns: the song length in milliseconds,
- * 0 means that it couldn't be calculated
- */
-static int getsonglength(int fd, struct mp3entry *entry)
-{
- unsigned long filetime = 0;
- struct mp3info info;
- long bytecount;
-
- /* Start searching after ID3v2 header */
- if(-1 == lseek(fd, entry->id3v2len, SEEK_SET))
- return 0;
-
- bytecount = get_mp3file_info(fd, &info);
-
- logf("Space between ID3V2 tag and first audio frame: 0x%lx bytes",
- bytecount);
-
- if(bytecount < 0)
- return -1;
-
- bytecount += entry->id3v2len;
-
- /* Validate byte count, in case the file has been edited without
- * updating the header.
- */
- if (info.byte_count)
- {
- const unsigned long expected = entry->filesize - entry->id3v1len
- - entry->id3v2len;
- const unsigned long diff = MAX(10240, info.byte_count / 20);
-
- if ((info.byte_count > expected + diff)
- || (info.byte_count < expected - diff))
- {
- logf("Note: info.byte_count differs from expected value by "
- "%ld bytes", labs((long) (expected - info.byte_count)));
- info.byte_count = 0;
- info.frame_count = 0;
- info.file_time = 0;
- info.enc_padding = 0;
-
- /* Even if the bitrate was based on "known bad" values, it
- * should still be better for VBR files than using the bitrate
- * of the first audio frame.
- */
- }
- }
-
- entry->bitrate = info.bitrate;
- entry->frequency = info.frequency;
- entry->layer = info.layer;
- switch(entry->layer) {
-#if CONFIG_CODEC==SWCODEC
- case 0:
- entry->codectype=AFMT_MPA_L1;
- break;
-#endif
- case 1:
- entry->codectype=AFMT_MPA_L2;
- break;
- case 2:
- entry->codectype=AFMT_MPA_L3;
- break;
- }
-
- /* If the file time hasn't been established, this may be a fixed
- rate MP3, so just use the default formula */
-
- filetime = info.file_time;
-
- if(filetime == 0)
- {
- /* Prevent a division by zero */
- if (info.bitrate < 8)
- filetime = 0;
- else
- filetime = (entry->filesize - bytecount) / (info.bitrate / 8);
- /* bitrate is in kbps so this delivers milliseconds. Doing bitrate / 8
- * instead of filesize * 8 is exact, because mpeg audio bitrates are
- * always multiples of 8, and it avoids overflows. */
- }
-
- entry->frame_count = info.frame_count;
-
- entry->vbr = info.is_vbr;
- entry->has_toc = info.has_toc;
-
-#if CONFIG_CODEC==SWCODEC
- if (!entry->lead_trim)
- entry->lead_trim = info.enc_delay;
- if (!entry->tail_trim)
- entry->tail_trim = info.enc_padding;
-#endif
-
- memcpy(entry->toc, info.toc, sizeof(info.toc));
-
- /* Update the seek point for the first playable frame */
- entry->first_frame_offset = bytecount;
- logf("First frame is at %lx", entry->first_frame_offset);
-
- return filetime;
-}
-
-/*
- * Checks all relevant information (such as ID3v1 tag, ID3v2 tag, length etc)
- * about an MP3 file and updates it's entry accordingly.
- *
- Note, that this returns true for successful, false for error! */
-bool get_mp3_metadata(int fd, struct mp3entry *entry)
-{
- entry->title = NULL;
- entry->filesize = filesize(fd);
- entry->id3v2len = getid3v2len(fd);
- entry->tracknum = 0;
- entry->discnum = 0;
-
- if (entry->id3v2len)
- setid3v2title(fd, entry);
- int len = getsonglength(fd, entry);
- if (len < 0)
- return false;
- entry->length = len;
-
- /* Subtract the meta information from the file size to get
- the true size of the MP3 stream */
- entry->filesize -= entry->first_frame_offset;
-
- /* only seek to end of file if no id3v2 tags were found */
- if (!entry->id3v2len) {
- setid3v1title(fd, entry);
- }
-
- if(!entry->length || (entry->filesize < 8 ))
- /* no song length or less than 8 bytes is hereby considered to be an
- invalid mp3 and won't be played by us! */
- return false;
-
- return true;
-}
diff --git a/apps/metadata/mp4.c b/apps/metadata/mp4.c
deleted file mode 100644
index df164436f5..0000000000
--- a/apps/metadata/mp4.c
+++ /dev/null
@@ -1,842 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Magnus Holmgren
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "errno.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-#include "debug.h"
-#include "replaygain.h"
-
-#ifdef DEBUGF
-#undef DEBUGF
-#define DEBUGF(...)
-#endif
-
-#define MP4_3gp6 FOURCC('3', 'g', 'p', '6')
-#define MP4_aART FOURCC('a', 'A', 'R', 'T')
-#define MP4_alac FOURCC('a', 'l', 'a', 'c')
-#define MP4_calb FOURCC(0xa9, 'a', 'l', 'b')
-#define MP4_cART FOURCC(0xa9, 'A', 'R', 'T')
-#define MP4_cgrp FOURCC(0xa9, 'g', 'r', 'p')
-#define MP4_cgen FOURCC(0xa9, 'g', 'e', 'n')
-#define MP4_chpl FOURCC('c', 'h', 'p', 'l')
-#define MP4_cnam FOURCC(0xa9, 'n', 'a', 'm')
-#define MP4_cwrt FOURCC(0xa9, 'w', 'r', 't')
-#define MP4_ccmt FOURCC(0xa9, 'c', 'm', 't')
-#define MP4_cday FOURCC(0xa9, 'd', 'a', 'y')
-#define MP4_covr FOURCC('c', 'o', 'v', 'r')
-#define MP4_disk FOURCC('d', 'i', 's', 'k')
-#define MP4_esds FOURCC('e', 's', 'd', 's')
-#define MP4_ftyp FOURCC('f', 't', 'y', 'p')
-#define MP4_gnre FOURCC('g', 'n', 'r', 'e')
-#define MP4_hdlr FOURCC('h', 'd', 'l', 'r')
-#define MP4_ilst FOURCC('i', 'l', 's', 't')
-#define MP4_isom FOURCC('i', 's', 'o', 'm')
-#define MP4_M4A FOURCC('M', '4', 'A', ' ')
-#define MP4_m4a FOURCC('m', '4', 'a', ' ') /*technically its "M4A "*/
-#define MP4_M4B FOURCC('M', '4', 'B', ' ') /*but files exist with lower case*/
-#define MP4_mdat FOURCC('m', 'd', 'a', 't')
-#define MP4_mdia FOURCC('m', 'd', 'i', 'a')
-#define MP4_mdir FOURCC('m', 'd', 'i', 'r')
-#define MP4_meta FOURCC('m', 'e', 't', 'a')
-#define MP4_minf FOURCC('m', 'i', 'n', 'f')
-#define MP4_moov FOURCC('m', 'o', 'o', 'v')
-#define MP4_mp4a FOURCC('m', 'p', '4', 'a')
-#define MP4_mp42 FOURCC('m', 'p', '4', '2')
-#define MP4_qt FOURCC('q', 't', ' ', ' ')
-#define MP4_soun FOURCC('s', 'o', 'u', 'n')
-#define MP4_stbl FOURCC('s', 't', 'b', 'l')
-#define MP4_stsd FOURCC('s', 't', 's', 'd')
-#define MP4_stts FOURCC('s', 't', 't', 's')
-#define MP4_trak FOURCC('t', 'r', 'a', 'k')
-#define MP4_trkn FOURCC('t', 'r', 'k', 'n')
-#define MP4_udta FOURCC('u', 'd', 't', 'a')
-#define MP4_extra FOURCC('-', '-', '-', '-')
-
-/* Read the tag data from an MP4 file, storing up to buffer_size bytes in
- * buffer.
- */
-static unsigned long read_mp4_tag(int fd, unsigned int size_left, char* buffer,
- unsigned int buffer_left)
-{
- unsigned int bytes_read = 0;
-
- if (buffer_left == 0)
- {
- lseek(fd, size_left, SEEK_CUR); /* Skip everything */
- }
- else
- {
- /* Skip the data tag header - maybe we should parse it properly? */
- lseek(fd, 16, SEEK_CUR);
- size_left -= 16;
-
- if (size_left > buffer_left)
- {
- read(fd, buffer, buffer_left);
- lseek(fd, size_left - buffer_left, SEEK_CUR);
- bytes_read = buffer_left;
- }
- else
- {
- read(fd, buffer, size_left);
- bytes_read = size_left;
- }
- }
-
- return bytes_read;
-}
-
-/* Read a string tag from an MP4 file */
-static unsigned int read_mp4_tag_string(int fd, int size_left, char** buffer,
- unsigned int* buffer_left, char** dest)
-{
- unsigned int bytes_read = read_mp4_tag(fd, size_left, *buffer,
- *buffer_left > 0 ? *buffer_left - 1 : 0);
- unsigned int length = 0;
-
- if (bytes_read)
- {
- /* Do not overwrite already available metadata. Especially when reading
- * tags with e.g. multiple genres / artists. This way only the first
- * of multiple entries is used, all following are dropped. */
- if (*dest == NULL)
- {
- (*buffer)[bytes_read] = 0; /* zero-terminate for correct strlen().*/
- length = strlen(*buffer) + 1;
- length = MIN(length, ID3V2_MAX_ITEM_SIZE); /* Limit item size. */
-
- *dest = *buffer;
- (*buffer)[length-1] = 0; /* zero-terminate buffer. */
- *buffer_left -= length;
- *buffer += length;
- }
- }
- else
- {
- *dest = NULL;
- }
-
- return length;
-}
-
-static unsigned int read_mp4_atom(int fd, uint32_t* size,
- uint32_t* type, uint32_t size_left)
-{
- read_uint32be(fd, size);
- read_uint32be(fd, type);
-
- if (*size == 1)
- {
- /* FAT32 doesn't support files this big, so something seems to
- * be wrong. (64-bit sizes should only be used when required.)
- */
- errno = EFBIG;
- *type = 0;
- return 0;
- }
-
- if (*size > 0)
- {
- if (*size > size_left)
- {
- size_left = 0;
- }
- else
- {
- size_left -= *size;
- }
-
- *size -= 8;
- }
- else
- {
- *size = size_left;
- size_left = 0;
- }
-
- return size_left;
-}
-
-static unsigned int read_mp4_length(int fd, uint32_t* size)
-{
- unsigned int length = 0;
- int bytes = 0;
- unsigned char c;
-
- do
- {
- read(fd, &c, 1);
- bytes++;
- (*size)--;
- length = (length << 7) | (c & 0x7F);
- }
- while ((c & 0x80) && (bytes < 4) && (*size > 0));
-
- return length;
-}
-
-static bool read_mp4_esds(int fd, struct mp3entry* id3, uint32_t* size)
-{
- unsigned char buf[8];
- bool sbr = false;
-
- lseek(fd, 4, SEEK_CUR); /* Version and flags. */
- read(fd, buf, 1); /* Verify ES_DescrTag. */
- *size -= 5;
-
- if (*buf == 3)
- {
- /* read length */
- if (read_mp4_length(fd, size) < 20)
- {
- return sbr;
- }
-
- lseek(fd, 3, SEEK_CUR);
- *size -= 3;
- }
- else
- {
- lseek(fd, 2, SEEK_CUR);
- *size -= 2;
- }
-
- read(fd, buf, 1); /* Verify DecoderConfigDescrTab. */
- *size -= 1;
-
- if (*buf != 4)
- {
- return sbr;
- }
-
- if (read_mp4_length(fd, size) < 13)
- {
- return sbr;
- }
-
- lseek(fd, 13, SEEK_CUR); /* Skip audio type, bit rates, etc. */
- read(fd, buf, 1);
- *size -= 14;
-
- if (*buf != 5) /* Verify DecSpecificInfoTag. */
- {
- return sbr;
- }
-
- {
- static const int sample_rates[] =
- {
- 96000, 88200, 64000, 48000, 44100, 32000,
- 24000, 22050, 16000, 12000, 11025, 8000
- };
- unsigned long bits;
- unsigned int length;
- unsigned int index;
- unsigned int type;
-
- /* Read the (leading part of the) decoder config. */
- length = read_mp4_length(fd, size);
- length = MIN(length, *size);
- length = MIN(length, sizeof(buf));
- memset(buf, 0, sizeof(buf));
- read(fd, buf, length);
- *size -= length;
-
- /* Maybe time to write a simple read_bits function... */
-
- /* Decoder config format:
- * Object type - 5 bits
- * Frequency index - 4 bits
- * Channel configuration - 4 bits
- */
- bits = get_long_be(buf);
- type = bits >> 27; /* Object type - 5 bits */
- index = (bits >> 23) & 0xf; /* Frequency index - 4 bits */
-
- if (index < (sizeof(sample_rates) / sizeof(*sample_rates)))
- {
- id3->frequency = sample_rates[index];
- }
-
- if (type == 5)
- {
- unsigned int old_index = index;
-
- sbr = true;
- index = (bits >> 15) & 0xf; /* Frequency index - 4 bits */
-
- if (index == 15)
- {
- /* 17 bits read so far... */
- bits = get_long_be(&buf[2]);
- id3->frequency = (bits >> 7) & 0x00ffffff;
- }
- else if (index < (sizeof(sample_rates) / sizeof(*sample_rates)))
- {
- id3->frequency = sample_rates[index];
- }
-
- if (old_index == index)
- {
- /* Downsampled SBR */
- id3->frequency *= 2;
- }
- }
- /* Skip 13 bits from above, plus 3 bits, then read 11 bits */
- else if ((length >= 4) && (((bits >> 5) & 0x7ff) == 0x2b7))
- {
- /* We found an extensionAudioObjectType */
- type = bits & 0x1f; /* Object type - 5 bits*/
- bits = get_long_be(&buf[4]);
-
- if (type == 5)
- {
- sbr = bits >> 31;
-
- if (sbr)
- {
- unsigned int old_index = index;
-
- /* 1 bit read so far */
- index = (bits >> 27) & 0xf; /* Frequency index - 4 bits */
-
- if (index == 15)
- {
- /* 5 bits read so far */
- id3->frequency = (bits >> 3) & 0x00ffffff;
- }
- else if (index < (sizeof(sample_rates) / sizeof(*sample_rates)))
- {
- id3->frequency = sample_rates[index];
- }
-
- if (old_index == index)
- {
- /* Downsampled SBR */
- id3->frequency *= 2;
- }
- }
- }
- }
-
- if (!sbr && (id3->frequency <= 24000) && (length <= 2))
- {
- /* Double the frequency for low-frequency files without a "long"
- * DecSpecificConfig header. The file may or may not contain SBR,
- * but here we guess it does if the header is short. This can
- * fail on some files, but it's the best we can do, short of
- * decoding (parts of) the file.
- */
- id3->frequency *= 2;
- sbr = true;
- }
- }
-
- return sbr;
-}
-
-static bool read_mp4_tags(int fd, struct mp3entry* id3,
- uint32_t size_left)
-{
- uint32_t size;
- uint32_t type;
- unsigned int buffer_left = sizeof(id3->id3v2buf) + sizeof(id3->id3v1buf);
- char* buffer = id3->id3v2buf;
- bool cwrt = false;
-
- do
- {
- size_left = read_mp4_atom(fd, &size, &type, size_left);
-
- /* DEBUGF("Tag atom: '%c%c%c%c' (%d bytes left)\n", type >> 24 & 0xff,
- type >> 16 & 0xff, type >> 8 & 0xff, type & 0xff, size); */
-
- switch (type)
- {
- case MP4_cnam:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->title);
- break;
-
- case MP4_cART:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->artist);
- break;
-
- case MP4_aART:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->albumartist);
- break;
-
- case MP4_cgrp:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->grouping);
- break;
-
- case MP4_calb:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->album);
- break;
-
- case MP4_cwrt:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->composer);
- cwrt = false;
- break;
-
- case MP4_ccmt:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->comment);
- break;
-
- case MP4_cday:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->year_string);
-
- /* Try to parse it as a year, for the benefit of the database.
- */
- if(id3->year_string)
- {
- id3->year = atoi(id3->year_string);
- if (id3->year < 1900)
- {
- id3->year = 0;
- }
- }
- else
- id3->year = 0;
-
- break;
-
- case MP4_gnre:
- {
- unsigned short genre;
-
- read_mp4_tag(fd, size, (char*) &genre, sizeof(genre));
- id3->genre_string = id3_get_num_genre(betoh16(genre) - 1);
- }
- break;
-
- case MP4_cgen:
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->genre_string);
- break;
-
- case MP4_disk:
- {
- unsigned short n[2];
-
- read_mp4_tag(fd, size, (char*) &n, sizeof(n));
- id3->discnum = betoh16(n[1]);
- }
- break;
-
- case MP4_trkn:
- {
- unsigned short n[2];
-
- read_mp4_tag(fd, size, (char*) &n, sizeof(n));
- id3->tracknum = betoh16(n[1]);
- }
- break;
-
-#ifdef HAVE_ALBUMART
- case MP4_covr:
- {
- int pos = lseek(fd, 0, SEEK_CUR) + 16;
-
- read_mp4_tag(fd, size, buffer, 8);
- id3->albumart.type = AA_TYPE_UNKNOWN;
- if (memcmp(buffer, "\xff\xd8\xff\xe0", 4) == 0)
- {
- id3->albumart.type = AA_TYPE_JPG;
- }
- else if (memcmp(buffer, "\x89\x50\x4e\x47\x0d\x0a\x1a\x0a", 8) == 0)
- {
- id3->albumart.type = AA_TYPE_PNG;
- }
-
- if (id3->albumart.type != AA_TYPE_UNKNOWN)
- {
- id3->albumart.pos = pos;
- id3->albumart.size = size - 16;
- id3->has_embedded_albumart = true;
- }
- }
- break;
-#endif
-
- case MP4_extra:
- {
- char tag_name[TAG_NAME_LENGTH];
- uint32_t sub_size;
-
- /* "mean" atom */
- read_uint32be(fd, &sub_size);
- size -= sub_size;
- lseek(fd, sub_size - 4, SEEK_CUR);
- /* "name" atom */
- read_uint32be(fd, &sub_size);
- size -= sub_size;
- lseek(fd, 8, SEEK_CUR);
- sub_size -= 12;
-
- if (sub_size > sizeof(tag_name) - 1)
- {
- read(fd, tag_name, sizeof(tag_name) - 1);
- lseek(fd, sub_size - (sizeof(tag_name) - 1), SEEK_CUR);
- tag_name[sizeof(tag_name) - 1] = 0;
- }
- else
- {
- read(fd, tag_name, sub_size);
- tag_name[sub_size] = 0;
- }
-
- if ((strcasecmp(tag_name, "composer") == 0) && !cwrt)
- {
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->composer);
- }
- else if (strcasecmp(tag_name, "iTunSMPB") == 0)
- {
- char value[TAG_VALUE_LENGTH];
- char* value_p = value;
- char* any;
- unsigned int length = sizeof(value);
-
- read_mp4_tag_string(fd, size, &value_p, &length, &any);
- id3->lead_trim = get_itunes_int32(value, 1);
- id3->tail_trim = get_itunes_int32(value, 2);
- DEBUGF("AAC: lead_trim %d, tail_trim %d\n",
- id3->lead_trim, id3->tail_trim);
- }
- else if (strcasecmp(tag_name, "musicbrainz track id") == 0)
- {
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->mb_track_id);
- }
- else if ((strcasecmp(tag_name, "album artist") == 0))
- {
- read_mp4_tag_string(fd, size, &buffer, &buffer_left,
- &id3->albumartist);
- }
- else
- {
- char* any = NULL;
- unsigned int length = read_mp4_tag_string(fd, size,
- &buffer, &buffer_left, &any);
-
- if (length > 0)
- {
- /* Re-use the read buffer as the dest buffer... */
- buffer -= length;
- buffer_left += length;
-
- parse_replaygain(tag_name, buffer, id3);
- }
- }
- }
- break;
-
- default:
- lseek(fd, size, SEEK_CUR);
- break;
- }
- }
- while ((size_left > 0) && (errno == 0));
-
- return true;
-}
-
-static bool read_mp4_container(int fd, struct mp3entry* id3,
- uint32_t size_left)
-{
- uint32_t size = 0;
- uint32_t type = 0;
- uint32_t handler = 0;
- bool rc = true;
- bool done = false;
-
- do
- {
- size_left = read_mp4_atom(fd, &size, &type, size_left);
-
- /* DEBUGF("Atom: '%c%c%c%c' (0x%08lx, %lu bytes left)\n",
- (int) ((type >> 24) & 0xff), (int) ((type >> 16) & 0xff),
- (int) ((type >> 8) & 0xff), (int) (type & 0xff),
- type, size); */
-
- switch (type)
- {
- case MP4_ftyp:
- {
- uint32_t id;
-
- read_uint32be(fd, &id);
- size -= 4;
-
- if ((id != MP4_M4A) && (id != MP4_M4B) && (id != MP4_mp42)
- && (id != MP4_qt) && (id != MP4_3gp6) && (id != MP4_m4a)
- && (id != MP4_isom))
- {
- DEBUGF("Unknown MP4 file type: '%c%c%c%c'\n",
- (int)(id >> 24 & 0xff), (int)(id >> 16 & 0xff),
- (int)(id >> 8 & 0xff), (int)(id & 0xff));
- return false;
- }
- }
- break;
-
- case MP4_meta:
- lseek(fd, 4, SEEK_CUR); /* Skip version */
- size -= 4;
- /* Fall through */
-
- case MP4_moov:
- case MP4_udta:
- case MP4_mdia:
- case MP4_stbl:
- case MP4_trak:
- rc = read_mp4_container(fd, id3, size);
- size = 0;
- break;
-
- case MP4_ilst:
- /* We need at least a size of 8 to read the next atom. */
- if (handler == MP4_mdir && size>8)
- {
- rc = read_mp4_tags(fd, id3, size);
- size = 0;
- }
- break;
-
- case MP4_minf:
- if (handler == MP4_soun)
- {
- rc = read_mp4_container(fd, id3, size);
- size = 0;
- }
- break;
-
- case MP4_stsd:
- lseek(fd, 8, SEEK_CUR);
- size -= 8;
- rc = read_mp4_container(fd, id3, size);
- size = 0;
- break;
-
- case MP4_hdlr:
- lseek(fd, 8, SEEK_CUR);
- read_uint32be(fd, &handler);
- size -= 12;
- /* DEBUGF(" Handler '%c%c%c%c'\n", handler >> 24 & 0xff,
- handler >> 16 & 0xff, handler >> 8 & 0xff,handler & 0xff); */
- break;
-
- case MP4_stts:
- {
- uint32_t entries;
- unsigned int i;
-
- /* Reset to false. */
- id3->needs_upsampling_correction = false;
-
- lseek(fd, 4, SEEK_CUR);
- read_uint32be(fd, &entries);
- id3->samples = 0;
-
- for (i = 0; i < entries; i++)
- {
- uint32_t n;
- uint32_t l;
-
- read_uint32be(fd, &n);
- read_uint32be(fd, &l);
-
- /* Some AAC file use HE profile. In this case the number
- * of output samples is doubled to a maximum of 2048
- * samples per frame. This means that files which already
- * report a frame size of 2048 in their header will not
- * need any further special handling. */
- if (id3->codectype==AFMT_MP4_AAC_HE && l<=1024)
- {
- id3->samples += n * l * 2;
- id3->needs_upsampling_correction = true;
- }
- else
- {
- id3->samples += n * l;
- }
- }
-
- size = 0;
- }
- break;
-
- case MP4_mp4a:
- {
- uint32_t subsize;
- uint32_t subtype;
-
- /* Move to the next expected mp4 atom. */
- lseek(fd, 28, SEEK_CUR);
- read_mp4_atom(fd, &subsize, &subtype, size);
- size -= 36;
-
- if (subtype == MP4_esds)
- {
- /* Read esds metadata and return if AAC-HE/SBR is used. */
- if (read_mp4_esds(fd, id3, &size))
- id3->codectype = AFMT_MP4_AAC_HE;
- else
- id3->codectype = AFMT_MP4_AAC;
- }
- }
- break;
-
- case MP4_alac:
- {
- uint32_t frequency;
- uint32_t subsize;
- uint32_t subtype;
-
- /* Move to the next expected mp4 atom. */
- lseek(fd, 28, SEEK_CUR);
- read_mp4_atom(fd, &subsize, &subtype, size);
- size -= 36;
-#if 0
- /* We might need to parse for the alac metadata atom. */
- while (!((subsize==28) && (subtype==MP4_alac)) && (size>0))
- {
- lseek(fd, -7, SEEK_CUR);
- read_mp4_atom(fd, &subsize, &subtype, size);
- size -= 1;
- errno = 0; /* will most likely be set while parsing */
- }
-#endif
- if (subtype == MP4_alac)
- {
- lseek(fd, 24, SEEK_CUR);
- read_uint32be(fd, &frequency);
- size -= 28;
- id3->frequency = frequency;
- id3->codectype = AFMT_MP4_ALAC;
- }
- }
- break;
-
- case MP4_mdat:
- /* Some AAC files appear to contain additional empty mdat chunks.
- Ignore them. */
- if(size == 0)
- break;
- id3->filesize = size;
- if(id3->samples > 0) {
- /* We've already seen the moov chunk. */
- done = true;
- }
- break;
-
- case MP4_chpl:
- {
- /* ADDME: add support for real chapters. Right now it's only
- * used for Nero's gapless hack */
- uint8_t chapters;
- uint64_t timestamp;
-
- lseek(fd, 8, SEEK_CUR);
- read_uint8(fd, &chapters);
- size -= 9;
-
- /* the first chapter will be used as the lead_trim */
- if (chapters > 0) {
- read_uint64be(fd, &timestamp);
- id3->lead_trim = (timestamp * id3->frequency) / 10000000;
- size -= 8;
- }
- }
- break;
-
- default:
- break;
- }
-
- /* Skip final seek. */
- if (!done)
- {
- lseek(fd, size, SEEK_CUR);
- }
- } while (rc && (size_left > 0) && (errno == 0) && !done);
-
- return rc;
-}
-
-bool get_mp4_metadata(int fd, struct mp3entry* id3)
-{
- id3->codectype = AFMT_UNKNOWN;
- id3->filesize = 0;
- errno = 0;
-
- if (read_mp4_container(fd, id3, filesize(fd)) && (errno == 0)
- && (id3->samples > 0) && (id3->frequency > 0)
- && (id3->filesize > 0))
- {
- if (id3->codectype == AFMT_UNKNOWN)
- {
- logf("Not an ALAC or AAC file");
- return false;
- }
-
- id3->length = ((int64_t) id3->samples * 1000) / id3->frequency;
-
- id3->vbr = true; /* ALAC is native VBR, AAC very unlikely is CBR. */
-
- if (id3->length <= 0)
- {
- logf("mp4 length invalid!");
- return false;
- }
-
- id3->bitrate = ((int64_t) id3->filesize * 8) / id3->length;
- DEBUGF("MP4 bitrate %d, frequency %ld Hz, length %ld ms\n",
- id3->bitrate, id3->frequency, id3->length);
- }
- else
- {
- logf("MP4 metadata error");
- DEBUGF("MP4 metadata error. errno %d, samples %ld, frequency %ld, "
- "filesize %ld\n", errno, id3->samples, id3->frequency,
- id3->filesize);
- return false;
- }
-
- return true;
-}
diff --git a/apps/metadata/mpc.c b/apps/metadata/mpc.c
deleted file mode 100644
index 0b75ed04dd..0000000000
--- a/apps/metadata/mpc.c
+++ /dev/null
@@ -1,220 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Thom Johansen
- * Copyright (C) 2010 Andree Buschmann
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <string.h>
-#include <stdio.h>
-#include <inttypes.h>
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-#include "replaygain.h"
-#include "fixedpoint.h"
-
-/* Needed for replay gain and clipping prevention of SV8 files. */
-#define SV8_TO_SV7_CONVERT_GAIN (6482) /* 64.82 * 100, MPC_OLD_GAIN_REF */
-#define SV8_TO_SV7_CONVERT_PEAK (23119) /* 256 * 20 * log10(32768) */
-
-static int set_replaygain_sv7(struct mp3entry* id3,
- bool album,
- long value,
- long used)
-{
- long gain = (int16_t) ((value >> 16) & 0xffff);
- long peak = (uint16_t) (value & 0xffff);
-
- /* We use a peak value of 0 to indicate a given gain type isn't used. */
- if (peak != 0) {
- /* Save the ReplayGain data to id3-structure for further processing. */
- parse_replaygain_int(album, gain * 512 / 100, peak << 9, id3);
- }
-
- return used;
-}
-
-static int set_replaygain_sv8(struct mp3entry* id3,
- bool album,
- long gain,
- long peak,
- long used)
-{
- gain = (long)(SV8_TO_SV7_CONVERT_GAIN - ((gain*100)/256));
-
- /* Transform SV8's logarithmic peak representation to the desired linear
- * representation: linear = pow(10, peak/256/20).
- *
- * FP_BITS = 24 bits = desired fp representation for dsp routines
- * FRAC_BITS = 12 bits = resolution used for fp_bits
- * fp_factor(peak*(1<<FRAC_BITS)/256, FRAC_BITS) << (FP_BITS-FRAC_BITS)
- **/
- peak = (fp_factor((peak-SV8_TO_SV7_CONVERT_PEAK)*16, 12) << 12);
-
- /* We use a peak value of 0 to indicate a given gain type isn't used. */
- if (peak != 0) {
- /* Save the ReplayGain data to id3-structure for further processing. */
- parse_replaygain_int(album, gain * 512 / 100, peak, id3);
- }
-
- return used;
-}
-
-static int sv8_get_size(uint8_t *buffer, int index, uint64_t *p_size)
-{
- unsigned char tmp;
- uint64_t size = 0;
-
- do {
- tmp = buffer[index++];
- size = (size << 7) | (tmp & 0x7F);
- } while((tmp & 0x80));
-
- *p_size = size;
- return index;
-}
-
-bool get_musepack_metadata(int fd, struct mp3entry *id3)
-{
- static const int32_t sfreqs[4] = { 44100, 48000, 37800, 32000 };
- uint32_t header[8];
- uint64_t samples = 0;
- int i;
-
- if (!skip_id3v2(fd, id3))
- return false;
- if (read(fd, header, 4*8) != 4*8) return false;
- /* Musepack files are little endian, might need swapping */
- for (i = 1; i < 8; i++)
- header[i] = letoh32(header[i]);
- if (!memcmp(header, "MP+", 3)) { /* Compare to sig "MP+" */
- unsigned int streamversion;
- header[0] = letoh32(header[0]);
- streamversion = (header[0] >> 24) & 15;
- if (streamversion == 7) {
- unsigned int gapless = (header[5] >> 31) & 0x0001;
- unsigned int last_frame_samples = (header[5] >> 20) & 0x07ff;
- unsigned int bufused = 0;
-
- id3->frequency = sfreqs[(header[2] >> 16) & 0x0003];
- samples = (uint64_t)header[1]*1152; /* 1152 is mpc frame size */
- if (gapless)
- samples -= 1152 - last_frame_samples;
- else
- samples -= 481; /* Musepack subband synth filter delay */
-
- bufused = set_replaygain_sv7(id3, false, header[3], bufused);
- bufused = set_replaygain_sv7(id3, true , header[4], bufused);
-
- id3->codectype = AFMT_MPC_SV7;
- } else {
- return false; /* only SV7 is allowed within a "MP+" signature */
- }
- } else if (!memcmp(header, "MPCK", 4)) { /* Compare to sig "MPCK" */
- uint8_t sv8_header[32];
- /* 4 bytes 'MPCK' */
- lseek(fd, 4, SEEK_SET);
- if (read(fd, sv8_header, 2) != 2) return false; /* read frame ID */
- if (!memcmp(sv8_header, "SH", 2)) { /* Stream Header ID */
- int32_t k = 0;
- uint32_t streamversion;
- uint64_t size = 0; /* tag size */
- uint64_t dummy = 0; /* used to dummy read data from header */
-
- /* 4 bytes 'MPCK' + 2 'SH' */
- lseek(fd, 6, SEEK_SET);
- if (read(fd, sv8_header, 32) != 32) return false;
-
- /* Read the size of 'SH'-tag */
- k = sv8_get_size(sv8_header, k, &size);
-
- /* Skip crc32 */
- k += 4;
-
- /* Read stream version */
- streamversion = sv8_header[k++];
- if (streamversion != 8) return false; /* Only SV8 is allowed. */
-
- /* Number of samples */
- k = sv8_get_size(sv8_header, k, &samples);
-
- /* Number of leading zero-samples */
- k = sv8_get_size(sv8_header, k, &dummy);
-
- /* Sampling frequency */
- id3->frequency = sfreqs[(sv8_header[k++] >> 5) & 0x0003];
-
- /* Number of channels */
- id3->channels = (sv8_header[k++] >> 4) + 1;
-
- /* Skip to next tag: k = size -2 */
- k = size - 2;
-
- if (!memcmp(sv8_header+k, "RG", 2)) { /* Replay Gain ID */
- long peak, gain;
- int bufused = 0;
-
- k += 2; /* 2 bytes 'RG' */
-
- /* sv8_get_size must be called to skip the right amount of
- * bits within the header data. */
- k = sv8_get_size(sv8_header, k, &size);
-
- /* Read and set replay gain */
- if (sv8_header[k++] == 1) {
- /* Title's peak and gain */
- gain = (int16_t) ((sv8_header[k]<<8) + sv8_header[k+1]); k += 2;
- peak = (uint16_t)((sv8_header[k]<<8) + sv8_header[k+1]); k += 2;
- bufused += set_replaygain_sv8(id3, false, gain, peak, bufused);
-
- /* Album's peak and gain */
- gain = (int16_t) ((sv8_header[k]<<8) + sv8_header[k+1]); k += 2;
- peak = (uint16_t)((sv8_header[k]<<8) + sv8_header[k+1]); k += 2;
- bufused += set_replaygain_sv8(id3, true , gain, peak, bufused);
- }
- }
-
- id3->codectype = AFMT_MPC_SV8;
- } else {
- /* No sv8 stream header found */
- return false;
- }
- } else {
- return false; /* SV4-6 is not supported anymore */
- }
-
- id3->vbr = true;
- /* Estimate bitrate, we should probably subtract the various header sizes
- here for super-accurate results */
- id3->length = ((int64_t) samples * 1000) / id3->frequency;
-
- if (id3->length <= 0)
- {
- logf("mpc length invalid!");
- return false;
- }
-
- id3->filesize = filesize(fd);
- id3->bitrate = id3->filesize * 8 / id3->length;
-
- read_ape_tags(fd, id3);
- return true;
-}
diff --git a/apps/metadata/nsf.c b/apps/metadata/nsf.c
deleted file mode 100644
index 2fa6f36b12..0000000000
--- a/apps/metadata/nsf.c
+++ /dev/null
@@ -1,278 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-#include "string-extra.h"
-
-/* NOTE: This file was modified to work properly with the new nsf codec based
- on Game_Music_Emu */
-
-struct NESM_HEADER
-{
- uint32_t nHeader;
- uint8_t nHeaderExtra;
- uint8_t nVersion;
- uint8_t nTrackCount;
- uint8_t nInitialTrack;
- uint16_t nLoadAddress;
- uint16_t nInitAddress;
- uint16_t nPlayAddress;
- uint8_t szGameTitle[32];
- uint8_t szArtist[32];
- uint8_t szCopyright[32];
- uint16_t nSpeedNTSC;
- uint8_t nBankSwitch[8];
- uint16_t nSpeedPAL;
- uint8_t nNTSC_PAL;
- uint8_t nExtraChip;
- uint8_t nExpansion[4];
-} __attribute__((packed));
-
-struct NSFE_INFOCHUNK
-{
- uint16_t nLoadAddress;
- uint16_t nInitAddress;
- uint16_t nPlayAddress;
- uint8_t nIsPal;
- uint8_t nExt;
- uint8_t nTrackCount;
- uint8_t nStartingTrack;
-} __attribute__((packed));
-
-
-#define CHAR4_CONST(a, b, c, d) FOURCC(a, b, c, d)
-#define CHUNK_INFO 0x0001
-#define CHUNK_DATA 0x0002
-#define CHUNK_NEND 0x0004
-#define CHUNK_plst 0x0008
-#define CHUNK_time 0x0010
-#define CHUNK_fade 0x0020
-#define CHUNK_tlbl 0x0040
-#define CHUNK_auth 0x0080
-#define CHUNK_BANK 0x0100
-
-static bool parse_nsfe(int fd, struct mp3entry *id3)
-{
- unsigned int chunks_found = 0;
- long track_count = 0;
- long playlist_count = 0;
-
- struct NSFE_INFOCHUNK info;
- memset(&info, 0, sizeof(struct NSFE_INFOCHUNK));
-
- /* default values */
- info.nTrackCount = 1;
- id3->length = 150 * 1000;
-
- /* begin reading chunks */
- while (!(chunks_found & CHUNK_NEND))
- {
- uint32_t chunk_size, chunk_type;
-
- if (read_uint32le(fd, &chunk_size) != (int)sizeof(uint32_t))
- return false;
-
- if (read_uint32be(fd, &chunk_type) != (int)sizeof(uint32_t))
- return false;
-
- switch (chunk_type)
- {
- /* first three types are mandatory (but don't worry about NEND
- anyway) */
- case CHAR4_CONST('I', 'N', 'F', 'O'):
- {
- if (chunks_found & CHUNK_INFO)
- return false; /* only one info chunk permitted */
-
- chunks_found |= CHUNK_INFO;
-
- /* minimum size */
- if (chunk_size < 8)
- return false;
-
- ssize_t size = MIN(sizeof(struct NSFE_INFOCHUNK), chunk_size);
-
- if (read(fd, &info, size) != size)
- return false;
-
- if (size >= 9)
- track_count = info.nTrackCount;
-
- chunk_size -= size;
- break;
- }
-
- case CHAR4_CONST('D', 'A', 'T', 'A'):
- {
- if (!(chunks_found & CHUNK_INFO))
- return false;
-
- if (chunks_found & CHUNK_DATA)
- return false; /* only one may exist */
-
- if (chunk_size < 1)
- return false;
-
- chunks_found |= CHUNK_DATA;
- break;
- }
-
- case CHAR4_CONST('N', 'E', 'N', 'D'):
- {
- /* just end parsing regardless of whether or not this really is the
- last chunk/data (but it _should_ be) */
- chunks_found |= CHUNK_NEND;
- continue;
- }
-
- /* remaining types are optional */
-
- case CHAR4_CONST('a', 'u', 't', 'h'):
- {
- if (chunks_found & CHUNK_auth)
- return false; /* only one may exist */
-
- chunks_found |= CHUNK_auth;
-
- /* szGameTitle, szArtist, szCopyright */
- char ** const ar[] = { &id3->title, &id3->artist, &id3->album };
-
- char *p = id3->id3v2buf;
- long buf_rem = sizeof (id3->id3v2buf);
- unsigned int i;
-
- for (i = 0; i < ARRAYLEN(ar) && chunk_size && buf_rem; i++)
- {
- long len = read_string(fd, p, buf_rem, '\0', chunk_size);
-
- if (len < 0)
- return false;
-
- *ar[i] = p;
- p += len;
- buf_rem -= len;
-
- if (chunk_size >= (uint32_t)len)
- chunk_size -= len;
- else
- chunk_size = 0;
- }
-
- break;
- }
-
- case CHAR4_CONST('p', 'l', 's', 't'):
- {
- if (chunks_found & CHUNK_plst)
- return false; /* only one may exist */
-
- chunks_found |= CHUNK_plst;
-
- /* each byte is the index of one track */
- playlist_count = chunk_size;
- break;
- }
-
- case CHAR4_CONST('t', 'i', 'm', 'e'):
- case CHAR4_CONST('f', 'a', 'd', 'e'):
- case CHAR4_CONST('t', 'l', 'b', 'l'): /* we unfortunately can't use these anyway */
- {
- /* don't care how many of these there are even though there should
- be only one */
- if (!(chunks_found & CHUNK_INFO))
- return false;
-
- case CHAR4_CONST('B', 'A', 'N', 'K'):
- break;
- }
-
- default: /* unknown chunk */
- {
- /* check the first byte */
- chunk_type = (uint8_t)chunk_type;
-
- /* chunk is vital... don't continue */
- if(chunk_type >= 'A' && chunk_type <= 'Z')
- return false;
-
- /* otherwise, just skip it */
- break;
- }
- } /* end switch */
-
- lseek(fd, chunk_size, SEEK_CUR);
- } /* end while */
-
- if (track_count | playlist_count)
- id3->length = MAX(track_count, playlist_count)*1000;
-
- /* Single subtrack files will be treated differently
- by gme's nsf codec */
- if (id3->length <= 1000) id3->length = 150 * 1000;
-
- /*
- * if we exited the while loop without a 'return', we must have hit an NEND
- * chunk if this is the case, the file was layed out as it was expected.
- * now.. make sure we found both an info chunk, AND a data chunk... since
- * these are minimum requirements for a valid NSFE file
- */
- return (chunks_found & (CHUNK_INFO | CHUNK_DATA)) ==
- (CHUNK_INFO | CHUNK_DATA);
-}
-
-static bool parse_nesm(int fd, struct mp3entry *id3)
-{
- struct NESM_HEADER hdr;
- char *p = id3->id3v2buf;
-
- lseek(fd, 0, SEEK_SET);
- if (read(fd, &hdr, sizeof(hdr)) != sizeof(hdr))
- return false;
-
- /* Length */
- id3->length = (hdr.nTrackCount > 1 ? hdr.nTrackCount : 150) * 1000;
-
- /* Title */
- id3->title = p;
- p += strlcpy(p, hdr.szGameTitle, 32) + 1;
-
- /* Artist */
- id3->artist = p;
- p += strlcpy(p, hdr.szArtist, 32) + 1;
-
- /* Copyright (per codec) */
- id3->album = p;
- strlcpy(p, hdr.szCopyright, 32);
-
- return true;
-}
-
-bool get_nsf_metadata(int fd, struct mp3entry* id3)
-{
- uint32_t nsf_type;
- if (lseek(fd, 0, SEEK_SET) < 0 ||
- read_uint32be(fd, &nsf_type) != (int)sizeof(nsf_type))
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
- /* we only render 16 bits, 44.1KHz, Mono */
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- if (nsf_type == CHAR4_CONST('N', 'S', 'F', 'E'))
- return parse_nsfe(fd, id3);
- else if (nsf_type == CHAR4_CONST('N', 'E', 'S', 'M'))
- return parse_nesm(fd, id3);
-
- /* not a valid format*/
- return false;
-}
-
diff --git a/apps/metadata/ogg.c b/apps/metadata/ogg.c
deleted file mode 100644
index 3a3cb29998..0000000000
--- a/apps/metadata/ogg.c
+++ /dev/null
@@ -1,215 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-/* A simple parser to read vital metadata from an Ogg Vorbis file.
- * Can also handle parsing Ogg Speex files for metadata. Returns
- * false if metadata needed by the codec couldn't be read.
- */
-bool get_ogg_metadata(int fd, struct mp3entry* id3)
-{
- /* An Ogg File is split into pages, each starting with the string
- * "OggS". Each page has a timestamp (in PCM samples) referred to as
- * the "granule position".
- *
- * An Ogg Vorbis has the following structure:
- * 1) Identification header (containing samplerate, numchannels, etc)
- * 2) Comment header - containing the Vorbis Comments
- * 3) Setup header - containing codec setup information
- * 4) Many audio packets...
- *
- * An Ogg Speex has the following structure:
- * 1) Identification header (containing samplerate, numchannels, etc)
- * Described in this page: (http://www.speex.org/manual2/node7.html)
- * 2) Comment header - containing the Vorbis Comments
- * 3) Many audio packets.
- */
-
- /* Use the path name of the id3 structure as a temporary buffer. */
- unsigned char* buf = (unsigned char *)id3->path;
- long comment_size;
- long remaining = 0;
- long last_serial = 0;
- long serial, r;
- int segments, header_size;
- int i;
- bool eof = false;
-
- /* 92 bytes is enough for both Vorbis and Speex headers */
- if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 92) < 92))
- {
- return false;
- }
-
- /* All Ogg streams start with OggS */
- if (memcmp(buf, "OggS", 4) != 0)
- {
- return false;
- }
-
- /* Check for format magic and then get metadata */
- if (memcmp(&buf[29], "vorbis", 6) == 0)
- {
- id3->codectype = AFMT_OGG_VORBIS;
- id3->frequency = get_long_le(&buf[40]);
- id3->vbr = true;
-
- /* Comments are in second Ogg page (byte 58 onwards for Vorbis) */
- if (lseek(fd, 58, SEEK_SET) < 0)
- {
- return false;
- }
- }
- else if (memcmp(&buf[28], "Speex ", 8) == 0)
- {
- id3->codectype = AFMT_SPEEX;
- id3->frequency = get_slong(&buf[64]);
- id3->vbr = get_long_le(&buf[88]);
-
- header_size = get_long_le(&buf[60]);
-
- /* Comments are in second Ogg page (byte 108 onwards for Speex) */
- if (lseek(fd, 28 + header_size, SEEK_SET) < 0)
- {
- return false;
- }
- }
- else
- {
- /* Unsupported format, try to print the marker, catches Ogg/FLAC at least */
- DEBUGF("Usupported format in Ogg stream: %16s\n", &buf[28]);
- return false;
- }
-
- id3->filesize = filesize(fd);
-
- /* We need to ensure the serial number from this page is the same as the
- * one from the last page (since we only support a single bitstream).
- */
- serial = get_long_le(&buf[14]);
- comment_size = read_vorbis_tags(fd, id3, remaining);
-
- /* We now need to search for the last page in the file - identified by
- * by ('O','g','g','S',0) and retrieve totalsamples.
- */
-
- /* A page is always < 64 kB */
- if (lseek(fd, -(MIN(64 * 1024, id3->filesize)), SEEK_END) < 0)
- {
- return false;
- }
-
- remaining = 0;
-
- while (!eof)
- {
- r = read(fd, &buf[remaining], MAX_PATH - remaining);
-
- if (r <= 0)
- {
- eof = true;
- }
- else
- {
- remaining += r;
- }
-
- /* Inefficient (but simple) search */
- i = 0;
-
- while (i < (remaining - 3))
- {
- if ((buf[i] == 'O') && (memcmp(&buf[i], "OggS", 4) == 0))
- {
- if (i < (remaining - 17))
- {
- /* Note that this only reads the low 32 bits of a
- * 64 bit value.
- */
- id3->samples = get_long_le(&buf[i + 6]);
- last_serial = get_long_le(&buf[i + 14]);
-
- /* If this page is very small the beginning of the next
- * header could be in buffer. Jump near end of this header
- * and continue */
- i += 27;
- }
- else
- {
- break;
- }
- }
- else
- {
- i++;
- }
- }
-
- if (i < remaining)
- {
- /* Move the remaining bytes to start of buffer.
- * Reuse var 'segments' as it is no longer needed */
- segments = 0;
- while (i < remaining)
- {
- buf[segments++] = buf[i++];
- }
- remaining = segments;
- }
- else
- {
- /* Discard the rest of the buffer */
- remaining = 0;
- }
- }
-
- /* This file has mutiple vorbis bitstreams (or is corrupt). */
- /* FIXME we should display an error here. */
- if (serial != last_serial)
- {
- logf("serialno mismatch");
- logf("%ld", serial);
- logf("%ld", last_serial);
- return false;
- }
-
- id3->length = ((int64_t) id3->samples * 1000) / id3->frequency;
- if (id3->length <= 0)
- {
- logf("ogg length invalid!");
- return false;
- }
-
- id3->bitrate = (((int64_t) id3->filesize - comment_size) * 8) / id3->length;
-
- return true;
-}
-
diff --git a/apps/metadata/oma.c b/apps/metadata/oma.c
deleted file mode 100644
index b82c0a4f73..0000000000
--- a/apps/metadata/oma.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * Sony OpenMG (OMA) demuxer
- *
- * Copyright (c) 2008 Maxim Poliakovski
- * 2008 Benjamin Larsson
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file oma.c
- * This is a demuxer for Sony OpenMG Music files
- *
- * Known file extensions: ".oma", "aa3"
- * The format of such files consists of three parts:
- * - "ea3" header carrying overall info and metadata.
- * - "EA3" header is a Sony-specific header containing information about
- * the OpenMG file: codec type (usually ATRAC, can also be MP3 or WMA),
- * codec specific info (packet size, sample rate, channels and so on)
- * and DRM related info (file encryption, content id).
- * - Sound data organized in packets follow the EA3 header
- * (can be encrypted using the Sony DRM!).
- *
- * LIMITATIONS: This version supports only plain (unencrypted) OMA files.
- * If any DRM-protected (encrypted) file is encountered you will get the
- * corresponding error message. Try to remove the encryption using any
- * Sony software (for example SonicStage).
- * CODEC SUPPORT: Only ATRAC3 codec is currently supported!
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <inttypes.h>
-#include <string.h>
-#include "metadata.h"
-#include "metadata_parsers.h"
-
-#define EA3_HEADER_SIZE 96
-
-#if 0
-#define DEBUGF printf
-#else
-#define DEBUGF(...)
-#endif
-
-/* Various helper macros taken from ffmpeg for reading *
- * and writing buffers with a specified endianess. */
-# define AV_RB16(x) \
- ((((const uint8_t*)(x))[0] << 8) | \
- ((const uint8_t*)(x))[1])
-# define AV_RB24(x) \
- ((((const uint8_t*)(x))[0] << 16) | \
- (((const uint8_t*)(x))[1] << 8) | \
- ((const uint8_t*)(x))[2])
-# define AV_RB32(x) \
- ((((const uint8_t*)(x))[0] << 24) | \
- (((const uint8_t*)(x))[1] << 16) | \
- (((const uint8_t*)(x))[2] << 8) | \
- ((const uint8_t*)(x))[3])
-# define AV_WL32(p, d) do { \
- ((uint8_t*)(p))[0] = (d); \
- ((uint8_t*)(p))[1] = (d)>>8; \
- ((uint8_t*)(p))[2] = (d)>>16; \
- ((uint8_t*)(p))[3] = (d)>>24; \
- } while(0)
-# define AV_WL16(p, d) do { \
- ((uint8_t*)(p))[0] = (d); \
- ((uint8_t*)(p))[1] = (d)>>8; \
- } while(0)
-
-/* Different codecs that could be present in a Sony OMA *
- * container file. */
-enum {
- OMA_CODECID_ATRAC3 = 0,
- OMA_CODECID_ATRAC3P = 1,
- OMA_CODECID_MP3 = 3,
- OMA_CODECID_LPCM = 4,
- OMA_CODECID_WMA = 5,
-};
-
-/* FIXME: This functions currently read different file *
- * parameters required for decoding. It still *
- * does not read the metadata - which should be *
- * present in the ea3 (first) header. The *
- * metadata in ea3 is stored as a variation of *
- * the ID3v2 metadata format. */
-static int oma_read_header(int fd, struct mp3entry* id3)
-{
- static const uint16_t srate_tab[6] = {320,441,480,882,960,0};
- int ret, ea3_taglen, EA3_pos, jsflag;
- uint32_t codec_params;
- int16_t eid;
- uint8_t buf[EA3_HEADER_SIZE];
-
- ret = read(fd, buf, 10);
- if (ret != 10)
- return -1;
-
- ea3_taglen = ((buf[6] & 0x7f) << 21) | ((buf[7] & 0x7f) << 14) | ((buf[8] & 0x7f) << 7) | (buf[9] & 0x7f);
-
- EA3_pos = ea3_taglen + 10;
- if (buf[5] & 0x10)
- EA3_pos += 10;
-
- lseek(fd, EA3_pos, SEEK_SET);
- ret = read(fd, buf, EA3_HEADER_SIZE);
- if (ret != EA3_HEADER_SIZE)
- return -1;
-
- if (memcmp(buf, ((const uint8_t[]){'E', 'A', '3'}),3) || buf[4] != 0 || buf[5] != EA3_HEADER_SIZE) {
- DEBUGF("Couldn't find the EA3 header !\n");
- return -1;
- }
-
- eid = AV_RB16(&buf[6]);
- if (eid != -1 && eid != -128) {
- DEBUGF("Encrypted file! Eid: %d\n", eid);
- return -1;
- }
-
- codec_params = AV_RB24(&buf[33]);
-
- switch (buf[32]) {
- case OMA_CODECID_ATRAC3:
- id3->frequency = srate_tab[(codec_params >> 13) & 7]*100;
- if (id3->frequency != 44100) {
- DEBUGF("Unsupported sample rate, send sample file to developers: %d\n", id3->frequency);
- return -1;
- }
-
- id3->bytesperframe = (codec_params & 0x3FF) * 8;
- id3->codectype = AFMT_OMA_ATRAC3;
- jsflag = (codec_params >> 17) & 1; /* get stereo coding mode, 1 for joint-stereo */
-
- id3->bitrate = id3->frequency * id3->bytesperframe * 8 / (1024 * 1000);
-
- /* fake the atrac3 extradata (wav format, makes stream copy to wav work) */
- /* ATRAC3 expects and extra-data size of 14 bytes for wav format, and *
- * looks for that in the id3v2buf. */
- id3->extradata_size = 14;
- AV_WL16(&id3->id3v2buf[0], 1); // always 1
- AV_WL32(&id3->id3v2buf[2], id3->frequency); // samples rate
- AV_WL16(&id3->id3v2buf[6], jsflag); // coding mode
- AV_WL16(&id3->id3v2buf[8], jsflag); // coding mode
- AV_WL16(&id3->id3v2buf[10], 1); // always 1
- AV_WL16(&id3->id3v2buf[12], 0); // always 0
-
- id3->channels = 2;
- DEBUGF("sample_rate = %d\n", id3->frequency);
- DEBUGF("frame_size = %d\n", id3->bytesperframe);
- DEBUGF("stereo_coding_mode = %d\n", jsflag);
- break;
- default:
- DEBUGF("Unsupported codec %d!\n",buf[32]);
- return -1;
- break;
- }
-
- /* Store the the offset of the first audio frame, to be able to seek to it *
- * directly in atrac3_oma.codec. */
- id3->first_frame_offset = EA3_pos + EA3_HEADER_SIZE;
- return 0;
-}
-
-bool get_oma_metadata(int fd, struct mp3entry* id3)
-{
- if(oma_read_header(fd, id3) < 0)
- return false;
-
- /* Currently, there's no means of knowing the duration *
- * directly from the the file so we calculate it. */
- id3->filesize = filesize(fd);
- id3->length = ((id3->filesize - id3->first_frame_offset) * 8) / id3->bitrate;
- return true;
-}
diff --git a/apps/metadata/rm.c b/apps/metadata/rm.c
deleted file mode 100644
index 27f541cb25..0000000000
--- a/apps/metadata/rm.c
+++ /dev/null
@@ -1,464 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2009 Mohamed Tarek
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include <codecs/librm/rm.h>
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-/* Uncomment the following line for debugging */
-//#define DEBUG_RM
-#ifndef DEBUG_RM
-#undef DEBUGF
-#define DEBUGF(...)
-#endif
-
-#define ID3V1_OFFSET -128
-#define METADATA_FOOTER_OFFSET -140
-
-static inline void print_cook_extradata(RMContext *rmctx) {
-
- DEBUGF(" cook_version = 0x%08lx\n", rm_get_uint32be(rmctx->codec_extradata));
- DEBUGF(" samples_per_frame_per_channel = %d\n", rm_get_uint16be(&rmctx->codec_extradata[4]));
- DEBUGF(" number_of_subbands_in_freq_domain = %d\n", rm_get_uint16be(&rmctx->codec_extradata[6]));
- if(rmctx->extradata_size == 16) {
- DEBUGF(" joint_stereo_subband_start = %d\n",rm_get_uint16be(&rmctx->codec_extradata[12]));
- DEBUGF(" joint_stereo_vlc_bits = %d\n", rm_get_uint16be(&rmctx->codec_extradata[14]));
- }
-}
-
-
-struct real_object_t
-{
- uint32_t fourcc;
- uint32_t size;
- uint16_t version;
-};
-
-static int real_read_object_header(int fd, struct real_object_t* obj)
-{
- int n;
-
- if ((n = read_uint32be(fd, &obj->fourcc)) <= 0)
- return n;
- if ((n = read_uint32be(fd, &obj->size)) <= 0)
- return n;
- if ((n = read_uint16be(fd, &obj->version)) <= 0)
- return n;
-
- return 1;
-}
-
-#if (defined(SIMULATOR) && defined(DEBUG_RM))
-static char* fourcc2str(uint32_t f)
-{
- static char res[5];
-
- res[0] = (f & 0xff000000) >> 24;
- res[1] = (f & 0xff0000) >> 16;
- res[2] = (f & 0xff00) >> 8;
- res[3] = (f & 0xff);
- res[4] = 0;
-
- return res;
-}
-#endif
-
-static inline int real_read_audio_stream_info(int fd, RMContext *rmctx)
-{
- int skipped = 0;
- uint32_t version;
- struct real_object_t obj;
-#ifdef SIMULATOR
- uint32_t header_size;
- uint16_t flavor;
- uint32_t coded_framesize;
- uint8_t interleaver_id_length;
- uint8_t fourcc_length;
-#endif
- uint32_t interleaver_id;
- uint32_t fourcc = 0;
-
- memset(&obj,0,sizeof(obj));
- read_uint32be(fd, &version);
- skipped += 4;
-
- DEBUGF(" version=0x%04lx\n",((version >> 16) & 0xff));
- if (((version >> 16) & 0xff) == 3) {
- /* Very old version */
- } else {
-#ifdef SIMULATOR
- real_read_object_header(fd, &obj);
- read_uint32be(fd, &header_size);
- /* obj.size will be filled with an unknown value, replaced with header_size */
- DEBUGF(" Object: %s, size: %ld bytes, version: 0x%04x\n",fourcc2str(obj.fourcc),header_size,obj.version);
-
- read_uint16be(fd, &flavor);
- read_uint32be(fd, &coded_framesize);
-#else
- lseek(fd, 20, SEEK_CUR);
-#endif
- lseek(fd, 12, SEEK_CUR); /* unknown */
- read_uint16be(fd, &rmctx->sub_packet_h);
- read_uint16be(fd, &rmctx->block_align);
- read_uint16be(fd, &rmctx->sub_packet_size);
- lseek(fd, 2, SEEK_CUR); /* unknown */
- skipped += 40;
- if (((version >> 16) & 0xff) == 5)
- {
- lseek(fd, 6, SEEK_CUR); /* unknown */
- skipped += 6;
- }
- read_uint16be(fd, &rmctx->sample_rate);
- lseek(fd, 4, SEEK_CUR); /* unknown */
- read_uint16be(fd, &rmctx->nb_channels);
- skipped += 8;
- if (((version >> 16) & 0xff) == 4)
- {
-#ifdef SIMULATOR
- read_uint8(fd, &interleaver_id_length);
- read_uint32be(fd, &interleaver_id);
- read_uint8(fd, &fourcc_length);
-#else
- lseek(fd, 6, SEEK_CUR);
-#endif
- read_uint32be(fd, &fourcc);
- skipped += 10;
- }
- if (((version >> 16) & 0xff) == 5)
- {
- read_uint32be(fd, &interleaver_id);
- read_uint32be(fd, &fourcc);
- skipped += 8;
- }
- lseek(fd, 3, SEEK_CUR); /* unknown */
- skipped += 3;
- if (((version >> 16) & 0xff) == 5)
- {
- lseek(fd, 1, SEEK_CUR); /* unknown */
- skipped += 1;
- }
-
- switch(fourcc) {
- case FOURCC('c','o','o','k'):
- rmctx->codec_type = CODEC_COOK;
- read_uint32be(fd, &rmctx->extradata_size);
- skipped += 4;
- read(fd, rmctx->codec_extradata, rmctx->extradata_size);
- skipped += rmctx->extradata_size;
- break;
-
- case FOURCC('r','a','a','c'):
- case FOURCC('r','a','c','p'):
- rmctx->codec_type = CODEC_AAC;
- read_uint32be(fd, &rmctx->extradata_size);
- skipped += 4;
- read(fd, rmctx->codec_extradata, rmctx->extradata_size);
- skipped += rmctx->extradata_size;
- break;
-
- case FOURCC('d','n','e','t'):
- rmctx->codec_type = CODEC_AC3;
- break;
-
- case FOURCC('a','t','r','c'):
- rmctx->codec_type = CODEC_ATRAC;
- read_uint32be(fd, &rmctx->extradata_size);
- skipped += 4;
- read(fd, rmctx->codec_extradata, rmctx->extradata_size);
- skipped += rmctx->extradata_size;
- break;
-
- default: /* Not a supported codec */
- return -1;
- }
-
- DEBUGF(" flavor = %d\n",flavor);
- DEBUGF(" coded_frame_size = %ld\n",coded_framesize);
- DEBUGF(" sub_packet_h = %d\n",rmctx->sub_packet_h);
- DEBUGF(" frame_size = %d\n",rmctx->block_align);
- DEBUGF(" sub_packet_size = %d\n",rmctx->sub_packet_size);
- DEBUGF(" sample_rate= %d\n",rmctx->sample_rate);
- DEBUGF(" channels= %d\n",rmctx->nb_channels);
- DEBUGF(" fourcc = %s\n",fourcc2str(fourcc));
- DEBUGF(" codec_extra_data_length = %ld\n",rmctx->extradata_size);
- DEBUGF(" codec_extradata :\n");
- if(rmctx->codec_type == CODEC_COOK) {
- DEBUGF(" cook_extradata :\n");
- print_cook_extradata(rmctx);
- }
-
- }
-
- return skipped;
-}
-
-static int rm_parse_header(int fd, RMContext *rmctx, struct mp3entry *id3)
-{
- struct real_object_t obj;
- int res;
- int skipped;
- off_t curpos __attribute__((unused));
- uint8_t len; /* Holds a string_length, which is then passed to read_string() */
-
-#ifdef SIMULATOR
- uint32_t avg_bitrate = 0;
- uint32_t max_packet_size;
- uint32_t avg_packet_size;
- uint32_t packet_count;
- uint32_t duration;
- uint32_t preroll;
- uint32_t index_offset;
- uint16_t stream_id;
- uint32_t start_time;
- uint32_t codec_data_size;
-#endif
- uint32_t v;
- uint32_t max_bitrate;
- uint16_t num_streams;
- uint32_t next_data_off;
- uint8_t header_end;
-
- memset(&obj,0,sizeof(obj));
- curpos = lseek(fd, 0, SEEK_SET);
- res = real_read_object_header(fd, &obj);
-
- if (obj.fourcc == FOURCC('.','r','a',0xfd))
- {
- /* Very old .ra format - not yet supported */
- return -1;
- }
- else if (obj.fourcc != FOURCC('.','R','M','F'))
- {
- return -1;
- }
-
- lseek(fd, 8, SEEK_CUR); /* unknown */
-
- DEBUGF("Object: %s, size: %d bytes, version: 0x%04x, pos: %d\n",fourcc2str(obj.fourcc),(int)obj.size,obj.version,(int)curpos);
-
- res = real_read_object_header(fd, &obj);
- header_end = 0;
- while(res)
- {
- DEBUGF("Object: %s, size: %d bytes, version: 0x%04x, pos: %d\n",fourcc2str(obj.fourcc),(int)obj.size,obj.version,(int)curpos);
- skipped = 10;
- if(obj.fourcc == FOURCC('I','N','D','X'))
- break;
- switch (obj.fourcc)
- {
- case FOURCC('P','R','O','P'): /* File properties */
- read_uint32be(fd, &max_bitrate);
- read_uint32be(fd, &rmctx->bit_rate); /*avg bitrate*/
-#ifdef SIMULATOR
- read_uint32be(fd, &max_packet_size);
- read_uint32be(fd, &avg_packet_size);
- read_uint32be(fd, &packet_count);
-#else
- lseek(fd, 3*sizeof(uint32_t), SEEK_CUR);
-#endif
- read_uint32be(fd, &rmctx->duration);
-#ifdef SIMULATOR
- read_uint32be(fd, &preroll);
- read_uint32be(fd, &index_offset);
-#else
- lseek(fd, 2*sizeof(uint32_t), SEEK_CUR);
-#endif
- read_uint32be(fd, &rmctx->data_offset);
- read_uint16be(fd, &num_streams);
- read_uint16be(fd, &rmctx->flags);
- skipped += 40;
-
- DEBUGF(" max_bitrate = %ld\n",max_bitrate);
- DEBUGF(" avg_bitrate = %ld\n",rmctx->bit_rate);
- DEBUGF(" max_packet_size = %ld\n",max_packet_size);
- DEBUGF(" avg_packet_size = %ld\n",avg_packet_size);
- DEBUGF(" packet_count = %ld\n",packet_count);
- DEBUGF(" duration = %ld\n",rmctx->duration);
- DEBUGF(" preroll = %ld\n",preroll);
- DEBUGF(" index_offset = %ld\n",index_offset);
- DEBUGF(" data_offset = %ld\n",rmctx->data_offset);
- DEBUGF(" num_streams = %d\n",num_streams);
- DEBUGF(" flags=0x%04x\n",rmctx->flags);
- break;
-
- case FOURCC('C','O','N','T'):
- /* Four strings - Title, Author, Copyright, Comment */
- read_uint8(fd,&len);
- skipped += (int)read_string(fd, id3->id3v1buf[0], sizeof(id3->id3v1buf[0]), '\0', len);
- read_uint8(fd,&len);
- skipped += (int)read_string(fd, id3->id3v1buf[1], sizeof(id3->id3v1buf[1]), '\0', len);
- read_uint8(fd,&len);
- skipped += (int)read_string(fd, id3->id3v1buf[2], sizeof(id3->id3v1buf[2]), '\0', len);
- read_uint8(fd,&len);
- skipped += (int)read_string(fd, id3->id3v1buf[3], sizeof(id3->id3v1buf[3]), '\0', len);
- skipped += 4;
-
- DEBUGF(" title=\"%s\"\n",id3->id3v1buf[0]);
- DEBUGF(" author=\"%s\"\n",id3->id3v1buf[1]);
- DEBUGF(" copyright=\"%s\"\n",id3->id3v1buf[2]);
- DEBUGF(" comment=\"%s\"\n",id3->id3v1buf[3]);
- break;
-
- case FOURCC('M','D','P','R'): /* Media properties */
-#ifdef SIMULATOR
- read_uint16be(fd,&stream_id);
- read_uint32be(fd,&max_bitrate);
- read_uint32be(fd,&avg_bitrate);
- read_uint32be(fd,&max_packet_size);
- read_uint32be(fd,&avg_packet_size);
- read_uint32be(fd,&start_time);
- read_uint32be(fd,&preroll);
- read_uint32be(fd,&duration);
-#else
- lseek(fd, 30, SEEK_CUR);
-#endif
- skipped += 30;
- read_uint8(fd,&len);
- skipped += 1;
- lseek(fd, len, SEEK_CUR); /* desc */
- skipped += len;
- read_uint8(fd,&len);
- skipped += 1;
-#ifdef SIMULATOR
- lseek(fd, len, SEEK_CUR); /* mimetype */
- read_uint32be(fd,&codec_data_size);
-#else
- lseek(fd, len + 4, SEEK_CUR);
-#endif
- skipped += len + 4;
- read_uint32be(fd,&v);
- skipped += 4;
-
- DEBUGF(" stream_id = 0x%04x\n",stream_id);
- DEBUGF(" max_bitrate = %ld\n",max_bitrate);
- DEBUGF(" avg_bitrate = %ld\n",avg_bitrate);
- DEBUGF(" max_packet_size = %ld\n",max_packet_size);
- DEBUGF(" avg_packet_size = %ld\n",avg_packet_size);
- DEBUGF(" start_time = %ld\n",start_time);
- DEBUGF(" preroll = %ld\n",preroll);
- DEBUGF(" duration = %ld\n",duration);
- DEBUGF(" codec_data_size = %ld\n",codec_data_size);
- DEBUGF(" v=\"%s\"\n", fourcc2str(v));
-
- if (v == FOURCC('.','r','a',0xfd))
- {
- int temp;
- temp= real_read_audio_stream_info(fd, rmctx);
- if(temp < 0)
- return -1;
- else
- skipped += temp;
- }
- else if (v == FOURCC('L','S','D',':'))
- {
- DEBUGF("Real audio lossless is not supported.");
- return -1;
- }
- else
- {
- /* We shall not abort with -1 here. *.rm file often seem
- * to have a second media properties header that contains
- * other metadata. */
- DEBUGF("Unknown header signature :\"%s\"\n", fourcc2str(v));
- }
-
-
- break;
-
- case FOURCC('D','A','T','A'):
- read_uint32be(fd,&rmctx->nb_packets);
- skipped += 4;
- read_uint32be(fd,&next_data_off);
- skipped += 4;
-
- /***
- * nb_packets correction :
- * in some samples, number of packets may not exactly form
- * an integer number of scrambling units. This is corrected
- * by constructing a partially filled unit out of the few
- * remaining samples at the end of decoding.
- ***/
- if(rmctx->nb_packets % rmctx->sub_packet_h)
- rmctx->nb_packets += rmctx->sub_packet_h - (rmctx->nb_packets % rmctx->sub_packet_h);
-
- DEBUGF(" data_nb_packets = %ld\n",rmctx->nb_packets);
- DEBUGF(" next DATA offset = %ld\n",next_data_off);
- header_end = 1;
- break;
- }
- if(header_end) break;
- curpos = lseek(fd, obj.size - skipped, SEEK_CUR);
- res = real_read_object_header(fd, &obj);
- }
-
-
- return 0;
-}
-
-
-bool get_rm_metadata(int fd, struct mp3entry* id3)
-{
- RMContext *rmctx = (RMContext*) (( (intptr_t)id3->id3v2buf + 3 ) &~ 3);
- memset(rmctx,0,sizeof(RMContext));
- if(rm_parse_header(fd, rmctx, id3) < 0)
- return false;
-
- if (!setid3v1title(fd, id3)) {
- /* file has no id3v1 tags, use the tags from CONT chunk */
- id3->title = id3->id3v1buf[0];
- id3->artist = id3->id3v1buf[1];
- id3->comment= id3->id3v1buf[3];
- }
-
- switch(rmctx->codec_type)
- {
- case CODEC_COOK:
- /* Already set, do nothing */
- break;
- case CODEC_AAC:
- id3->codectype = AFMT_RM_AAC;
- break;
-
- case CODEC_AC3:
- id3->codectype = AFMT_RM_AC3;
- break;
-
- case CODEC_ATRAC:
- id3->codectype = AFMT_RM_ATRAC3;
- break;
- }
-
- id3->channels = rmctx->nb_channels;
- id3->extradata_size = rmctx->extradata_size;
- id3->bitrate = rmctx->bit_rate / 1000;
- id3->frequency = rmctx->sample_rate;
- id3->length = rmctx->duration;
- id3->filesize = filesize(fd);
- return true;
-}
diff --git a/apps/metadata/sgc.c b/apps/metadata/sgc.c
deleted file mode 100644
index 78cacb9b1b..0000000000
--- a/apps/metadata/sgc.c
+++ /dev/null
@@ -1,67 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-static bool parse_sgc_header(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
-
- lseek(fd, 0, SEEK_SET);
- if (read(fd, buf, 0xA0) < 0xA0)
- return false;
-
- /* calculate track length with number of tracks */
- id3->length = buf[37] * 1000;
-
- /* If meta info was found in the m3u skip next step */
- if (id3->title && id3->title[0]) return true;
-
- char *p = id3->id3v2buf;
-
- /* Some metadata entries have 32 bytes length */
- /* Game */
- memcpy(p, &buf[64], 32); *(p + 33) = '\0';
- id3->title = p;
- p += strlen(p)+1;
-
- /* Artist */
- memcpy(p, &buf[96], 32); *(p + 33) = '\0';
- id3->artist = p;
- p += strlen(p)+1;
-
- /* Copyright */
- memcpy(p, &buf[128], 32); *(p + 33) = '\0';
- id3->album = p;
- p += strlen(p)+1;
- return true;
-}
-
-
-bool get_sgc_metadata(int fd, struct mp3entry* id3)
-{
- uint32_t sgc_type;
- if ((lseek(fd, 0, SEEK_SET) < 0) ||
- read_uint32be(fd, &sgc_type) != (int)sizeof(sgc_type))
- return false;
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
- /* we only render 16 bits, 44.1KHz, Stereo */
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* Make sure this is an SGC file */
- if (sgc_type != FOURCC('S','G','C',0x1A))
- return false;
-
- return parse_sgc_header(fd, id3);
-}
diff --git a/apps/metadata/sid.c b/apps/metadata/sid.c
deleted file mode 100644
index 50b879b56d..0000000000
--- a/apps/metadata/sid.c
+++ /dev/null
@@ -1,89 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-/* PSID metadata info is available here:
- http://www.unusedino.de/ec64/technical/formats/sidplay.html */
-bool get_sid_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- char *p;
-
-
- if ((lseek(fd, 0, SEEK_SET) < 0)
- || (read(fd, buf, 0x80) < 0x80))
- {
- return false;
- }
-
- if ((memcmp(buf, "PSID", 4) != 0))
- {
- return false;
- }
-
- p = id3->id3v2buf;
-
- /* Copy Title (assumed max 0x1f letters + 1 zero byte) */
- id3->title = p;
- buf[0x16+0x1f] = 0;
- p = iso_decode(&buf[0x16], p, 0, strlen(&buf[0x16])+1);
-
- /* Copy Artist (assumed max 0x1f letters + 1 zero byte) */
- id3->artist = p;
- buf[0x36+0x1f] = 0;
- p = iso_decode(&buf[0x36], p, 0, strlen(&buf[0x36])+1);
-
- /* Copy Year (assumed max 4 letters + 1 zero byte) */
- buf[0x56+0x4] = 0;
- id3->year = atoi(&buf[0x56]);
-
- /* Copy Album (assumed max 0x1f-0x05 letters + 1 zero byte) */
- id3->album = p;
- buf[0x56+0x1f] = 0;
- iso_decode(&buf[0x5b], p, 0, strlen(&buf[0x5b])+1);
-
- id3->bitrate = 706;
- id3->frequency = 44100;
- /* New idea as posted by Marco Alanen (ravon):
- * Set the songlength in seconds to the number of subsongs
- * so every second represents a subsong.
- * Users can then skip the current subsong by seeking
- *
- * Note: the number of songs is a 16bit value at 0xE, so this code only
- * uses the lower 8 bits of the counter.
- */
- id3->length = (buf[0xf]-1)*1000;
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- return true;
-}
diff --git a/apps/metadata/smaf.c b/apps/metadata/smaf.c
deleted file mode 100644
index 1b745d3fa1..0000000000
--- a/apps/metadata/smaf.c
+++ /dev/null
@@ -1,470 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2010 Yoshihisa Uchida
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <inttypes.h>
-#include <stdio.h>
-
-#include "string-extra.h"
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-#include "logf.h"
-
-static const int basebits[4] = { 4, 8, 12, 16 };
-
-static const int frequency[5] = { 4000, 8000, 11025, 22050, 44100 };
-
-static const int support_codepages[5] = {
-#ifdef HAVE_LCD_BITMAP
- SJIS, ISO_8859_1, -1, GB_2312, BIG_5,
-#else
- -1, ISO_8859_1, -1, -1, -1,
-#endif
-};
-
-/* extra codepage */
-#define UCS2 (NUM_CODEPAGES + 1)
-
-/* support id3 tag */
-#define TAG_TITLE (('S'<<8)|'T')
-#define TAG_ARTIST (('A'<<8)|'N')
-#define TAG_COMPOSER (('S'<<8)|'W')
-
-/* convert functions */
-#define CONVERT_SMAF_CHANNELS(c) (((c) >> 7) + 1)
-
-
-static inline int convert_smaf_audio_basebit(unsigned int basebit)
-{
- if (basebit > 3)
- return 0;
- return basebits[basebit];
-}
-
-static inline int convert_smaf_audio_frequency(unsigned int freq)
-{
- if (freq > 4)
- return 0;
- return frequency[freq];
-}
-
-static int convert_smaf_codetype(unsigned int codetype)
-{
- if (codetype < 5)
- return support_codepages[codetype];
- else if (codetype == 0x20 || codetype == 0x24) /* In Rockbox, UCS2 and UTF-16 are same. */
- return UCS2;
- else if (codetype == 0x23)
- return UTF_8;
- else if (codetype == 0xff)
- return ISO_8859_1;
- return -1;
-}
-
-static void set_length(struct mp3entry *id3, unsigned int ch, unsigned int basebit,
- unsigned int numbytes)
-{
- int bitspersample = convert_smaf_audio_basebit(basebit);
-
- if (bitspersample != 0 && id3->frequency != 0)
- {
- /* Calculate track length [ms] and bitrate [kbit/s] */
- id3->length = (uint64_t)numbytes * 8000LL
- / (bitspersample * CONVERT_SMAF_CHANNELS(ch) * id3->frequency);
- id3->bitrate = bitspersample * id3->frequency / 1000;
- }
-
- /* output contents/wave data/id3 info (for debug) */
- DEBUGF("contents info ----\n");
- DEBUGF(" TITLE: %s\n", (id3->title)? id3->title : "(NULL)");
- DEBUGF(" ARTIST: %s\n", (id3->artist)? id3->artist : "(NULL)");
- DEBUGF(" COMPOSER: %s\n", (id3->composer)? id3->composer : "(NULL)");
- DEBUGF("wave data info ----\n");
- DEBUGF(" channels: %u\n", CONVERT_SMAF_CHANNELS(ch));
- DEBUGF(" bitspersample: %d\n", bitspersample);
- DEBUGF(" numbytes; %u\n", numbytes);
- DEBUGF("id3 info ----\n");
- DEBUGF(" frquency: %u\n", (unsigned int)id3->frequency);
- DEBUGF(" bitrate: %d\n", id3->bitrate);
- DEBUGF(" length: %u\n", (unsigned int)id3->length);
-}
-
-/* contents parse functions */
-
-/* Note:
- * 1) When the codepage is UTF-8 or UCS2, contents data do not start BOM.
- * 2) The byte order of contents data is big endian.
- */
-
-static void decode2utf8(const unsigned char *src, unsigned char **dst,
- int srcsize, int *dstsize, int codepage)
-{
- unsigned char tmpbuf[srcsize * 3 + 1];
- unsigned char *p;
- int utf8size;
-
- if (codepage < NUM_CODEPAGES)
- p = iso_decode(src, tmpbuf, codepage, srcsize);
- else /* codepage == UCS2 */
- p = utf16BEdecode(src, tmpbuf, srcsize);
-
- *p = '\0';
-
- strlcpy(*dst, tmpbuf, *dstsize);
- utf8size = (p - tmpbuf) + 1;
- if (utf8size > *dstsize)
- {
- DEBUGF("metadata warning: data length: %d > contents store buffer size: %d\n",
- utf8size, *dstsize);
- utf8size = *dstsize;
- }
- *dst += utf8size;
- *dstsize -= utf8size;
-}
-
-static int read_audio_track_contets(int fd, int codepage, unsigned char **dst,
- int *dstsize)
-{
- /* value length <= 256 bytes */
- unsigned char buf[256];
- unsigned char *p = buf;
- unsigned char *q = buf;
- int datasize;
-
- read(fd, buf, 256);
-
- while (p - buf < 256 && *p != ',')
- {
- /* skip yen mark */
- if (codepage != UCS2)
- {
- if (*p == '\\')
- p++;
- }
- else if (*p == '\0' && *(p+1) == '\\')
- p += 2;
-
- if (*p > 0x7f)
- {
- if (codepage == UTF_8)
- {
- while ((*p & MASK) != COMP)
- *q++ = *p++;
- }
-#ifdef HAVE_LCD_BITMAP
- else if (codepage == SJIS)
- {
- if (*p <= 0xa0 || *p >= 0xe0)
- *q++ = *p++;
- }
-#endif
- }
-
- *q++ = *p++;
- if (codepage == UCS2)
- *q++ = *p++;
- }
- datasize = p - buf + 1;
- lseek(fd, datasize - 256, SEEK_CUR);
-
- if (dst != NULL)
- decode2utf8(buf, dst, q - buf, dstsize, codepage);
-
- return datasize;
-}
-
-static void read_score_track_contets(int fd, int codepage, int datasize,
- unsigned char **dst, int *dstsize)
-{
- unsigned char buf[datasize];
-
- read(fd, buf, datasize);
- decode2utf8(buf, dst, datasize, dstsize, codepage);
-}
-
-/* traverse chunk functions */
-
-static unsigned int search_chunk(int fd, const unsigned char *name, int nlen)
-{
- unsigned char buf[8];
- unsigned int chunksize;
-
- while (read(fd, buf, 8) > 0)
- {
- chunksize = get_long_be(buf + 4);
- if (memcmp(buf, name, nlen) == 0)
- return chunksize;
-
- lseek(fd, chunksize, SEEK_CUR);
- }
- DEBUGF("metadata error: missing '%s' chunk\n", name);
- return 0;
-}
-
-static bool parse_smaf_audio_track(int fd, struct mp3entry *id3, unsigned int datasize)
-{
- /* temporary buffer */
- unsigned char *tmp = (unsigned char*)id3->path;
- /* contents stored buffer */
- unsigned char *buf = id3->id3v2buf;
- int bufsize = sizeof(id3->id3v2buf);
-
- unsigned int chunksize = datasize;
- int valsize;
-
- int codepage;
-
- /* parse contents info */
- read(fd, tmp, 5);
- codepage = convert_smaf_codetype(tmp[2]);
- if (codepage < 0)
- {
- DEBUGF("metadata error: smaf unsupport codetype: %d\n", tmp[2]);
- return false;
- }
-
- datasize -= 5;
- while ((id3->title == NULL || id3->artist == NULL || id3->composer == NULL)
- && (datasize > 0 && bufsize > 0))
- {
- if (read(fd, tmp, 3) <= 0)
- return false;
-
- if (tmp[2] != ':')
- {
- DEBUGF("metadata error: illegal tag: %c%c%c\n", tmp[0], tmp[1], tmp[2]);
- return false;
- }
- switch ((tmp[0]<<8)|tmp[1])
- {
- case TAG_TITLE:
- id3->title = buf;
- valsize = read_audio_track_contets(fd, codepage, &buf, &bufsize);
- break;
- case TAG_ARTIST:
- id3->artist = buf;
- valsize = read_audio_track_contets(fd, codepage, &buf, &bufsize);
- break;
- case TAG_COMPOSER:
- id3->composer = buf;
- valsize = read_audio_track_contets(fd, codepage, &buf, &bufsize);
- break;
- default:
- valsize = read_audio_track_contets(fd, codepage, NULL, &bufsize);
- break;
- }
- datasize -= (valsize + 3);
- }
-
- /* search PCM Audio Track Chunk */
- lseek(fd, 16 + chunksize, SEEK_SET);
-
- chunksize = search_chunk(fd, "ATR", 3);
- if (chunksize == 0)
- {
- DEBUGF("metadata error: missing PCM Audio Track Chunk\n");
- return false;
- }
-
- /*
- * get format
- * tmp
- * +0: Format Type
- * +1: Sequence Type
- * +2: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: frequency
- * +3: bit 4-7: base bit
- * +4: TimeBase_D
- * +5: TimeBase_G
- *
- * Note: If PCM Audio Track does not include Sequence Data Chunk,
- * tmp+6 is the start position of Wave Data Chunk.
- */
- read(fd, tmp, 6);
-
- /* search Wave Data Chunk */
- chunksize = search_chunk(fd, "Awa", 3);
- if (chunksize == 0)
- {
- DEBUGF("metadata error: missing Wave Data Chunk\n");
- return false;
- }
-
- /* set track length and bitrate */
- id3->frequency = convert_smaf_audio_frequency(tmp[2] & 0x0f);
- set_length(id3, tmp[2], tmp[3] >> 4, chunksize);
- return true;
-}
-
-static bool parse_smaf_score_track(int fd, struct mp3entry *id3)
-{
- /* temporary buffer */
- unsigned char *tmp = (unsigned char*)id3->path;
- unsigned char *p = tmp;
- /* contents stored buffer */
- unsigned char *buf = id3->id3v2buf;
- int bufsize = sizeof(id3->id3v2buf);
-
- unsigned int chunksize;
- unsigned int datasize;
- int valsize;
-
- int codepage;
-
- /* parse Optional Data Chunk */
- read(fd, tmp, 21);
- if (memcmp(tmp + 5, "OPDA", 4) != 0)
- {
- DEBUGF("metadata error: missing Optional Data Chunk\n");
- return false;
- }
-
- /* Optional Data Chunk size */
- chunksize = get_long_be(tmp + 9);
-
- /* parse Data Chunk */
- if (memcmp(tmp + 13, "Dch", 3) != 0)
- {
- DEBUGF("metadata error: missing Data Chunk\n");
- return false;
- }
-
- codepage = convert_smaf_codetype(tmp[16]);
- if (codepage < 0)
- {
- DEBUGF("metadata error: smaf unsupport codetype: %d\n", tmp[16]);
- return false;
- }
-
- /* Data Chunk size */
- datasize = get_long_be(tmp + 17);
- while ((id3->title == NULL || id3->artist == NULL || id3->composer == NULL)
- && (datasize > 0 && bufsize > 0))
- {
- if (read(fd, tmp, 4) <= 0)
- return false;
-
- valsize = (tmp[2] << 8) | tmp[3];
- datasize -= (valsize + 4);
- switch ((tmp[0]<<8)|tmp[1])
- {
- case TAG_TITLE:
- id3->title = buf;
- read_score_track_contets(fd, codepage, valsize, &buf, &bufsize);
- break;
- case TAG_ARTIST:
- id3->artist = buf;
- read_score_track_contets(fd, codepage, valsize, &buf, &bufsize);
- break;
- case TAG_COMPOSER:
- id3->composer = buf;
- read_score_track_contets(fd, codepage, valsize, &buf, &bufsize);
- break;
- default:
- lseek(fd, valsize, SEEK_CUR);
- break;
- }
- }
-
- /* search Score Track Chunk */
- lseek(fd, 29 + chunksize, SEEK_SET);
-
- if (search_chunk(fd, "MTR", 3) == 0)
- {
- DEBUGF("metadata error: missing Score Track Chunk\n");
- return false;
- }
-
- /*
- * search next chunk
- * usually, next chunk ('M***') found within 40 bytes.
- */
- chunksize = 40;
- read(fd, tmp, chunksize);
-
- tmp[chunksize] = 'M'; /* stopper */
- while (*p != 'M')
- p++;
-
- chunksize -= (p - tmp);
- if (chunksize == 0)
- {
- DEBUGF("metadata error: missing Score Track Stream PCM Data Chunk");
- return false;
- }
-
- /* search Score Track Stream PCM Data Chunk */
- lseek(fd, -chunksize, SEEK_CUR);
- if (search_chunk(fd, "Mtsp", 4) == 0)
- {
- DEBUGF("metadata error: missing Score Track Stream PCM Data Chunk\n");
- return false;
- }
-
- /*
- * parse Score Track Stream Wave Data Chunk
- * tmp
- * +4-7: chunk size (WaveType(3bytes) + wave data count)
- * +8: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: base bit
- * +9: frequency (MSB)
- * +10: frequency (LSB)
- */
- read(fd, tmp, 11);
- if (memcmp(tmp, "Mwa", 3) != 0)
- {
- DEBUGF("metadata error: missing Score Track Stream Wave Data Chunk\n");
- return false;
- }
-
- /* set track length and bitrate */
- id3->frequency = (tmp[9] << 8) | tmp[10];
- set_length(id3, tmp[8], tmp[8] & 0x0f, get_long_be(tmp + 4) - 3);
- return true;
-}
-
-bool get_smaf_metadata(int fd, struct mp3entry* id3)
-{
- /* temporary buffer */
- unsigned char *tmp = (unsigned char *)id3->path;
- unsigned int chunksize;
-
- id3->title = NULL;
- id3->artist = NULL;
- id3->composer = NULL;
-
- id3->vbr = false; /* All SMAF files are CBR */
- id3->filesize = filesize(fd);
-
- /* check File Chunk and Contents Info Chunk */
- lseek(fd, 0, SEEK_SET);
- read(fd, tmp, 16);
- if ((memcmp(tmp, "MMMD", 4) != 0) || (memcmp(tmp + 8, "CNTI", 4) != 0))
- {
- DEBUGF("metadata error: does not smaf format\n");
- return false;
- }
-
- chunksize = get_long_be(tmp + 12);
- if (chunksize > 5)
- return parse_smaf_audio_track(fd, id3, chunksize);
-
- return parse_smaf_score_track(fd, id3);
-}
diff --git a/apps/metadata/spc.c b/apps/metadata/spc.c
deleted file mode 100644
index 1c0206205d..0000000000
--- a/apps/metadata/spc.c
+++ /dev/null
@@ -1,130 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "debug.h"
-#include "rbunicode.h"
-
-bool get_spc_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char * buf = (unsigned char *)id3->path;
- char * p;
-
- unsigned long length;
- unsigned long fade;
- bool isbinary = true;
- int i;
-
- /* try to get the ID666 tag */
- if ((lseek(fd, 0x2e, SEEK_SET) < 0)
- || (read(fd, buf, 0xD2) < 0xD2))
- {
- DEBUGF("lseek or read failed\n");
- return false;
- }
-
- p = id3->id3v2buf;
-
- id3->title = p;
- buf[31] = 0;
- p = iso_decode(buf, p, 0, 32);
- buf += 32;
-
- id3->album = p;
- buf[31] = 0;
- p = iso_decode(buf, p, 0, 32);
- buf += 48;
-
- id3->comment = p;
- buf[31] = 0;
- p = iso_decode(buf, p, 0, 32);
- buf += 32;
-
- /* Date check */
- if(buf[2] == '/' && buf[5] == '/')
- isbinary = false;
-
- /* Reserved bytes check */
- if(buf[0xD2 - 0x2E - 112] >= '0' &&
- buf[0xD2 - 0x2E - 112] <= '9' &&
- buf[0xD3 - 0x2E - 112] == 0x00)
- isbinary = false;
-
- /* is length & fade only digits? */
- for (i=0;i<8 && (
- (buf[0xA9 - 0x2E - 112+i]>='0'&&buf[0xA9 - 0x2E - 112+i]<='9') ||
- buf[0xA9 - 0x2E - 112+i]=='\0');
- i++);
- if (i==8) isbinary = false;
-
- if(isbinary) {
- id3->year = buf[0] | (buf[1]<<8);
- buf += 11;
-
- length = (buf[0] | (buf[1]<<8) | (buf[2]<<16)) * 1000;
- buf += 3;
-
- fade = (buf[0] | (buf[1]<<8) | (buf[2]<<16) | (buf[3]<<24));
- buf += 4;
- } else {
- char tbuf[6];
-
- buf += 6;
- buf[4] = 0;
- id3->year = atoi(buf);
- buf += 5;
-
- memcpy(tbuf, buf, 3);
- tbuf[3] = 0;
- length = atoi(tbuf) * 1000;
- buf += 3;
-
- memcpy(tbuf, buf, 5);
- tbuf[5] = 0;
- fade = atoi(tbuf);
- buf += 5;
- }
-
- id3->artist = p;
- buf[31] = 0;
- iso_decode(buf, p, 0, 32);
-
- if (length==0) {
- length=3*60*1000; /* 3 minutes */
- fade=5*1000; /* 5 seconds */
- }
-
- id3->length = length+fade;
-
- id3->filesize = filesize(fd);
- id3->genre_string = id3_get_num_genre(36);
-
- return true;
-}
diff --git a/apps/metadata/tta.c b/apps/metadata/tta.c
deleted file mode 100644
index 1d3d95f118..0000000000
--- a/apps/metadata/tta.c
+++ /dev/null
@@ -1,123 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2010 Yoshihisa Uchida
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-#define TTA1_SIGN 0x31415454
-
-#define TTA_HEADER_ID 0
-#define TTA_HEADER_AUDIO_FORMAT (TTA_HEADER_ID + sizeof(unsigned int))
-#define TTA_HEADER_NUM_CHANNELS (TTA_HEADER_AUDIO_FORMAT + sizeof(unsigned short))
-#define TTA_HEADER_BITS_PER_SAMPLE (TTA_HEADER_NUM_CHANNELS + sizeof(unsigned short))
-#define TTA_HEADER_SAMPLE_RATE (TTA_HEADER_BITS_PER_SAMPLE + sizeof(unsigned short))
-#define TTA_HEADER_DATA_LENGTH (TTA_HEADER_SAMPLE_RATE + sizeof(unsigned int))
-#define TTA_HEADER_CRC32 (TTA_HEADER_DATA_LENGTH + sizeof(unsigned int))
-#define TTA_HEADER_SIZE (TTA_HEADER_CRC32 + sizeof(unsigned int))
-
-#define TTA_HEADER_GETTER_ID(x) get_long_le(x)
-#define TTA_HEADER_GETTER_AUDIO_FORMAT(x) get_short_le(x)
-#define TTA_HEADER_GETTER_NUM_CHANNELS(x) get_short_le(x)
-#define TTA_HEADER_GETTER_BITS_PER_SAMPLE(x) get_short_le(x)
-#define TTA_HEADER_GETTER_SAMPLE_RATE(x) get_long_le(x)
-#define TTA_HEADER_GETTER_DATA_LENGTH(x) get_long_le(x)
-#define TTA_HEADER_GETTER_CRC32(x) get_long_le(x)
-
-#define GET_HEADER(x, tag) TTA_HEADER_GETTER_ ## tag((x) + TTA_HEADER_ ## tag)
-
-static void read_id3_tags(int fd, struct mp3entry* id3)
-{
- id3->title = NULL;
- id3->filesize = filesize(fd);
- id3->id3v2len = getid3v2len(fd);
- id3->tracknum = 0;
- id3->discnum = 0;
- id3->vbr = false; /* All TTA files are CBR */
-
- /* first get id3v2 tags. if no id3v2 tags ware found, get id3v1 tags */
- if (id3->id3v2len)
- {
- setid3v2title(fd, id3);
- id3->first_frame_offset = id3->id3v2len;
- return;
- }
- setid3v1title(fd, id3);
-}
-
-bool get_tta_metadata(int fd, struct mp3entry* id3)
-{
- unsigned char ttahdr[TTA_HEADER_SIZE];
- unsigned int datasize;
- unsigned int origsize;
- int bps;
-
- lseek(fd, 0, SEEK_SET);
-
- /* read id3 tags */
- read_id3_tags(fd, id3);
- lseek(fd, id3->id3v2len, SEEK_SET);
-
- /* read TTA header */
- if (read(fd, ttahdr, TTA_HEADER_SIZE) < 0)
- return false;
-
- /* check for TTA3 signature */
- if ((GET_HEADER(ttahdr, ID)) != TTA1_SIGN)
- return false;
-
- /* skip check CRC */
-
- id3->channels = (GET_HEADER(ttahdr, NUM_CHANNELS));
- id3->frequency = (GET_HEADER(ttahdr, SAMPLE_RATE));
- id3->length = ((GET_HEADER(ttahdr, DATA_LENGTH)) / id3->frequency) * 1000LL;
- bps = (GET_HEADER(ttahdr, BITS_PER_SAMPLE));
-
- datasize = id3->filesize - id3->first_frame_offset;
- origsize = (GET_HEADER(ttahdr, DATA_LENGTH)) * ((bps + 7) / 8) * id3->channels;
-
- id3->bitrate = (int) ((uint64_t) datasize * id3->frequency * id3->channels * bps
- / (origsize * 1000LL));
-
- /* output header info (for debug) */
- DEBUGF("TTA header info ----\n");
- DEBUGF("id: %x\n", (unsigned int)(GET_HEADER(ttahdr, ID)));
- DEBUGF("channels: %d\n", id3->channels);
- DEBUGF("frequency: %ld\n", id3->frequency);
- DEBUGF("length: %ld\n", id3->length);
- DEBUGF("bitrate: %d\n", id3->bitrate);
- DEBUGF("bits per sample: %d\n", bps);
- DEBUGF("compressed size: %d\n", datasize);
- DEBUGF("original size: %d\n", origsize);
- DEBUGF("id3----\n");
- DEBUGF("artist: %s\n", id3->artist);
- DEBUGF("title: %s\n", id3->title);
- DEBUGF("genre: %s\n", id3->genre_string);
-
- return true;
-}
diff --git a/apps/metadata/vgm.c b/apps/metadata/vgm.c
deleted file mode 100644
index 9ea95b3939..0000000000
--- a/apps/metadata/vgm.c
+++ /dev/null
@@ -1,195 +0,0 @@
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-
-/* Ripped off from Game_Music_Emu 0.5.2. http://www.slack.net/~ant/ */
-
-typedef unsigned char byte;
-
-enum { header_size = 0x40 };
-enum { max_field = 64 };
-
-struct header_t
-{
- char tag [4];
- byte data_size [4];
- byte version [4];
- byte psg_rate [4];
- byte ym2413_rate [4];
- byte gd3_offset [4];
- byte track_duration [4];
- byte loop_offset [4];
- byte loop_duration [4];
- byte frame_rate [4];
- byte noise_feedback [2];
- byte noise_width;
- byte unused1;
- byte ym2612_rate [4];
- byte ym2151_rate [4];
- byte data_offset [4];
- byte unused2 [8];
-};
-
-static byte const* skip_gd3_str( byte const* in, byte const* end )
-{
- while ( end - in >= 2 )
- {
- in += 2;
- if ( !(in [-2] | in [-1]) )
- break;
- }
- return in;
-}
-
-static byte const* get_gd3_str( byte const* in, byte const* end, char* field )
-{
- byte const* mid = skip_gd3_str( in, end );
- int len = (mid - in) / 2 - 1;
- if ( field && len > 0 )
- {
- len = len < (int) max_field ? len : (int) max_field;
-
- field [len] = 0;
- /* Conver to utf8 */
- utf16LEdecode( in, field, len );
-
- /* Copy string back to id3v2buf */
- strcpy( (char*) in, field );
- }
- return mid;
-}
-
-static byte const* get_gd3_pair( byte const* in, byte const* end, char* field )
-{
- return skip_gd3_str( get_gd3_str( in, end, field ), end );
-}
-
-static void parse_gd3( byte const* in, byte const* end, struct mp3entry* id3 )
-{
- char* p = id3->path;
- id3->title = (char *) in;
- in = get_gd3_pair( in, end, p ); /* Song */
-
- id3->album = (char *) in;
- in = get_gd3_pair( in, end, p ); /* Game */
-
- in = get_gd3_pair( in, end, NULL ); /* System */
-
- id3->artist = (char *) in;
- in = get_gd3_pair( in, end, p ); /* Author */
-
-#if MEMORYSIZE > 2
- in = get_gd3_str ( in, end, NULL ); /* Copyright */
- in = get_gd3_pair( in, end, NULL ); /* Dumper */
-
- id3->comment = (char *) in;
- in = get_gd3_str ( in, end, p ); /* Comment */
-#endif
-}
-
-int const gd3_header_size = 12;
-
-static long check_gd3_header( byte* h, long remain )
-{
- if ( remain < gd3_header_size ) return 0;
- if ( memcmp( h, "Gd3 ", 4 ) ) return 0;
- if ( get_long_le( h + 4 ) >= 0x200 ) return 0;
-
- long gd3_size = get_long_le( h + 8 );
- if ( gd3_size > remain - gd3_header_size )
- gd3_size = remain - gd3_header_size;
-
- return gd3_size;
-}
-
-static void get_vgm_length( struct header_t* h, struct mp3entry* id3 )
-{
- long length = get_long_le( h->track_duration ) * 10 / 441;
- if ( length > 0 )
- {
- long loop_length = 0, intro_length = 0;
- long loop = get_long_le( h->loop_duration );
- if ( loop > 0 && get_long_le( h->loop_offset ) )
- {
- loop_length = loop * 10 / 441;
- intro_length = length - loop_length;
- }
- else
- {
- intro_length = length; /* make it clear that track is no longer than length */
- loop_length = 0;
- }
-
- id3->length = intro_length + 2 * loop_length; /* intro + 2 loops */
- return;
- }
-
- id3->length = 150 * 1000; /* 2.5 minutes */
-}
-
-bool get_vgm_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the id3v2 part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->id3v2buf;
- int read_bytes;
-
- memset(buf, 0, ID3V2_BUF_SIZE);
- if ((lseek(fd, 0, SEEK_SET) < 0)
- || ((read_bytes = read(fd, buf, header_size)) < header_size))
- {
- return false;
- }
-
- id3->vbr = false;
- id3->filesize = filesize(fd);
-
- id3->bitrate = 706;
- id3->frequency = 44100;
-
- /* If file is gzipped, will get metadata later */
- if (memcmp(buf, "Vgm ", 4))
- {
- /* We must set a default song length here because
- the codec can't do it anymore */
- id3->length = 150 * 1000; /* 2.5 minutes */
- return true;
- }
-
- /* Get song length from header */
- struct header_t* header = (struct header_t*) buf;
- get_vgm_length( header, id3 );
-
- long gd3_offset = get_long_le( header->gd3_offset ) - 0x2C;
-
- /* No gd3 tag found */
- if ( gd3_offset < 0 )
- return true;
-
- /* Seek to gd3 offset and read as
- many bytes posible */
- gd3_offset = id3->filesize - (header_size + gd3_offset);
- if ((lseek(fd, -gd3_offset, SEEK_END) < 0)
- || ((read_bytes = read(fd, buf, ID3V2_BUF_SIZE)) <= 0))
- return true;
-
- byte* gd3 = buf;
- long gd3_size = check_gd3_header( gd3, read_bytes );
-
- /* GD3 tag is zero */
- if ( gd3_size == 0 )
- return true;
-
- /* Finally, parse gd3 tag */
- if ( gd3 )
- parse_gd3( gd3 + gd3_header_size, gd3 + read_bytes, id3 );
-
- return true;
-}
diff --git a/apps/metadata/vorbis.c b/apps/metadata/vorbis.c
deleted file mode 100644
index 58bd781873..0000000000
--- a/apps/metadata/vorbis.c
+++ /dev/null
@@ -1,381 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "structec.h"
-
-/* Define LOGF_ENABLE to enable logf output in this file */
-/*#define LOGF_ENABLE*/
-#include "logf.h"
-
-struct file
-{
- int fd;
- bool packet_ended;
- long packet_remaining;
-};
-
-
-/* Read an Ogg page header. file->packet_remaining is set to the size of the
- * first packet on the page; file->packet_ended is set to true if the packet
- * ended on the current page. Returns true if the page header was
- * successfully read.
- */
-static bool file_read_page_header(struct file* file)
-{
- unsigned char buffer[64];
- ssize_t table_left;
-
- /* Size of page header without segment table */
- if (read(file->fd, buffer, 27) != 27)
- {
- return false;
- }
-
- if (memcmp("OggS", buffer, 4))
- {
- return false;
- }
-
- /* Skip pattern (4), version (1), flags (1), granule position (8),
- * serial (4), pageno (4), checksum (4)
- */
- table_left = buffer[26];
- file->packet_remaining = 0;
-
- /* Read segment table for the first packet */
- do
- {
- ssize_t count = MIN(sizeof(buffer), (size_t) table_left);
- int i;
-
- if (read(file->fd, buffer, count) < count)
- {
- return false;
- }
-
- table_left -= count;
-
- for (i = 0; i < count; i++)
- {
- file->packet_remaining += buffer[i];
-
- if (buffer[i] < 255)
- {
- file->packet_ended = true;
-
- /* Skip remainder of the table */
- if (lseek(file->fd, table_left, SEEK_CUR) < 0)
- {
- return false;
- }
-
- table_left = 0;
- break;
- }
- }
- }
- while (table_left > 0);
-
- return true;
-}
-
-
-/* Read (up to) buffer_size of data from the file. If buffer is NULL, just
- * skip ahead buffer_size bytes (like lseek). Returns number of bytes read,
- * 0 if there is no more data to read (in the packet or the file), < 0 if a
- * read error occurred.
- */
-static ssize_t file_read(struct file* file, void* buffer, size_t buffer_size)
-{
- ssize_t done = 0;
- ssize_t count = -1;
-
- do
- {
- if (file->packet_remaining <= 0)
- {
- if (file->packet_ended)
- {
- break;
- }
-
- if (!file_read_page_header(file))
- {
- count = -1;
- break;
- }
- }
-
- count = MIN(buffer_size, (size_t) file->packet_remaining);
-
- if (buffer)
- {
- count = read(file->fd, buffer, count);
- }
- else
- {
- if (lseek(file->fd, count, SEEK_CUR) < 0)
- {
- count = -1;
- }
- }
-
- if (count <= 0)
- {
- break;
- }
-
- if (buffer)
- {
- buffer += count;
- }
-
- buffer_size -= count;
- done += count;
- file->packet_remaining -= count;
- }
- while (buffer_size > 0);
-
- return (count < 0 ? count : done);
-}
-
-
-/* Read an int32 from file. Returns false if a read error occurred.
- */
-static bool file_read_int32(struct file* file, int32_t* value)
-{
- char buf[sizeof(int32_t)];
-
- if (file_read(file, buf, sizeof(buf)) < (ssize_t) sizeof(buf))
- {
- return false;
- }
-
- *value = get_long_le(buf);
- return true;
-}
-
-
-/* Read a string from the file. Read up to buffer_size bytes, or, if eos
- * != -1, until the eos character is found (eos is not stored in buf,
- * unless it is nil). Writes up to buffer_size chars to buf, always
- * terminating with a nil. Returns number of chars read or < 0 if a read
- * error occurred.
- *
- * Unfortunately this is a slightly modified copy of read_string() in
- * metadata_common.c...
- */
-static long file_read_string(struct file* file, char* buffer,
- long buffer_size, int eos, long size)
-{
- long read_bytes = 0;
-
- while (size > 0)
- {
- char c;
-
- if (file_read(file, &c, 1) != 1)
- {
- read_bytes = -1;
- break;
- }
-
- read_bytes++;
- size--;
-
- if ((eos != -1) && (eos == (unsigned char) c))
- {
- break;
- }
-
- if (buffer_size > 1)
- {
- *buffer++ = c;
- buffer_size--;
- }
- else if (eos == -1)
- {
- /* No point in reading any more, skip remaining data */
- if (file_read(file, NULL, size) < 0)
- {
- read_bytes = -1;
- }
- else
- {
- read_bytes += size;
- }
-
- break;
- }
- }
-
- *buffer = 0;
- return read_bytes;
-}
-
-
-/* Init struct file for reading from fd. type is the AFMT_* codec type of
- * the file, and determines if Ogg pages are to be read. remaining is the
- * max amount to read if codec type is FLAC; it is ignored otherwise.
- * Returns true if the file was successfully initialized.
- */
-static bool file_init(struct file* file, int fd, int type, int remaining)
-{
- memset(file, 0, sizeof(*file));
- file->fd = fd;
-
- if (type == AFMT_OGG_VORBIS || type == AFMT_SPEEX)
- {
- if (!file_read_page_header(file))
- {
- return false;
- }
- }
-
- if (type == AFMT_OGG_VORBIS)
- {
- char buffer[7];
-
- /* Read packet header (type and id string) */
- if (file_read(file, buffer, sizeof(buffer)) < (ssize_t) sizeof(buffer))
- {
- return false;
- }
-
- /* The first byte of a packet is the packet type; comment packets
- * are type 3.
- */
- if (buffer[0] != 3)
- {
- return false;
- }
- }
- else if (type == AFMT_FLAC)
- {
- file->packet_remaining = remaining;
- file->packet_ended = true;
- }
-
- return true;
-}
-
-
-/* Read the items in a Vorbis comment packet. For Ogg files, the file must
- * be located on a page start, for other files, the beginning of the comment
- * data (i.e., the vendor string length). Returns total size of the
- * comments, or 0 if there was a read error.
- */
-long read_vorbis_tags(int fd, struct mp3entry *id3,
- long tag_remaining)
-{
- struct file file;
- char *buf = id3->id3v2buf;
- int32_t comment_count;
- int32_t len;
- long comment_size = 0;
- int buf_remaining = sizeof(id3->id3v2buf) + sizeof(id3->id3v1buf);
- int i;
-
- if (!file_init(&file, fd, id3->codectype, tag_remaining))
- {
- return 0;
- }
-
- /* Skip vendor string */
-
- if (!file_read_int32(&file, &len) || (file_read(&file, NULL, len) < 0))
- {
- return 0;
- }
-
- if (!file_read_int32(&file, &comment_count))
- {
- return 0;
- }
-
- comment_size += 4 + len + 4;
-
- for (i = 0; i < comment_count && file.packet_remaining > 0; i++)
- {
- char name[TAG_NAME_LENGTH];
- int32_t read_len;
-
- if (!file_read_int32(&file, &len))
- {
- return 0;
- }
-
- comment_size += 4 + len;
- read_len = file_read_string(&file, name, sizeof(name), '=', len);
-
- if (read_len < 0)
- {
- return 0;
- }
-
- len -= read_len;
- read_len = file_read_string(&file, id3->path, sizeof(id3->path), -1, len);
-
- if (read_len < 0)
- {
- return 0;
- }
-
- logf("Vorbis comment %d: %s=%s", i, name, id3->path);
-
- /* Is it an embedded cuesheet? */
- if (!strcasecmp(name, "CUESHEET"))
- {
- id3->has_embedded_cuesheet = true;
- id3->embedded_cuesheet.pos = lseek(file.fd, 0, SEEK_CUR) - read_len;
- id3->embedded_cuesheet.size = len;
- id3->embedded_cuesheet.encoding = CHAR_ENC_UTF_8;
- }
- else
- {
- len = parse_tag(name, id3->path, id3, buf, buf_remaining,
- TAGTYPE_VORBIS);
- }
-
- buf += len;
- buf_remaining -= len;
- }
-
- /* Skip to the end of the block (needed by FLAC) */
- if (file.packet_remaining)
- {
- if (file_read(&file, NULL, file.packet_remaining) < 0)
- {
- return 0;
- }
- }
-
- return comment_size;
-}
diff --git a/apps/metadata/vox.c b/apps/metadata/vox.c
deleted file mode 100644
index f6bc849a88..0000000000
--- a/apps/metadata/vox.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2010 Yoshihisa Uchida
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-bool get_vox_metadata(int fd, struct mp3entry* id3)
-{
- /*
- * vox is headerless format
- *
- * frequency: 8000 Hz
- * channels: mono
- * bitspersample: 4
- */
- id3->frequency = 8000;
- id3->bitrate = 8000 * 4 / 1000;
- id3->vbr = false; /* All VOX files are CBR */
- id3->filesize = filesize(fd);
- id3->length = id3->filesize >> 2;
-
- return true;
-}
diff --git a/apps/metadata/wave.c b/apps/metadata/wave.c
deleted file mode 100644
index 45acea1fa1..0000000000
--- a/apps/metadata/wave.c
+++ /dev/null
@@ -1,432 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Dave Chapman
- * Copyright (C) 2010 Yoshihisa Uchida
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "rbunicode.h"
-#include "logf.h"
-
-#ifdef DEBUGF
-#undef DEBUGF
-#define DEBUGF(...)
-#endif
-
-/* Wave(RIFF)/Wave64 format */
-
-
-# define AV_WL32(p, d) do { \
- ((uint8_t*)(p))[0] = (d); \
- ((uint8_t*)(p))[1] = (d)>>8; \
- ((uint8_t*)(p))[2] = (d)>>16; \
- ((uint8_t*)(p))[3] = (d)>>24; \
- } while(0)
-# define AV_WL16(p, d) do { \
- ((uint8_t*)(p))[0] = (d); \
- ((uint8_t*)(p))[1] = (d)>>8; \
- } while(0)
-
-enum {
- RIFF_CHUNK = 0,
- WAVE_CHUNK,
- FMT_CHUNK,
- FACT_CHUNK,
- DATA_CHUNK,
- LIST_CHUNK,
-};
-
-/* Wave chunk names */
-#define WAVE_CHUNKNAME_LENGTH 4
-#define WAVE_CHUNKSIZE_LENGTH 4
-
-static const unsigned char * const wave_chunklist
- = "RIFF"
- "WAVE"
- "fmt "
- "fact"
- "data"
- "LIST";
-
-/* Wave64 GUIDs */
-#define WAVE64_CHUNKNAME_LENGTH 16
-#define WAVE64_CHUNKSIZE_LENGTH 8
-
-static const unsigned char * const wave64_chunklist
- = "riff\x2e\x91\xcf\x11\xa5\xd6\x28\xdb\x04\xc1\x00\x00"
- "wave\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
- "fmt \xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
- "fact\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
- "data\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
- "\xbc\x94\x5f\x92\x5a\x52\xd2\x11\x86\xdc\x00\xc0\x4f\x8e\xdb\x8a";
-
-/* list/info chunk */
-
-struct info_chunk {
- const unsigned char* tag;
- size_t offset;
-};
-
-/* info chunk names are common wave/wave64 */
-static const struct info_chunk info_chunks[] = {
- { "INAM", offsetof(struct mp3entry, title), }, /* title */
- { "IART", offsetof(struct mp3entry, artist), }, /* artist */
- { "ISBJ", offsetof(struct mp3entry, albumartist), }, /* albumartist */
- { "IPRD", offsetof(struct mp3entry, album), }, /* album */
- { "IWRI", offsetof(struct mp3entry, composer), }, /* composer */
- { "ICMT", offsetof(struct mp3entry, comment), }, /* comment */
- { "ISRF", offsetof(struct mp3entry, grouping), }, /* grouping */
- { "IGNR", offsetof(struct mp3entry, genre_string), }, /* genre */
- { "ICRD", offsetof(struct mp3entry, year_string), }, /* date */
- { "IPRT", offsetof(struct mp3entry, track_string), }, /* track/trackcount */
- { "IFRM", offsetof(struct mp3entry, disc_string), }, /* disc/disccount */
-};
-
-#define INFO_CHUNK_COUNT ((int)ARRAYLEN(info_chunks))
-
-/* support formats */
-enum
-{
- WAVE_FORMAT_PCM = 0x0001, /* Microsoft PCM Format */
- WAVE_FORMAT_ADPCM = 0x0002, /* Microsoft ADPCM Format */
- WAVE_FORMAT_IEEE_FLOAT = 0x0003, /* IEEE Float */
- WAVE_FORMAT_ALAW = 0x0006, /* Microsoft ALAW */
- WAVE_FORMAT_MULAW = 0x0007, /* Microsoft MULAW */
- WAVE_FORMAT_DVI_ADPCM = 0x0011, /* Intel's DVI ADPCM */
- WAVE_FORMAT_DIALOGIC_OKI_ADPCM = 0x0017, /* Dialogic OKI ADPCM */
- WAVE_FORMAT_YAMAHA_ADPCM = 0x0020, /* Yamaha ADPCM */
- WAVE_FORMAT_XBOX_ADPCM = 0x0069, /* XBOX ADPCM */
- IBM_FORMAT_MULAW = 0x0101, /* same as WAVE_FORMAT_MULAW */
- IBM_FORMAT_ALAW = 0x0102, /* same as WAVE_FORMAT_ALAW */
- WAVE_FORMAT_ATRAC3 = 0x0270, /* Atrac3 stream */
- WAVE_FORMAT_SWF_ADPCM = 0x5346, /* Adobe SWF ADPCM */
- WAVE_FORMAT_EXTENSIBLE = 0xFFFE,
-};
-
-struct wave_fmt {
- unsigned int formattag;
- unsigned int channels;
- unsigned int blockalign;
- unsigned int bitspersample;
- unsigned int samplesperblock;
- uint32_t totalsamples;
- uint64_t numbytes;
-};
-
-static unsigned char *convert_utf8(const unsigned char *src, unsigned char *dst,
- int size, bool is_64)
-{
- if (is_64)
- {
- /* Note: wave64: metadata codepage is UTF-16 only */
- return utf16LEdecode(src, dst, size);
- }
- return iso_decode(src, dst, -1, size);
-}
-
-static void set_totalsamples(struct wave_fmt *fmt, struct mp3entry* id3)
-{
- switch (fmt->formattag)
- {
- case WAVE_FORMAT_PCM:
- case WAVE_FORMAT_IEEE_FLOAT:
- case WAVE_FORMAT_ALAW:
- case WAVE_FORMAT_MULAW:
- case IBM_FORMAT_ALAW:
- case IBM_FORMAT_MULAW:
- fmt->blockalign = fmt->bitspersample * fmt->channels >> 3;
- fmt->samplesperblock = 1;
- break;
- case WAVE_FORMAT_YAMAHA_ADPCM:
- if (id3->channels != 0)
- {
- fmt->samplesperblock =
- (fmt->blockalign == ((id3->frequency / 60) + 4) * fmt->channels)?
- id3->frequency / 30 : (fmt->blockalign << 1) / fmt->channels;
- }
- break;
- case WAVE_FORMAT_DIALOGIC_OKI_ADPCM:
- fmt->blockalign = 1;
- fmt->samplesperblock = 2;
- break;
- case WAVE_FORMAT_SWF_ADPCM:
- if (fmt->bitspersample != 0 && id3->channels != 0)
- {
- fmt->samplesperblock
- = (((fmt->blockalign << 3) - 2) / fmt->channels - 22)
- / fmt->bitspersample + 1;
- }
- break;
- default:
- break;
- }
-
- if (fmt->blockalign != 0)
- fmt->totalsamples = (fmt->numbytes / fmt->blockalign) * fmt->samplesperblock;
-}
-
-static void parse_riff_format(unsigned char* buf, int fmtsize, struct wave_fmt *fmt,
- struct mp3entry* id3)
-{
- /* wFormatTag */
- fmt->formattag = buf[0] | (buf[1] << 8);
- /* wChannels */
- fmt->channels = buf[2] | (buf[3] << 8);
- /* dwSamplesPerSec */
- id3->frequency = get_long_le(&buf[4]);
- /* dwAvgBytesPerSec */
- id3->bitrate = (get_long_le(&buf[8]) * 8) / 1000;
- /* wBlockAlign */
- fmt->blockalign = buf[12] | (buf[13] << 8);
- /* wBitsPerSample */
- fmt->bitspersample = buf[14] | (buf[15] << 8);
-
- if (fmt->formattag != WAVE_FORMAT_EXTENSIBLE)
- {
- if (fmtsize > 19)
- {
- /* wSamplesPerBlock */
- fmt->samplesperblock = buf[18] | (buf[19] << 8);
- }
- }
- else if (fmtsize > 25)
- {
- /* wValidBitsPerSample */
- fmt->bitspersample = buf[18] | (buf[19] << 8);
- /* SubFormat */
- fmt->formattag = buf[24] | (buf[25] << 8);
- }
-
- /* Check for ATRAC3 stream */
- if (fmt->formattag == WAVE_FORMAT_ATRAC3)
- {
- int jsflag = 0;
- if(id3->bitrate == 66 || id3->bitrate == 94)
- jsflag = 1;
-
- id3->extradata_size = 14;
- id3->channels = 2;
- id3->codectype = AFMT_OMA_ATRAC3;
- id3->bytesperframe = fmt->blockalign;
-
- /* Store the extradata for the codec */
- AV_WL16(&id3->id3v2buf[0], 1); // always 1
- AV_WL32(&id3->id3v2buf[2], id3->frequency);// samples rate
- AV_WL16(&id3->id3v2buf[6], jsflag); // coding mode
- AV_WL16(&id3->id3v2buf[8], jsflag); // coding mode
- AV_WL16(&id3->id3v2buf[10], 1); // always 1
- AV_WL16(&id3->id3v2buf[12], 0); // always 0
- }
-}
-
-static void parse_list_chunk(int fd, struct mp3entry* id3, int chunksize, bool is_64)
-{
- unsigned char tmpbuf[ID3V2_BUF_SIZE];
- unsigned char *bp = tmpbuf;
- unsigned char *endp;
- unsigned char *data_pos;
- unsigned char *tag_pos = id3->id3v2buf;
- int datasize;
- int infosize;
- int remain;
- int i;
-
- if (is_64)
- lseek(fd, 4, SEEK_CUR);
- else if (read(fd, bp, 4) < 4 || memcmp(bp, "INFO", 4))
- return;
-
- /* decrease skip bytes */
- chunksize -= 4;
-
- infosize = read(fd, bp, (ID3V2_BUF_SIZE > chunksize)? chunksize : ID3V2_BUF_SIZE);
- if (infosize <= 8)
- return;
-
- endp = bp + infosize;
- while (bp < endp)
- {
- datasize = get_long_le(bp + 4);
- data_pos = bp + 8;
- remain = ID3V2_BUF_SIZE - (tag_pos - (unsigned char*)id3->id3v2buf);
- if (remain < 1)
- break;
-
- for (i = 0; i < INFO_CHUNK_COUNT; i++)
- {
- if (memcmp(bp, info_chunks[i].tag, 4) == 0)
- {
- *((char **)(((char*)id3) + info_chunks[i].offset)) = tag_pos;
- tag_pos = convert_utf8(data_pos, tag_pos,
- (datasize + 1 >= remain )? remain - 1 : datasize,
- is_64);
- *tag_pos++ = 0;
- break;
- }
- }
- bp = data_pos + datasize + (datasize & 1);
- };
-}
-
-static bool read_header(int fd, struct mp3entry* id3, const unsigned char *chunknames,
- bool is_64)
-{
- /* Use the temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
-
- struct wave_fmt fmt;
-
- const unsigned int namelen = (is_64)? WAVE64_CHUNKNAME_LENGTH : WAVE_CHUNKNAME_LENGTH;
- const unsigned int sizelen = (is_64)? WAVE64_CHUNKSIZE_LENGTH : WAVE_CHUNKSIZE_LENGTH;
- const unsigned int len = namelen + sizelen;
- uint64_t chunksize;
- uint64_t offset = len + namelen;
- int read_data;
-
- memset(&fmt, 0, sizeof(struct wave_fmt));
-
- id3->vbr = false; /* All Wave/Wave64 files are CBR */
- id3->filesize = filesize(fd);
-
- /* get RIFF chunk header */
- lseek(fd, 0, SEEK_SET);
- read(fd, buf, offset);
-
- if ((memcmp(buf, chunknames + RIFF_CHUNK * namelen, namelen) != 0) ||
- (memcmp(buf + len, chunknames + WAVE_CHUNK * namelen, namelen) != 0))
- {
- DEBUGF("metadata error: missing riff header.\n");
- return false;
- }
-
- /* iterate over WAVE chunks until 'data' chunk */
- while (read(fd, buf, len) > 0)
- {
- offset += len;
-
- /* get chunk size (when the header is wave64, chunksize includes GUID and data length) */
- chunksize = (is_64) ? get_uint64_le(buf + namelen) - len :
- get_long_le(buf + namelen);
-
- read_data = 0;
- if (memcmp(buf, chunknames + FMT_CHUNK * namelen, namelen) == 0)
- {
- DEBUGF("find 'fmt ' chunk\n");
-
- if (chunksize < 16)
- {
- DEBUGF("metadata error: 'fmt ' chunk is too small: %d\n", (int)chunksize);
- return false;
- }
-
- /* get and parse format */
- read_data = (chunksize > 25)? 26 : chunksize;
-
- read(fd, buf, read_data);
- parse_riff_format(buf, read_data, &fmt, id3);
- }
- else if (memcmp(buf, chunknames + FACT_CHUNK * namelen, namelen) == 0)
- {
- DEBUGF("find 'fact' chunk\n");
-
- /* dwSampleLength */
- if (chunksize >= sizelen)
- {
- /* get totalsamples */
- read_data = sizelen;
- read(fd, buf, read_data);
- fmt.totalsamples = (is_64)? get_uint64_le(buf) : get_long_le(buf);
- }
- }
- else if (memcmp(buf, chunknames + DATA_CHUNK * namelen, namelen) == 0)
- {
- DEBUGF("find 'data' chunk\n");
- fmt.numbytes = chunksize;
- if (fmt.formattag == WAVE_FORMAT_ATRAC3)
- id3->first_frame_offset = offset;
- }
- else if (memcmp(buf, chunknames + LIST_CHUNK * namelen, namelen) == 0)
- {
- DEBUGF("find 'LIST' chunk\n");
- parse_list_chunk(fd, id3, chunksize, is_64);
- lseek(fd, offset, SEEK_SET);
- }
-
- /* padded to next chunk */
- chunksize += ((is_64)? ((1 + ~chunksize) & 0x07) : (chunksize & 1));
-
- offset += chunksize;
- if (offset >= id3->filesize)
- break;
-
- lseek(fd, chunksize - read_data, SEEK_CUR);
- }
-
- if (fmt.numbytes == 0)
- {
- DEBUGF("metadata error: read error or missing 'data' chunk.\n");
- return false;
- }
-
- if (fmt.totalsamples == 0)
- set_totalsamples(&fmt, id3);
-
- if (id3->frequency == 0 || id3->bitrate == 0)
- {
- DEBUGF("metadata error: frequency or bitrate is 0\n");
- return false;
- }
-
- /* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
- id3->length = (fmt.formattag != WAVE_FORMAT_ATRAC3)?
- (uint64_t)fmt.totalsamples * 1000 / id3->frequency :
- ((id3->filesize - id3->first_frame_offset) * 8) / id3->bitrate;
-
- /* output header/id3 info (for debug) */
- DEBUGF("%s header info ----\n", (is_64)? "wave64" : "wave");
- DEBUGF(" format: %04x\n", (int)fmt.formattag);
- DEBUGF(" channels: %u\n", fmt.channels);
- DEBUGF(" blockalign: %u\n", fmt.blockalign);
- DEBUGF(" bitspersample: %u\n", fmt.bitspersample);
- DEBUGF(" samplesperblock: %u\n", fmt.samplesperblock);
- DEBUGF(" totalsamples: %u\n", (unsigned int)fmt.totalsamples);
- DEBUGF(" numbytes: %u\n", (unsigned int)fmt.numbytes);
- DEBUGF("id3 info ----\n");
- DEBUGF(" frequency: %u\n", (unsigned int)id3->frequency);
- DEBUGF(" bitrate: %d\n", id3->bitrate);
- DEBUGF(" length: %u\n", (unsigned int)id3->length);
-
- return true;
-}
-
-bool get_wave_metadata(int fd, struct mp3entry* id3)
-{
- return read_header(fd, id3, wave_chunklist, false);
-}
-
-bool get_wave64_metadata(int fd, struct mp3entry* id3)
-{
- return read_header(fd, id3, wave64_chunklist, true);
-}
diff --git a/apps/metadata/wavpack.c b/apps/metadata/wavpack.c
deleted file mode 100644
index f2811df8f3..0000000000
--- a/apps/metadata/wavpack.c
+++ /dev/null
@@ -1,160 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2007 David Bryant
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
-#include <inttypes.h>
-
-#include "system.h"
-#include "metadata.h"
-#include "metadata_common.h"
-#include "metadata_parsers.h"
-#include "logf.h"
-
-#define ID_UNIQUE 0x3f
-#define ID_LARGE 0x80
-#define ID_SAMPLE_RATE 0x27
-
-#define MONO_FLAG 4
-#define HYBRID_FLAG 8
-
-static const long wavpack_sample_rates [] =
-{
- 6000, 8000, 9600, 11025, 12000, 16000, 22050, 24000,
- 32000, 44100, 48000, 64000, 88200, 96000, 192000
-};
-
-/* A simple parser to read basic information from a WavPack file. This
- * now works with self-extrating WavPack files and also will scan the
- * metadata for non-standard sampling rates. This no longer fails on
- * WavPack files containing floating-point audio data because these are
- * now converted to standard Rockbox format in the decoder, and also
- * handles the case where up to 15 non-audio blocks might occur at the
- * beginning of the file.
- */
-
-bool get_wavpack_metadata(int fd, struct mp3entry* id3)
-{
- /* Use the trackname part of the id3 structure as a temporary buffer */
- unsigned char* buf = (unsigned char *)id3->path;
- uint32_t totalsamples = (uint32_t) -1;
- int i;
-
- for (i = 0; i < 256; ++i) {
-
- /* at every 256 bytes into file, try to read a WavPack header */
-
- if ((lseek(fd, i * 256, SEEK_SET) < 0) || (read(fd, buf, 32) < 32))
- return false;
-
- /* if valid WavPack 4 header version, break */
-
- if (memcmp (buf, "wvpk", 4) == 0 && buf [9] == 4 &&
- (buf [8] >= 2 && buf [8] <= 0x10))
- break;
- }
-
- if (i == 256) {
- logf ("Not a WavPack file");
- return false;
- }
-
- id3->vbr = true; /* All WavPack files are VBR */
- id3->filesize = filesize (fd);
-
- /* check up to 16 headers before we give up finding one with audio */
-
- for (i = 0; i < 16; ++i) {
- uint32_t meta_bytes = get_long_le(&buf [4]) - 24;
- uint32_t trial_totalsamples = get_long_le(&buf[12]);
- uint32_t blockindex = get_long_le(&buf[16]);
- uint32_t blocksamples = get_long_le(&buf[20]);
- uint32_t flags = get_long_le(&buf[24]);
-
- if (totalsamples == (uint32_t) -1 && blockindex == 0)
- totalsamples = trial_totalsamples;
-
- if (blocksamples) {
- int srindx = ((buf [26] >> 7) & 1) + ((buf [27] << 1) & 14);
-
- if (srindx == 15) {
- uint32_t meta_size;
-
- id3->frequency = 44100;
-
- while (meta_bytes >= 6) {
- if (read(fd, buf, 2) < 2)
- break;
-
- if (buf [0] & ID_LARGE) {
- if (read(fd, buf + 2, 2) < 2)
- break;
-
- meta_size = (buf [1] << 1) + (buf [2] << 9) + (buf [3] << 17);
- meta_bytes -= meta_size + 4;
- }
- else {
- meta_size = buf [1] << 1;
- meta_bytes -= meta_size + 2;
-
- if ((buf [0] & ID_UNIQUE) == ID_SAMPLE_RATE) {
- if (meta_size == 4 && read(fd, buf + 2, 4) == 4)
- id3->frequency = buf [2] + (buf [3] << 8) + (buf [4] << 16);
-
- break;
- }
- }
-
- if (meta_size > 0 && lseek(fd, meta_size, SEEK_CUR) < 0)
- break;
- }
- }
- else
- id3->frequency = wavpack_sample_rates[srindx];
-
- /* if the total number of samples is still unknown, make a guess on the high side (for now) */
-
- if (totalsamples == (uint32_t) -1) {
- totalsamples = id3->filesize * 3;
-
- if (!(flags & HYBRID_FLAG))
- totalsamples /= 2;
-
- if (!(flags & MONO_FLAG))
- totalsamples /= 2;
- }
-
- id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
- id3->bitrate = id3->filesize / (id3->length / 8);
-
- read_ape_tags(fd, id3);
- return true;
- }
- else { /* block did not contain audio, so seek to the end and see if there's another */
- if ((meta_bytes > 0 && lseek(fd, meta_bytes, SEEK_CUR) < 0) ||
- read(fd, buf, 32) < 32 || memcmp (buf, "wvpk", 4) != 0)
- break;
- }
- }
-
- return false;
-}
diff --git a/apps/mp3data.c b/apps/mp3data.c
deleted file mode 100644
index 13ff0a87a7..0000000000
--- a/apps/mp3data.c
+++ /dev/null
@@ -1,849 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2002 by Daniel Stenberg
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-/*
- * Parts of this code has been stolen from the Ample project and was written
- * by David Härdeman. It has since been extended and enhanced pretty much by
- * all sorts of friendly Rockbox people.
- *
- * A nice reference for MPEG header info:
- * http://rockbox.haxx.se/docs/mpeghdr.html
- *
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <stdbool.h>
-#include <limits.h>
-#include "debug.h"
-#include "logf.h"
-#include "mp3data.h"
-#include "file.h"
-#include "system.h"
-
-//#define DEBUG_VERBOSE
-
-#ifdef DEBUG_VERBOSE
-#define VDEBUGF DEBUGF
-#else
-#define VDEBUGF(...) do { } while(0)
-#endif
-
-#define SYNC_MASK (0x7ffL << 21)
-#define VERSION_MASK (3L << 19)
-#define LAYER_MASK (3L << 17)
-#define PROTECTION_MASK (1L << 16)
-#define BITRATE_MASK (0xfL << 12)
-#define SAMPLERATE_MASK (3L << 10)
-#define PADDING_MASK (1L << 9)
-#define PRIVATE_MASK (1L << 8)
-#define CHANNELMODE_MASK (3L << 6)
-#define MODE_EXT_MASK (3L << 4)
-#define COPYRIGHT_MASK (1L << 3)
-#define ORIGINAL_MASK (1L << 2)
-#define EMPHASIS_MASK (3L)
-
-/* Maximum number of bytes needed by Xing/Info/VBRI parser. */
-#define VBR_HEADER_MAX_SIZE (180)
-
-/* MPEG Version table, sorted by version index */
-static const signed char version_table[4] = {
- MPEG_VERSION2_5, -1, MPEG_VERSION2, MPEG_VERSION1
-};
-
-/* Bitrate table for mpeg audio, indexed by row index and birate index */
-static const short bitrates[5][16] = {
- {0,32,64,96,128,160,192,224,256,288,320,352,384,416,448,0}, /* V1 L1 */
- {0,32,48,56, 64, 80, 96,112,128,160,192,224,256,320,384,0}, /* V1 L2 */
- {0,32,40,48, 56, 64, 80, 96,112,128,160,192,224,256,320,0}, /* V1 L3 */
- {0,32,48,56, 64, 80, 96,112,128,144,160,176,192,224,256,0}, /* V2 L1 */
- {0, 8,16,24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160,0} /* V2 L2+L3 */
-};
-
-/* Bitrate pointer table, indexed by version and layer */
-static const short *bitrate_table[3][3] =
-{
- {bitrates[0], bitrates[1], bitrates[2]},
- {bitrates[3], bitrates[4], bitrates[4]},
- {bitrates[3], bitrates[4], bitrates[4]}
-};
-
-/* Sampling frequency table, indexed by version and frequency index */
-static const unsigned short freq_table[3][3] =
-{
- {44100, 48000, 32000}, /* MPEG Version 1 */
- {22050, 24000, 16000}, /* MPEG version 2 */
- {11025, 12000, 8000}, /* MPEG version 2.5 */
-};
-
-unsigned long bytes2int(unsigned long b0, unsigned long b1,
- unsigned long b2, unsigned long b3)
-{
- return (b0 & 0xFF) << (3*8) |
- (b1 & 0xFF) << (2*8) |
- (b2 & 0xFF) << (1*8) |
- (b3 & 0xFF) << (0*8);
-}
-
-/* check if 'head' is a valid mp3 frame header */
-static bool is_mp3frameheader(unsigned long head)
-{
- if ((head & SYNC_MASK) != (unsigned long)SYNC_MASK) /* bad sync? */
- return false;
- if ((head & VERSION_MASK) == (1L << 19)) /* bad version? */
- return false;
- if (!(head & LAYER_MASK)) /* no layer? */
- return false;
-#if CONFIG_CODEC != SWCODEC
- /* The MAS can't decode layer 1, so treat layer 1 data as invalid */
- if ((head & LAYER_MASK) == LAYER_MASK)
- return false;
-#endif
- if ((head & BITRATE_MASK) == BITRATE_MASK) /* bad bitrate? */
- return false;
- if (!(head & BITRATE_MASK)) /* no bitrate? */
- return false;
- if ((head & SAMPLERATE_MASK) == SAMPLERATE_MASK) /* bad sample rate? */
- return false;
-
- return true;
-}
-
-static bool mp3headerinfo(struct mp3info *info, unsigned long header)
-{
- int bitindex, freqindex;
-
- /* MPEG Audio Version */
- if ((header & VERSION_MASK) >> 19 >= sizeof(version_table))
- return false;
-
- info->version = version_table[(header & VERSION_MASK) >> 19];
- if (info->version < 0)
- return false;
-
- /* Layer */
- info->layer = 3 - ((header & LAYER_MASK) >> 17);
- if (info->layer == 3)
- return false;
-
-/* Rockbox: not used
- info->protection = (header & PROTECTION_MASK) ? true : false;
-*/
-
- /* Bitrate */
- bitindex = (header & BITRATE_MASK) >> 12;
- info->bitrate = bitrate_table[info->version][info->layer][bitindex];
- if(info->bitrate == 0)
- return false;
-
- /* Sampling frequency */
- freqindex = (header & SAMPLERATE_MASK) >> 10;
- if (freqindex == 3)
- return false;
- info->frequency = freq_table[info->version][freqindex];
-
- info->padding = (header & PADDING_MASK) ? 1 : 0;
-
- /* Calculate number of bytes, calculation depends on layer */
- if (info->layer == 0) {
- info->frame_samples = 384;
- info->frame_size = (12000 * info->bitrate / info->frequency
- + info->padding) * 4;
- }
- else {
- if ((info->version > MPEG_VERSION1) && (info->layer == 2))
- info->frame_samples = 576;
- else
- info->frame_samples = 1152;
- info->frame_size = (1000/8) * info->frame_samples * info->bitrate
- / info->frequency + info->padding;
- }
-
- /* Frametime fraction denominator */
- if (freqindex != 0) { /* 48/32/24/16/12/8 kHz */
- info->ft_den = 1; /* integer number of milliseconds */
- }
- else { /* 44.1/22.05/11.025 kHz */
- if (info->layer == 0) /* layer 1 */
- info->ft_den = 147;
- else /* layer 2+3 */
- info->ft_den = 49;
- }
- /* Frametime fraction numerator */
- info->ft_num = 1000 * info->ft_den * info->frame_samples / info->frequency;
-
- info->channel_mode = (header & CHANNELMODE_MASK) >> 6;
-/* Rockbox: not used
- info->mode_extension = (header & MODE_EXT_MASK) >> 4;
- info->emphasis = header & EMPHASIS_MASK;
-*/
- VDEBUGF( "Header: %08lx, Ver %d, lay %d, bitr %d, freq %ld, "
- "chmode %d, bytes: %d time: %d/%d\n",
- header, info->version, info->layer+1, info->bitrate,
- info->frequency, info->channel_mode,
- info->frame_size, info->ft_num, info->ft_den);
- return true;
-}
-
-static bool headers_have_same_type(unsigned long header1,
- unsigned long header2)
-{
- /* Compare MPEG version, layer and sampling frequency. If header1 is zero
- * it is assumed both frame headers are of same type. */
- unsigned int mask = SYNC_MASK | VERSION_MASK | LAYER_MASK | SAMPLERATE_MASK;
- header1 &= mask;
- header2 &= mask;
- return header1 ? (header1 == header2) : true;
-}
-
-/* Helper function to read 4-byte in big endian format. */
-static void read_uint32be_mp3data(int fd, unsigned long *data)
-{
-#ifdef ROCKBOX_BIG_ENDIAN
- (void)read(fd, (char*)data, 4);
-#else
- (void)read(fd, (char*)data, 4);
- *data = betoh32(*data);
-#endif
-}
-
-static unsigned long __find_next_frame(int fd, long *offset, long max_offset,
- unsigned long reference_header,
- int(*getfunc)(int fd, unsigned char *c),
- bool single_header)
-{
- unsigned long header=0;
- unsigned char tmp;
- long pos = 0;
-
- /* We will search until we find two consecutive MPEG frame headers with
- * the same MPEG version, layer and sampling frequency. The first header
- * of this pair is assumed to be the first valid MPEG frame header of the
- * whole stream. */
- do {
- /* Read 1 new byte. */
- header <<= 8;
- if (!getfunc(fd, &tmp))
- return 0;
- header |= tmp;
- pos++;
-
- /* Abort if max_offset is reached. Stop parsing. */
- if (max_offset > 0 && pos > max_offset)
- return 0;
-
- if (is_mp3frameheader(header)) {
- if (single_header) {
- /* We search for one _single_ valid header that has the same
- * type as the reference_header (if reference_header != 0).
- * In this case we are finished. */
- if (headers_have_same_type(reference_header, header))
- break;
- } else {
- /* The current header is valid. Now gather the frame size,
- * seek to this byte position and check if there is another
- * valid MPEG frame header of the same type. */
- struct mp3info info;
-
- /* Gather frame size from given header and seek to next
- * frame header. */
- mp3headerinfo(&info, header);
- lseek(fd, info.frame_size-4, SEEK_CUR);
-
- /* Read possible next frame header and seek back to last frame
- * headers byte position. */
- reference_header = 0;
- read_uint32be_mp3data(fd, &reference_header);
- //
- lseek(fd, -info.frame_size, SEEK_CUR);
-
- /* If the current header is of the same type as the previous
- * header we are finished. */
- if (headers_have_same_type(header, reference_header))
- break;
- }
- }
-
- } while (true);
-
- *offset = pos - 4;
-
- if(*offset)
- VDEBUGF("Warning: skipping %ld bytes of garbage\n", *offset);
-
- return header;
-}
-
-static int fileread(int fd, unsigned char *c)
-{
- return read(fd, c, 1);
-}
-
-unsigned long find_next_frame(int fd,
- long *offset,
- long max_offset,
- unsigned long reference_header)
-{
- return __find_next_frame(fd, offset, max_offset, reference_header,
- fileread, true);
-}
-
-#ifndef __PCTOOL__
-static int fnf_read_index;
-static int fnf_buf_len;
-static unsigned char *fnf_buf;
-
-static int buf_getbyte(int fd, unsigned char *c)
-{
- if(fnf_read_index < fnf_buf_len)
- {
- *c = fnf_buf[fnf_read_index++];
- return 1;
- }
- else
- {
- fnf_buf_len = read(fd, fnf_buf, fnf_buf_len);
- if(fnf_buf_len < 0)
- return -1;
-
- fnf_read_index = 0;
-
- if(fnf_buf_len > 0)
- {
- *c = fnf_buf[fnf_read_index++];
- return 1;
- }
- else
- return 0;
- }
- return 0;
-}
-
-static int buf_seek(int fd, int len)
-{
- fnf_read_index += len;
- if(fnf_read_index > fnf_buf_len)
- {
- len = fnf_read_index - fnf_buf_len;
-
- fnf_buf_len = read(fd, fnf_buf, fnf_buf_len);
- if(fnf_buf_len < 0)
- return -1;
-
- fnf_read_index = 0;
- fnf_read_index += len;
- }
-
- if(fnf_read_index > fnf_buf_len)
- {
- return -1;
- }
- else
- return 0;
-}
-
-static void buf_init(unsigned char* buf, size_t buflen)
-{
- fnf_buf = buf;
- fnf_buf_len = buflen;
- fnf_read_index = 0;
-}
-
-static unsigned long buf_find_next_frame(int fd, long *offset, long max_offset)
-{
- return __find_next_frame(fd, offset, max_offset, 0, buf_getbyte, true);
-}
-
-static size_t mem_buflen;
-static unsigned char* mem_buf;
-static size_t mem_pos;
-static int mem_cnt;
-static int mem_maxlen;
-
-static int mem_getbyte(int dummy, unsigned char *c)
-{
- (void)dummy;
-
- *c = mem_buf[mem_pos++];
- if(mem_pos >= mem_buflen)
- mem_pos = 0;
-
- if(mem_cnt++ >= mem_maxlen)
- return 0;
- else
- return 1;
-}
-
-unsigned long mem_find_next_frame(int startpos,
- long *offset,
- long max_offset,
- unsigned long reference_header,
- unsigned char* buf, size_t buflen)
-{
- mem_buf = buf;
- mem_buflen = buflen;
- mem_pos = startpos;
- mem_cnt = 0;
- mem_maxlen = max_offset;
-
- return __find_next_frame(0, offset, max_offset, reference_header,
- mem_getbyte, true);
-}
-#endif
-
-/* Extract information from a 'Xing' or 'Info' header. */
-static void get_xing_info(struct mp3info *info, unsigned char *buf)
-{
- int i = 8;
-
- /* Is it a VBR file? */
- info->is_vbr = !memcmp(buf, "Xing", 4);
-
- if (buf[7] & VBR_FRAMES_FLAG) /* Is the frame count there? */
- {
- info->frame_count = bytes2int(buf[i], buf[i+1], buf[i+2], buf[i+3]);
- if (info->frame_count <= ULONG_MAX / info->ft_num)
- info->file_time = info->frame_count * info->ft_num / info->ft_den;
- else
- info->file_time = info->frame_count / info->ft_den * info->ft_num;
- i += 4;
- }
-
- if (buf[7] & VBR_BYTES_FLAG) /* Is byte count there? */
- {
- info->byte_count = bytes2int(buf[i], buf[i+1], buf[i+2], buf[i+3]);
- i += 4;
- }
-
- if (info->file_time && info->byte_count)
- {
- if (info->byte_count <= (ULONG_MAX/8))
- info->bitrate = info->byte_count * 8 / info->file_time;
- else
- info->bitrate = info->byte_count / (info->file_time >> 3);
- }
-
- if (buf[7] & VBR_TOC_FLAG) /* Is table-of-contents there? */
- {
- info->has_toc = true;
- memcpy( info->toc, buf+i, 100 );
- i += 100;
- }
- if (buf[7] & VBR_QUALITY_FLAG)
- {
- /* We don't care about this, but need to skip it */
- i += 4;
- }
-#if CONFIG_CODEC==SWCODEC
- i += 21;
- info->enc_delay = ((int)buf[i ] << 4) | (buf[i+1] >> 4);
- info->enc_padding = ((int)(buf[i+1]&0xF) << 8) | buf[i+2];
- /* TODO: This sanity checking is rather silly, seeing as how the LAME
- header contains a CRC field that can be used to verify integrity. */
- if (!(info->enc_delay >= 0 && info->enc_delay <= 2880 &&
- info->enc_padding >= 0 && info->enc_padding <= 2*1152))
- {
- /* Invalid data */
- info->enc_delay = -1;
- info->enc_padding = -1;
- }
-#endif
-}
-
-/* Extract information from a 'VBRI' header. */
-static void get_vbri_info(struct mp3info *info, unsigned char *buf)
-{
- /* We don't parse the TOC, since we don't yet know how to (FIXME) */
- /*
- int i, num_offsets, offset = 0;
- */
-
- info->is_vbr = true; /* Yes, it is a FhG VBR file */
- info->has_toc = false; /* We don't parse the TOC (yet) */
-
- info->byte_count = bytes2int(buf[10], buf[11], buf[12], buf[13]);
- info->frame_count = bytes2int(buf[14], buf[15], buf[16], buf[17]);
- if (info->frame_count <= ULONG_MAX / info->ft_num)
- info->file_time = info->frame_count * info->ft_num / info->ft_den;
- else
- info->file_time = info->frame_count / info->ft_den * info->ft_num;
-
- if (info->byte_count <= (ULONG_MAX/8))
- info->bitrate = info->byte_count * 8 / info->file_time;
- else
- info->bitrate = info->byte_count / (info->file_time >> 3);
-
- VDEBUGF("Frame size (%dkpbs): %d bytes (0x%x)\n",
- info->bitrate, info->frame_size, info->frame_size);
- VDEBUGF("Frame count: %lx\n", info->frame_count);
- VDEBUGF("Byte count: %lx\n", info->byte_count);
-
- /* We don't parse the TOC, since we don't yet know how to (FIXME) */
- /*
- num_offsets = bytes2int(0, 0, buf[18], buf[19]);
- VDEBUGF("Offsets: %d\n", num_offsets);
- VDEBUGF("Frames/entry: %ld\n", bytes2int(0, 0, buf[24], buf[25]));
-
- for(i = 0; i < num_offsets; i++)
- {
- offset += bytes2int(0, 0, buf[26+i*2], buf[27+i*2]);;
- VDEBUGF("%03d: %lx\n", i, offset - bytecount,);
- }
- */
-}
-
-/* Seek to next mpeg header and extract relevant information. */
-static int get_next_header_info(int fd, long *bytecount, struct mp3info *info,
- bool single_header)
-{
- long tmp;
- unsigned long header = 0;
-
- header = __find_next_frame(fd, &tmp, 0x20000, 0, fileread, single_header);
- if(header == 0)
- return -1;
-
- if(!mp3headerinfo(info, header))
- return -2;
-
- /* Next frame header is tmp bytes away. */
- *bytecount += tmp;
-
- return 0;
-}
-
-int get_mp3file_info(int fd, struct mp3info *info)
-{
- unsigned char frame[VBR_HEADER_MAX_SIZE], *vbrheader;
- long bytecount = 0;
- int result, buf_size;
-
- /* Initialize info and frame */
- memset(info, 0, sizeof(struct mp3info));
- memset(frame, 0, sizeof(frame));
-
-#if CONFIG_CODEC==SWCODEC
- /* These two are needed for proper LAME gapless MP3 playback */
- info->enc_delay = -1;
- info->enc_padding = -1;
-#endif
-
- /* Get the very first single MPEG frame. */
- result = get_next_header_info(fd, &bytecount, info, true);
- if(result)
- return result;
-
- /* Read the amount of frame data to the buffer that is required for the
- * vbr tag parsing. Skip the rest. */
- buf_size = MIN(info->frame_size-4, (int)sizeof(frame));
- if(read(fd, frame, buf_size) < 0)
- return -3;
- lseek(fd, info->frame_size - 4 - buf_size, SEEK_CUR);
-
- /* Calculate position of a possible VBR header */
- if (info->version == MPEG_VERSION1) {
- if (info->channel_mode == 3) /* mono */
- vbrheader = frame + 17;
- else
- vbrheader = frame + 32;
- } else {
- if (info->channel_mode == 3) /* mono */
- vbrheader = frame + 9;
- else
- vbrheader = frame + 17;
- }
-
- if (!memcmp(vbrheader, "Xing", 4) || !memcmp(vbrheader, "Info", 4))
- {
- VDEBUGF("-- XING header --\n");
-
- /* We want to skip the Xing frame when playing the stream */
- bytecount += info->frame_size;
-
- /* Now get the next frame to read the real info about the mp3 stream */
- result = get_next_header_info(fd, &bytecount, info, false);
- if(result)
- return result;
-
- get_xing_info(info, vbrheader);
- }
- else if (!memcmp(vbrheader, "VBRI", 4))
- {
- VDEBUGF("-- VBRI header --\n");
-
- /* We want to skip the VBRI frame when playing the stream */
- bytecount += info->frame_size;
-
- /* Now get the next frame to read the real info about the mp3 stream */
- result = get_next_header_info(fd, &bytecount, info, false);
- if(result)
- return result;
-
- get_vbri_info(info, vbrheader);
- }
- else
- {
- VDEBUGF("-- No VBR header --\n");
-
- /* There was no VBR header found. So, we seek back to beginning and
- * search for the first MPEG frame header of the mp3 stream. */
- lseek(fd, -info->frame_size, SEEK_CUR);
- result = get_next_header_info(fd, &bytecount, info, false);
- if(result)
- return result;
- }
-
- return bytecount;
-}
-
-#ifndef __PCTOOL__
-static void long2bytes(unsigned char *buf, long val)
-{
- buf[0] = (val >> 24) & 0xff;
- buf[1] = (val >> 16) & 0xff;
- buf[2] = (val >> 8) & 0xff;
- buf[3] = val & 0xff;
-}
-
-int count_mp3_frames(int fd, int startpos, int filesize,
- void (*progressfunc)(int),
- unsigned char* buf, size_t buflen)
-{
- unsigned long header = 0;
- struct mp3info info;
- int num_frames;
- long bytes;
- int cnt;
- long progress_chunk = filesize / 50; /* Max is 50%, in 1% increments */
- int progress_cnt = 0;
- bool is_vbr = false;
- int last_bitrate = 0;
- int header_template = 0;
-
- if(lseek(fd, startpos, SEEK_SET) < 0)
- return -1;
-
- buf_init(buf, buflen);
-
- /* Find out the total number of frames */
- num_frames = 0;
- cnt = 0;
-
- while((header = buf_find_next_frame(fd, &bytes, header_template))) {
- mp3headerinfo(&info, header);
-
- if(!header_template)
- header_template = header;
-
- /* See if this really is a VBR file */
- if(last_bitrate && info.bitrate != last_bitrate)
- {
- is_vbr = true;
- }
- last_bitrate = info.bitrate;
-
- buf_seek(fd, info.frame_size-4);
- num_frames++;
- if(progressfunc)
- {
- cnt += bytes + info.frame_size;
- if(cnt > progress_chunk)
- {
- progress_cnt++;
- progressfunc(progress_cnt);
- cnt = 0;
- }
- }
- }
- VDEBUGF("Total number of frames: %d\n", num_frames);
-
- if(is_vbr)
- return num_frames;
- else
- {
- DEBUGF("Not a VBR file\n");
- return 0;
- }
-}
-
-static const char cooltext[] = "Rockbox - rocks your box";
-
-/* buf needs to be the audio buffer with TOC generation enabled,
- and at least MAX_XING_HEADER_SIZE bytes otherwise */
-int create_xing_header(int fd, long startpos, long filesize,
- unsigned char *buf, unsigned long num_frames,
- unsigned long rec_time, unsigned long header_template,
- void (*progressfunc)(int), bool generate_toc,
- unsigned char *tempbuf, size_t tempbuflen )
-{
- struct mp3info info;
- unsigned char toc[100];
- unsigned long header = 0;
- unsigned long xing_header_template = header_template;
- unsigned long filepos;
- long pos, last_pos;
- long j;
- long bytes;
- int i;
- int index;
-
- DEBUGF("create_xing_header()\n");
-
- if(generate_toc)
- {
- lseek(fd, startpos, SEEK_SET);
- buf_init(tempbuf, tempbuflen);
-
- /* Generate filepos table */
- last_pos = 0;
- filepos = 0;
- header = 0;
- for(i = 0;i < 100;i++) {
- /* Calculate the absolute frame number for this seek point */
- pos = i * num_frames / 100;
-
- /* Advance from the last seek point to this one */
- for(j = 0;j < pos - last_pos;j++)
- {
- header = buf_find_next_frame(fd, &bytes, header_template);
- filepos += bytes;
- mp3headerinfo(&info, header);
- buf_seek(fd, info.frame_size-4);
- filepos += info.frame_size;
-
- if(!header_template)
- header_template = header;
- }
-
- /* Save a header for later use if header_template is empty.
- We only save one header, and we want to save one in the
- middle of the stream, just in case the first and the last
- headers are corrupt. */
- if(!xing_header_template && i == 1)
- xing_header_template = header;
-
- if(progressfunc)
- {
- progressfunc(50 + i/2);
- }
-
- /* Fill in the TOC entry */
- /* each toc is a single byte indicating how many 256ths of the
- * way through the file, is that percent of the way through the
- * song. the easy method, filepos*256/filesize, chokes when
- * the upper 8 bits of the file position are nonzero
- * (i.e. files over 16mb in size).
- */
- if (filepos > (ULONG_MAX/256))
- {
- /* instead of multiplying filepos by 256, we divide
- * filesize by 256.
- */
- toc[i] = filepos / (filesize >> 8);
- }
- else
- {
- toc[i] = filepos * 256 / filesize;
- }
-
- VDEBUGF("Pos %d: %ld relpos: %ld filepos: %lx tocentry: %x\n",
- i, pos, pos-last_pos, filepos, toc[i]);
-
- last_pos = pos;
- }
- }
-
- /* Use the template header and create a new one.
- We ignore the Protection bit even if the rest of the stream is
- protected. */
- header = xing_header_template & ~(BITRATE_MASK|PROTECTION_MASK|PADDING_MASK);
- header |= 8 << 12; /* This gives us plenty of space, 192..576 bytes */
-
- if (!mp3headerinfo(&info, header))
- return 0; /* invalid header */
-
- if (num_frames == 0 && rec_time) {
- /* estimate the number of frames based on the recording time */
- if (rec_time <= ULONG_MAX / info.ft_den)
- num_frames = rec_time * info.ft_den / info.ft_num;
- else
- num_frames = rec_time / info.ft_num * info.ft_den;
- }
-
- /* Clear the frame */
- memset(buf, 0, MAX_XING_HEADER_SIZE);
-
- /* Write the header to the buffer */
- long2bytes(buf, header);
-
- /* Calculate position of VBR header */
- if (info.version == MPEG_VERSION1) {
- if (info.channel_mode == 3) /* mono */
- index = 21;
- else
- index = 36;
- }
- else {
- if (info.channel_mode == 3) /* mono */
- index = 13;
- else
- index = 21;
- }
-
- /* Create the Xing data */
- memcpy(&buf[index], "Xing", 4);
- long2bytes(&buf[index+4], (num_frames ? VBR_FRAMES_FLAG : 0)
- | (filesize ? VBR_BYTES_FLAG : 0)
- | (generate_toc ? VBR_TOC_FLAG : 0));
- index += 8;
- if(num_frames)
- {
- long2bytes(&buf[index], num_frames);
- index += 4;
- }
-
- if(filesize)
- {
- long2bytes(&buf[index], filesize - startpos);
- index += 4;
- }
-
- /* Copy the TOC */
- memcpy(buf + index, toc, 100);
-
- /* And some extra cool info */
- memcpy(buf + index + 100, cooltext, sizeof(cooltext));
-
-#ifdef DEBUG
- for(i = 0;i < info.frame_size;i++)
- {
- if(i && !(i % 16))
- DEBUGF("\n");
-
- DEBUGF("%02x ", buf[i]);
- }
-#endif
-
- return info.frame_size;
-}
-
-#endif
diff --git a/apps/mp3data.h b/apps/mp3data.h
deleted file mode 100644
index 762c2f4583..0000000000
--- a/apps/mp3data.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2002 by Linus Nielsen Feltzing
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _MP3DATA_H_
-#define _MP3DATA_H_
-
-#define MPEG_VERSION1 0
-#define MPEG_VERSION2 1
-#define MPEG_VERSION2_5 2
-
-#include <string.h> /* size_t */
-
-struct mp3info {
- /* Standard MP3 frame header fields */
- int version;
- int layer;
- int bitrate;
- long frequency;
- int padding;
- int channel_mode;
- int frame_size; /* Frame size in bytes */
- int frame_samples;/* Samples per frame */
- int ft_num; /* Numerator of frametime in milliseconds */
- int ft_den; /* Denominator of frametime in milliseconds */
-
- bool is_vbr; /* True if the file is VBR */
- bool has_toc; /* True if there is a VBR header in the file */
- unsigned char toc[100];
- unsigned long frame_count; /* Number of frames in the file (if VBR) */
- unsigned long byte_count; /* File size in bytes */
- unsigned long file_time; /* Length of the whole file in milliseconds */
- int enc_delay; /* Encoder delay, fetched from LAME header */
- int enc_padding; /* Padded samples added to last frame. LAME header */
-};
-
-/* Xing header information */
-#define VBR_FRAMES_FLAG 0x01
-#define VBR_BYTES_FLAG 0x02
-#define VBR_TOC_FLAG 0x04
-#define VBR_QUALITY_FLAG 0x08
-
-#define MAX_XING_HEADER_SIZE 576
-
-unsigned long find_next_frame(int fd,
- long *offset,
- long max_offset,
- unsigned long reference_header);
-unsigned long mem_find_next_frame(int startpos,
- long *offset,
- long max_offset,
- unsigned long reference_header,
- unsigned char* buf, size_t buflen);
-int get_mp3file_info(int fd,
- struct mp3info *info);
-
-int count_mp3_frames(int fd, int startpos, int filesize,
- void (*progressfunc)(int),
- unsigned char* buf, size_t buflen);
-
-int create_xing_header(int fd, long startpos, long filesize,
- unsigned char *buf, unsigned long num_frames,
- unsigned long rec_time, unsigned long header_template,
- void (*progressfunc)(int), bool generate_toc,
- unsigned char *tempbuf, size_t tempbuflen );
-
-extern unsigned long bytes2int(unsigned long b0,
- unsigned long b1,
- unsigned long b2,
- unsigned long b3);
-
-#endif
diff --git a/apps/plugins/lrcplayer.c b/apps/plugins/lrcplayer.c
index cc0128b401..97385ff047 100644
--- a/apps/plugins/lrcplayer.c
+++ b/apps/plugins/lrcplayer.c
@@ -1113,7 +1113,6 @@ static void load_lrc_file(void)
/*******************************
* read lyrics from id3
*******************************/
-/* taken from apps/metadata/mp3.c */
static unsigned long unsync(unsigned long b0, unsigned long b1,
unsigned long b2, unsigned long b3)
{
diff --git a/apps/replaygain.c b/apps/replaygain.c
deleted file mode 100644
index a178321385..0000000000
--- a/apps/replaygain.c
+++ /dev/null
@@ -1,222 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Magnus Holmgren
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <ctype.h>
-#include <math.h>
-#include <stdbool.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <inttypes.h>
-#include "strlcpy.h"
-#include "strcasecmp.h"
-#include "system.h"
-#include "metadata.h"
-#include "debug.h"
-#include "replaygain.h"
-#include "fixedpoint.h"
-
-#define FP_BITS (12)
-#define FP_ONE (1 << FP_BITS)
-#define FP_MIN (-48 * FP_ONE)
-#define FP_MAX ( 17 * FP_ONE)
-
-void replaygain_itoa(char* buffer, int length, long int_gain)
-{
- /* int_gain uses Q19.12 format. */
- int one = abs(int_gain) >> FP_BITS;
- int cent = ((abs(int_gain) & 0x0fff) * 100 + (FP_ONE/2)) >> FP_BITS;
- snprintf(buffer, length, "%s%d.%02d dB", (int_gain<0) ? "-":"", one, cent);
-}
-
-static long fp_atof(const char* s, int precision)
-{
- long int_part = 0;
- long int_one = BIT_N(precision);
- long frac_part = 0;
- long frac_count = 0;
- long frac_max = ((precision * 4) + 12) / 13;
- long frac_max_int = 1;
- long sign = 1;
- bool point = false;
-
- while ((*s != '\0') && isspace(*s))
- {
- s++;
- }
-
- if (*s == '-')
- {
- sign = -1;
- s++;
- }
- else if (*s == '+')
- {
- s++;
- }
-
- while (*s != '\0')
- {
- if (*s == '.')
- {
- if (point)
- {
- break;
- }
-
- point = true;
- }
- else if (isdigit(*s))
- {
- if (point)
- {
- if (frac_count < frac_max)
- {
- frac_part = frac_part * 10 + (*s - '0');
- frac_count++;
- frac_max_int *= 10;
- }
- }
- else
- {
- int_part = int_part * 10 + (*s - '0');
- }
- }
- else
- {
- break;
- }
-
- s++;
- }
-
- while (frac_count < frac_max)
- {
- frac_part *= 10;
- frac_count++;
- frac_max_int *= 10;
- }
-
- return sign * ((int_part * int_one)
- + (((int64_t) frac_part * int_one) / frac_max_int));
-}
-
-static long convert_gain(long gain)
-{
- /* Don't allow unreasonably low or high gain changes.
- * Our math code can't handle it properly anyway. :) */
- gain = MAX(gain, FP_MIN);
- gain = MIN(gain, FP_MAX);
-
- return fp_factor(gain, FP_BITS) << (24 - FP_BITS);
-}
-
-/* Get the sample scale factor in Q19.12 format from a gain value. Returns 0
- * for no gain.
- *
- * str Gain in dB as a string. E.g., "-3.45 dB"; the "dB" part is ignored.
- */
-static long get_replaygain(const char* str)
-{
- return fp_atof(str, FP_BITS);
-}
-
-/* Get the peak volume in Q7.24 format.
- *
- * str Peak volume. Full scale is specified as "1.0". Returns 0 for no peak.
- */
-static long get_replaypeak(const char* str)
-{
- return fp_atof(str, 24);
-}
-
-/* Get a sample scale factor in Q7.24 format from a gain value.
- *
- * int_gain Gain in dB, multiplied by 100.
- */
-long get_replaygain_int(long int_gain)
-{
- return convert_gain(int_gain * FP_ONE / 100);
-}
-
-/* Parse a ReplayGain tag conforming to the "VorbisGain standard". If a
- * valid tag is found, update mp3entry struct accordingly. Existing values
- * are not overwritten.
- *
- * key Name of the tag.
- * value Value of the tag.
- * entry mp3entry struct to update.
- */
-void parse_replaygain(const char* key, const char* value,
- struct mp3entry* entry)
-{
- if (((strcasecmp(key, "replaygain_track_gain") == 0) ||
- (strcasecmp(key, "rg_radio") == 0)) &&
- !entry->track_gain)
- {
- entry->track_level = get_replaygain(value);
- entry->track_gain = convert_gain(entry->track_level);
- }
- else if (((strcasecmp(key, "replaygain_album_gain") == 0) ||
- (strcasecmp(key, "rg_audiophile") == 0)) &&
- !entry->album_gain)
- {
- entry->album_level = get_replaygain(value);
- entry->album_gain = convert_gain(entry->album_level);
- }
- else if (((strcasecmp(key, "replaygain_track_peak") == 0) ||
- (strcasecmp(key, "rg_peak") == 0)) &&
- !entry->track_peak)
- {
- entry->track_peak = get_replaypeak(value);
- }
- else if ((strcasecmp(key, "replaygain_album_peak") == 0) &&
- !entry->album_peak)
- {
- entry->album_peak = get_replaypeak(value);
- }
-}
-
-/* Set ReplayGain values from integers. Existing values are not overwritten.
- *
- * album If true, set album values, otherwise set track values.
- * gain Gain value in dB, multiplied by 512. 0 for no gain.
- * peak Peak volume in Q7.24 format, where 1.0 is full scale. 0 for no
- * peak volume.
- * entry mp3entry struct to update.
- */
-void parse_replaygain_int(bool album, long gain, long peak,
- struct mp3entry* entry)
-{
- gain = gain * FP_ONE / 512;
-
- if (album)
- {
- entry->album_level = gain;
- entry->album_gain = convert_gain(gain);
- entry->album_peak = peak;
- }
- else
- {
- entry->track_level = gain;
- entry->track_gain = convert_gain(gain);
- entry->track_peak = peak;
- }
-}
diff --git a/apps/replaygain.h b/apps/replaygain.h
deleted file mode 100644
index 215464dfdf..0000000000
--- a/apps/replaygain.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2005 Magnus Holmgren
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _REPLAYGAIN_H
-#define _REPLAYGAIN_H
-
-#include "metadata.h"
-
-long get_replaygain_int(long int_gain);
-void parse_replaygain(const char* key, const char* value,
- struct mp3entry* entry);
-void parse_replaygain_int(bool album, long gain, long peak,
- struct mp3entry* entry);
-void replaygain_itoa(char* buffer, int length, long int_gain);
-
-#endif
diff --git a/apps/tdspeed.c b/apps/tdspeed.c
deleted file mode 100644
index 731be12621..0000000000
--- a/apps/tdspeed.c
+++ /dev/null
@@ -1,450 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006 by Nicolas Pitre <nico@cam.org>
- * Copyright (C) 2006-2007 by Stéphane Doyon <s.doyon@videotron.ca>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#include <inttypes.h>
-#include <stddef.h>
-#include <stdio.h>
-#include <string.h>
-#include "sound.h"
-#include "core_alloc.h"
-#include "system.h"
-#include "tdspeed.h"
-#include "settings.h"
-
-#define assert(cond)
-
-#define MIN_RATE 8000
-#define MAX_RATE 48000 /* double buffer for double rate */
-#define MINFREQ 100
-
-#define FIXED_BUFSIZE 3072 /* 48KHz factor 3.0 */
-
-static int32_t** dsp_src;
-static int handles[4];
-static int32_t *overlap_buffer[2] = { NULL, NULL };
-static int32_t *outbuf[2] = { NULL, NULL };
-
-static int move_callback(int handle, void* current, void* new)
-{
- /* TODO */
- (void)handle;
- if (dsp_src)
- {
- int ch = (current == outbuf[0]) ? 0 : 1;
- dsp_src[ch] = outbuf[ch] = new;
- }
- return BUFLIB_CB_OK;
-}
-
-static struct buflib_callbacks ops = {
- .move_callback = move_callback,
- .shrink_callback = NULL,
-};
-
-static int ovl_move_callback(int handle, void* current, void* new)
-{
- /* TODO */
- (void)handle;
- if (dsp_src)
- {
- int ch = (current == overlap_buffer[0]) ? 0 : 1;
- overlap_buffer[ch] = new;
- }
- return BUFLIB_CB_OK;
-}
-
-static struct buflib_callbacks ovl_ops = {
- .move_callback = ovl_move_callback,
- .shrink_callback = NULL,
-};
-
-
-static struct tdspeed_state_s
-{
- bool stereo;
- int32_t shift_max; /* maximum displacement on a frame */
- int32_t src_step; /* source window pace */
- int32_t dst_step; /* destination window pace */
- int32_t dst_order; /* power of two for dst_step */
- int32_t ovl_shift; /* overlap buffer frame shift */
- int32_t ovl_size; /* overlap buffer used size */
- int32_t ovl_space; /* overlap buffer size */
- int32_t *ovl_buff[2]; /* overlap buffer */
-} tdspeed_state;
-
-void tdspeed_init(void)
-{
- if (!global_settings.timestretch_enabled)
- return;
-
- /* Allocate buffers */
- if (overlap_buffer[0] == NULL)
- {
- handles[0] = core_alloc_ex("tdspeed ovl left", FIXED_BUFSIZE * sizeof(int32_t), &ovl_ops);
- overlap_buffer[0] = core_get_data(handles[0]);
- }
- if (overlap_buffer[1] == NULL)
- {
- handles[1] = core_alloc_ex("tdspeed ovl right", FIXED_BUFSIZE * sizeof(int32_t), &ovl_ops);
- overlap_buffer[1] = core_get_data(handles[1]);
- }
- if (outbuf[0] == NULL)
- {
- handles[2] = core_alloc_ex("tdspeed left", TDSPEED_OUTBUFSIZE * sizeof(int32_t), &ops);
- outbuf[0] = core_get_data(handles[2]);
- }
- if (outbuf[1] == NULL)
- {
- handles[3] = core_alloc_ex("tdspeed right", TDSPEED_OUTBUFSIZE * sizeof(int32_t), &ops);
- outbuf[1] = core_get_data(handles[3]);
- }
-}
-
-void tdspeed_finish(void)
-{
- for(unsigned i = 0; i < ARRAYLEN(handles); i++)
- {
- if (handles[i] > 0)
- {
- core_free(handles[i]);
- handles[i] = 0;
- }
- }
- overlap_buffer[0] = overlap_buffer[1] = NULL;
- outbuf[0] = outbuf[1] = NULL;
-}
-
-bool tdspeed_config(int samplerate, bool stereo, int32_t factor)
-{
- struct tdspeed_state_s *st = &tdspeed_state;
- int src_frame_sz;
-
- /* Check buffers were allocated ok */
- if (overlap_buffer[0] == NULL || overlap_buffer[1] == NULL)
- return false;
-
- if (outbuf[0] == NULL || outbuf[1] == NULL)
- return false;
-
- /* Check parameters */
- if (factor == PITCH_SPEED_100)
- return false;
-
- if (samplerate < MIN_RATE || samplerate > MAX_RATE)
- return false;
-
- if (factor < STRETCH_MIN || factor > STRETCH_MAX)
- return false;
-
- st->stereo = stereo;
- st->dst_step = samplerate / MINFREQ;
-
- if (factor > PITCH_SPEED_100)
- st->dst_step = st->dst_step * PITCH_SPEED_100 / factor;
-
- st->dst_order = 1;
-
- while (st->dst_step >>= 1)
- st->dst_order++;
-
- st->dst_step = (1 << st->dst_order);
- st->src_step = st->dst_step * factor / PITCH_SPEED_100;
- st->shift_max = (st->dst_step > st->src_step) ? st->dst_step : st->src_step;
-
- src_frame_sz = st->shift_max + st->dst_step;
-
- if (st->dst_step > st->src_step)
- src_frame_sz += st->dst_step - st->src_step;
-
- st->ovl_space = ((src_frame_sz - 2) / st->src_step) * st->src_step
- + src_frame_sz;
-
- if (st->src_step > st->dst_step)
- st->ovl_space += 2*st->src_step - st->dst_step;
-
- if (st->ovl_space > FIXED_BUFSIZE)
- st->ovl_space = FIXED_BUFSIZE;
-
- st->ovl_size = 0;
- st->ovl_shift = 0;
-
- st->ovl_buff[0] = overlap_buffer[0];
-
- if (stereo)
- st->ovl_buff[1] = overlap_buffer[1];
- else
- st->ovl_buff[1] = st->ovl_buff[0];
-
- return true;
-}
-
-static int tdspeed_apply(int32_t *buf_out[2], int32_t *buf_in[2],
- int data_len, int last, int out_size)
-/* data_len in samples */
-{
- struct tdspeed_state_s *st = &tdspeed_state;
- int32_t *dest[2];
- int32_t next_frame, prev_frame, src_frame_sz;
- bool stereo = buf_in[0] != buf_in[1];
-
- assert(stereo == st->stereo);
-
- src_frame_sz = st->shift_max + st->dst_step;
-
- if (st->dst_step > st->src_step)
- src_frame_sz += st->dst_step - st->src_step;
-
- /* deal with overlap data first, if any */
- if (st->ovl_size)
- {
- int32_t have, copy, steps;
- have = st->ovl_size;
-
- if (st->ovl_shift > 0)
- have -= st->ovl_shift;
-
- /* append just enough data to have all of the overlap buffer consumed */
- steps = (have - 1) / st->src_step;
- copy = steps * st->src_step + src_frame_sz - have;
-
- if (copy < src_frame_sz - st->dst_step)
- copy += st->src_step; /* one more step to allow for pregap data */
-
- if (copy > data_len)
- copy = data_len;
-
- assert(st->ovl_size + copy <= FIXED_BUFSIZE);
- memcpy(st->ovl_buff[0] + st->ovl_size, buf_in[0],
- copy * sizeof(int32_t));
-
- if (stereo)
- memcpy(st->ovl_buff[1] + st->ovl_size, buf_in[1],
- copy * sizeof(int32_t));
-
- if (!last && have + copy < src_frame_sz)
- {
- /* still not enough to process at least one frame */
- st->ovl_size += copy;
- return 0;
- }
-
- /* recursively call ourselves to process the overlap buffer */
- have = st->ovl_size;
- st->ovl_size = 0;
-
- if (copy == data_len)
- {
- assert(have + copy <= FIXED_BUFSIZE);
- return tdspeed_apply(buf_out, st->ovl_buff, have+copy, last,
- out_size);
- }
-
- assert(have + copy <= FIXED_BUFSIZE);
- int i = tdspeed_apply(buf_out, st->ovl_buff, have+copy, -1, out_size);
-
- dest[0] = buf_out[0] + i;
- dest[1] = buf_out[1] + i;
-
- /* readjust pointers to account for data already consumed */
- next_frame = copy - src_frame_sz + st->src_step;
- prev_frame = next_frame - st->ovl_shift;
- }
- else
- {
- dest[0] = buf_out[0];
- dest[1] = buf_out[1];
-
- next_frame = prev_frame = 0;
-
- if (st->ovl_shift > 0)
- next_frame += st->ovl_shift;
- else
- prev_frame += -st->ovl_shift;
- }
-
- st->ovl_shift = 0;
-
- /* process all complete frames */
- while (data_len - next_frame >= src_frame_sz)
- {
- /* find frame overlap by autocorelation */
- int const INC1 = 8;
- int const INC2 = 32;
-
- int64_t min_delta = INT64_MAX; /* most positive */
- int shift = 0;
-
- /* Power of 2 of a 28bit number requires 56bits, can accumulate
- 256times in a 64bit variable. */
- assert(st->dst_step / INC2 <= 256);
- assert(next_frame + st->shift_max - 1 + st->dst_step - 1 < data_len);
- assert(prev_frame + st->dst_step - 1 < data_len);
-
- for (int i = 0; i < st->shift_max; i += INC1)
- {
- int64_t delta = 0;
-
- int32_t *curr = buf_in[0] + next_frame + i;
- int32_t *prev = buf_in[0] + prev_frame;
-
- for (int j = 0; j < st->dst_step; j += INC2, curr += INC2, prev += INC2)
- {
- int32_t diff = *curr - *prev;
- delta += abs(diff);
-
- if (delta >= min_delta)
- goto skip;
- }
-
- if (stereo)
- {
- curr = buf_in[1] + next_frame + i;
- prev = buf_in[1] + prev_frame;
-
- for (int j = 0; j < st->dst_step; j += INC2, curr += INC2, prev += INC2)
- {
- int32_t diff = *curr - *prev;
- delta += abs(diff);
-
- if (delta >= min_delta)
- goto skip;
- }
- }
-
- min_delta = delta;
- shift = i;
-skip:;
- }
-
- /* overlap fading-out previous frame with fading-in current frame */
- int32_t *curr = buf_in[0] + next_frame + shift;
- int32_t *prev = buf_in[0] + prev_frame;
-
- int32_t *d = dest[0];
-
- assert(next_frame + shift + st->dst_step - 1 < data_len);
- assert(prev_frame + st->dst_step - 1 < data_len);
- assert(dest[0] - buf_out[0] + st->dst_step - 1 < out_size);
-
- for (int i = 0, j = st->dst_step; j; i++, j--)
- {
- *d++ = (*curr++ * (int64_t)i +
- *prev++ * (int64_t)j) >> st->dst_order;
- }
-
- dest[0] = d;
-
- if (stereo)
- {
- curr = buf_in[1] + next_frame + shift;
- prev = buf_in[1] + prev_frame;
-
- d = dest[1];
-
- for (int i = 0, j = st->dst_step; j; i++, j--)
- {
- assert(d < buf_out[1] + out_size);
-
- *d++ = (*curr++ * (int64_t)i +
- *prev++ * (int64_t)j) >> st->dst_order;
- }
-
- dest[1] = d;
- }
-
- /* adjust pointers for next frame */
- prev_frame = next_frame + shift + st->dst_step;
- next_frame += st->src_step;
-
- /* here next_frame - prev_frame = src_step - dst_step - shift */
- assert(next_frame - prev_frame == st->src_step - st->dst_step - shift);
- }
-
- /* now deal with remaining partial frames */
- if (last == -1)
- {
- /* special overlap buffer processing: remember frame shift only */
- st->ovl_shift = next_frame - prev_frame;
- }
- else if (last != 0)
- {
- /* last call: purge all remaining data to output buffer */
- int i = data_len - prev_frame;
-
- assert(dest[0] + i <= buf_out[0] + out_size);
- memcpy(dest[0], buf_in[0] + prev_frame, i * sizeof(int32_t));
-
- dest[0] += i;
-
- if (stereo)
- {
- assert(dest[1] + i <= buf_out[1] + out_size);
- memcpy(dest[1], buf_in[1] + prev_frame, i * sizeof(int32_t));
- dest[1] += i;
- }
- }
- else
- {
- /* preserve remaining data + needed overlap data for next call */
- st->ovl_shift = next_frame - prev_frame;
- int i = (st->ovl_shift < 0) ? next_frame : prev_frame;
- st->ovl_size = data_len - i;
-
- assert(st->ovl_size <= FIXED_BUFSIZE);
- memcpy(st->ovl_buff[0], buf_in[0] + i, st->ovl_size * sizeof(int32_t));
-
- if (stereo)
- memcpy(st->ovl_buff[1], buf_in[1] + i, st->ovl_size * sizeof(int32_t));
- }
-
- return dest[0] - buf_out[0];
-}
-
-long tdspeed_est_output_size()
-{
- return TDSPEED_OUTBUFSIZE;
-}
-
-long tdspeed_est_input_size(long size)
-{
- struct tdspeed_state_s *st = &tdspeed_state;
-
- size = (size - st->ovl_size) * st->src_step / st->dst_step;
-
- if (size < 0)
- size = 0;
-
- return size;
-}
-
-int tdspeed_doit(int32_t *src[], int count)
-{
- dsp_src = src;
- count = tdspeed_apply( (int32_t *[2]) { outbuf[0], outbuf[1] },
- src, count, 0, TDSPEED_OUTBUFSIZE);
-
- src[0] = outbuf[0];
- src[1] = outbuf[1];
-
- return count;
-}
-
diff --git a/apps/tdspeed.h b/apps/tdspeed.h
deleted file mode 100644
index e91eeb1701..0000000000
--- a/apps/tdspeed.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/***************************************************************************
- * __________ __ ___.
- * Open \______ \ ____ ____ | | _\_ |__ _______ ___
- * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
- * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
- * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
- * \/ \/ \/ \/ \/
- * $Id$
- *
- * Copyright (C) 2006 by Nicolas Pitre <nico@cam.org>
- * Copyright (C) 2006-2007 by Stphane Doyon <s.doyon@videotron.ca>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
- * KIND, either express or implied.
- *
- ****************************************************************************/
-
-#ifndef _TDSPEED_H
-#define _TDSPEED_H
-
-#include "dsp.h"
-
-#define TDSPEED_OUTBUFSIZE 4096
-
-/* some #define functions to get the pitch, stretch and speed values based on */
-/* two known values. Remember that params are alphabetical. */
-#define GET_SPEED(pitch, stretch) \
- ((pitch * stretch + PITCH_SPEED_100 / 2L) / PITCH_SPEED_100)
-#define GET_PITCH(speed, stretch) \
- ((speed * PITCH_SPEED_100 + stretch / 2L) / stretch)
-#define GET_STRETCH(pitch, speed) \
- ((speed * PITCH_SPEED_100 + pitch / 2L) / pitch)
-
-void tdspeed_init(void);
-void tdspeed_finish(void);
-bool tdspeed_config(int samplerate, bool stereo, int32_t factor);
-long tdspeed_est_output_size(void);
-long tdspeed_est_input_size(long size);
-int tdspeed_doit(int32_t *src[], int count);
-
-#define STRETCH_MAX (250L * PITCH_SPEED_PRECISION) /* 250% */
-#define STRETCH_MIN (35L * PITCH_SPEED_PRECISION) /* 35% */
-
-#endif