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authorRani Hod <raenye@gmail.com>2006-09-25 16:35:03 +0000
committerRani Hod <raenye@gmail.com>2006-09-25 16:35:03 +0000
commitd5429307d519bd4a44e021e940c38f2d43fdd130 (patch)
treefb2823bd9351e31d18efbdfeb05464c5df6b88b0 /apps/codecs/adx.c
parenta875ed5d57ce33663fbf49331d2992d2e4f296d5 (diff)
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oops, forgot to 'cvs add' it [thanks for reminding me, linuxstb]
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11047 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps/codecs/adx.c')
-rw-r--r--apps/codecs/adx.c343
1 files changed, 343 insertions, 0 deletions
diff --git a/apps/codecs/adx.c b/apps/codecs/adx.c
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+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ *
+ * Copyright (C) 2006 Adam Gashlin (hcs)
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codeclib.h"
+#include "inttypes.h"
+
+CODEC_HEADER
+
+struct codec_api *rb;
+
+/* Maximum number of bytes to process in one iteration */
+#define WAV_CHUNK_SIZE (1024*2)
+
+/* Volume for ADX decoder */
+#define BASE_VOL 0x2000
+
+/* Number of times to loop looped tracks when repeat is disabled */
+#define LOOP_TIMES 2
+
+/* Length of fade-out for looped tracks (milliseconds) */
+#define FADE_LENGTH 10000L
+
+static int16_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api *api)
+{
+ struct codec_api *ci;
+ int channels;
+ int sampleswritten, i;
+ uint8_t *buf;
+ int32_t ch1_1, ch1_2, ch2_1, ch2_2; /* ADPCM history */
+ size_t n, bufsize;
+ int endofstream; /* end of stream flag */
+ uint32_t avgbytespersec;
+ int looping; /* looping flag */
+ int loop_count; /* number of loops done so far */
+ int fade_count; /* countdown for fadeout */
+ int fade_frames; /* length of fade in frames */
+ off_t start_adr, end_adr; /* loop points */
+ off_t chanstart, bufoff;
+
+ /* Generic codec initialisation */
+ rb = api;
+ ci = api;
+
+ /* we only render 16 bits */
+ ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)16);
+ /*ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));*/
+ ci->configure(DSP_DITHER, (bool *)false);
+
+next_track:
+ DEBUGF("ADX: next_track\n");
+ if (codec_init(api)) {
+ return CODEC_ERROR;
+ }
+ DEBUGF("ADX: after init\n");
+
+ /* init history */
+ ch1_1=ch1_2=ch2_1=ch2_2=0;
+
+ /* wait for track info to load */
+ while (!*ci->taginfo_ready && !ci->stop_codec)
+ ci->sleep(1);
+
+ /* Read the entire file (or as much as possible) */
+ DEBUGF("ADX: request initial buffer\n");
+ ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(ci->filesize));
+ ci->seek_buffer(0);
+ buf = ci->request_buffer(&n, ci->filesize);
+ if (!buf || n < 0x38) {
+ return CODEC_ERROR;
+ }
+ bufsize = n;
+ bufoff = 0;
+ DEBUGF("ADX: read size = %x\n",bufsize);
+
+ /* Get file header for starting offset, channel count */
+
+ chanstart = ((buf[2] << 8) | buf[3]) + 4;
+ channels = buf[7];
+
+ /* useful for seeking and reporting current playback position */
+ avgbytespersec = ci->id3->frequency * 18 * channels / 32;
+ DEBUGF("avgbytespersec=%d\n",avgbytespersec);
+
+ /* Get loop data */
+
+ looping = 0; start_adr = 0; end_adr = 0;
+ if (!memcmp(buf+0x10,"\x01\xF4\x03\x00",4)) {
+ /* Soul Calibur 2 style (type 03) */
+ DEBUGF("ADX: type 03 found\n");
+ /* check if header is too small for loop data */
+ if (chanstart-6 < 0x2c) looping=0;
+ else {
+ looping = (buf[0x18]) ||
+ (buf[0x19]) ||
+ (buf[0x1a]) ||
+ (buf[0x1b]);
+ end_adr = (buf[0x28]<<24) |
+ (buf[0x29]<<16) |
+ (buf[0x2a]<<8) |
+ (buf[0x2b]);
+
+ start_adr = (
+ (buf[0x1c]<<24) |
+ (buf[0x1d]<<16) |
+ (buf[0x1e]<<8) |
+ (buf[0x1f])
+ )/32*channels*18+chanstart;
+ }
+ } else if (!memcmp(buf+0x10,"\x01\xF4\x04\x00",4)) {
+ /* Standard (type 04) */
+ DEBUGF("ADX: type 04 found\n");
+ /* check if header is too small for loop data */
+ if (chanstart-6 < 0x38) looping=0;
+ else {
+ looping = (buf[0x24]) ||
+ (buf[0x25]) ||
+ (buf[0x26]) ||
+ (buf[0x27]);
+ end_adr = (buf[0x34]<<24) |
+ (buf[0x35]<<16) |
+ (buf[0x36]<<8) |
+ buf[0x37];
+ start_adr = (
+ (buf[0x28]<<24) |
+ (buf[0x29]<<16) |
+ (buf[0x2a]<<8) |
+ (buf[0x2b])
+ )/32*channels*18+chanstart;
+ }
+ } else {
+ DEBUGF("ADX: error, couldn't determine ADX type\n");
+ return CODEC_ERROR;
+ }
+
+ if (looping) {
+ DEBUGF("ADX: looped, start: %x end: %x\n",start_adr,end_adr);
+ } else {
+ DEBUGF("ADX: not looped\n");
+ }
+
+ /* advance to first frame */
+ /*ci->seek_buffer(chanstart);*/
+ DEBUGF("ADX: first frame at %x\n",chanstart);
+ bufoff = chanstart;
+
+ /* setup pcm buffer format */
+ ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
+ if (channels == 2) {
+ ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
+ } else if (channels == 1) {
+ ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_MONO);
+ } else {
+ DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
+ return CODEC_ERROR;
+ }
+
+ endofstream = 0;
+ loop_count = 0;
+ fade_count = -1; /* disable fade */
+ fade_frames = 1;
+
+ /* The main decoder loop */
+
+ while (!endofstream) {
+ ci->yield();
+ if (ci->stop_codec || ci->new_track) {
+ break;
+ }
+
+ /* do we need to loop? */
+ if (bufoff >= end_adr-18*channels && looping) {
+ DEBUGF("ADX: loop!\n");
+ /* check for endless looping */
+ if (ci->global_settings->repeat_mode==REPEAT_ONE) {
+ loop_count=0;
+ fade_count = -1; /* disable fade */
+ } else {
+ /* otherwise start fade after LOOP_TIMES loops */
+ loop_count++;
+ if (loop_count >= LOOP_TIMES && fade_count < 0) {
+ /* frames to fade over */
+ fade_frames = FADE_LENGTH*ci->id3->frequency/32/1000;
+ /* volume relative to fade_frames */
+ fade_count = fade_frames;
+ DEBUGF("ADX: fade_frames = %d\n",fade_frames);
+ }
+ }
+ bufoff = start_adr;
+ }
+
+ /* do we need to seek? */
+ if (ci->seek_time) {
+ uint32_t newpos;
+
+ DEBUGF("ADX: seek to %dms\n",ci->seek_time);
+
+ endofstream = 0;
+ loop_count = 0;
+ fade_count = -1; /* disable fade */
+ fade_frames = 1;
+
+ newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
+ / (1000LL*18*channels))*(18*channels);
+ bufoff = chanstart + newpos;
+ while (bufoff > end_adr-18*channels) {
+ bufoff-=end_adr-start_adr;
+ loop_count++;
+ }
+ ci->seek_complete();
+ }
+
+ if (bufoff>ci->filesize-channels*18) break; /* End of stream */
+
+ /* dance with the devil in the pale moonlight */
+ if ((bufoff > ci->curpos + (off_t)bufsize - channels*18) ||
+ bufoff < ci->curpos) {
+ DEBUGF("ADX: requesting another buffer at %x size %x\n",
+ bufoff,ci->filesize-bufoff);
+ ci->seek_buffer(bufoff);
+ buf = ci->request_buffer(&n, ci->filesize-bufoff);
+ bufsize = n;
+ DEBUGF("ADX: read size = %x\n",bufsize);
+ if ((off_t)bufsize < channels*18) {
+ /* if we can't get a full frame, just request a single
+ frame (should be able to fit it in the guard buffer) */
+ DEBUGF("ADX: requesting single frame at %x\n",bufoff);
+ buf = ci->request_buffer(&n, channels*18);
+ bufsize=n;
+ DEBUGF("ADX: read size = %x\n",bufsize);
+ }
+ if (!buf) {
+ DEBUGF("ADX: couldn't get buffer at %x size %x\n",
+ bufoff,ci->filesize-bufoff);
+ return CODEC_ERROR;
+ }
+ buf-=bufoff;
+ }
+
+ if (bufsize == 0) break; /* End of stream */
+
+ sampleswritten=0;
+
+ while (
+ /* Is there data in the file buffer? */
+ ((size_t)bufoff <= ci->curpos+bufsize-(18*channels)) &&
+ /* Is there space in the output buffer? */
+ (sampleswritten <= WAV_CHUNK_SIZE-(32*channels)) &&
+ /* Should we be looping? */
+ ((!looping) || bufoff < end_adr-18*channels)) {
+ /* decode 18 bytes to 32 samples (from bero) */
+ int32_t scale = ((buf[bufoff] << 8) | (buf[bufoff+1])) * BASE_VOL;
+
+ int32_t ch1_0, d;
+
+ for (i = 2; i < 18; i++)
+ {
+ d = (buf[bufoff+i] >> 4) & 15;
+ if (d & 8) d-= 16;
+ ch1_0 = (d*scale + 0x7298L*ch1_1 - 0x3350L*ch1_2) >> 14;
+ if (ch1_0 > 32767) ch1_0 = 32767;
+ else if (ch1_0 < -32768) ch1_0 = -32768;
+ samples[sampleswritten] = ch1_0;
+ sampleswritten+=channels;
+ ch1_2 = ch1_1; ch1_1 = ch1_0;
+ d = buf[bufoff+i] & 15;
+ if (d & 8) d -= 16;
+ ch1_0 = (d*scale + 0x7298L*ch1_1 - 0x3350L*ch1_2) >> 14;
+ if (ch1_0 > 32767) ch1_0 = 32767;
+ else if (ch1_0 < -32768) ch1_0 = -32768;
+ samples[sampleswritten] = ch1_0;
+ sampleswritten+=channels;
+ ch1_2 = ch1_1; ch1_1 = ch1_0;
+ }
+ bufoff+=18;
+
+ if (channels == 2) {
+ int32_t scale = ((buf[bufoff] << 8)|(buf[bufoff+1]))*BASE_VOL;
+
+ int32_t ch2_0, d;
+ sampleswritten-=63;
+
+ for (i = 2; i < 18; i++)
+ {
+ d = (buf[bufoff+i] >> 4) & 15;
+ if (d & 8) d-= 16;
+ ch2_0 = (d*scale + 0x7298L*ch2_1 - 0x3350L*ch2_2) >> 14;
+ if (ch2_0 > 32767) ch2_0 = 32767;
+ else if (ch2_0 < -32768) ch2_0 = -32768;
+ samples[sampleswritten] = ch2_0;
+ sampleswritten+=2;
+ ch2_2 = ch2_1; ch2_1 = ch2_0;
+ d = buf[bufoff+i] & 15;
+ if (d & 8) d -= 16;
+ ch2_0 = (d*scale + 0x7298L*ch2_1 - 0x3350L*ch2_2) >> 14;
+ if (ch2_0 > 32767) ch2_0 = 32767;
+ else if (ch2_0 < -32768) ch2_0 = -32768;
+ samples[sampleswritten] = ch2_0;
+ sampleswritten+=2;
+ ch2_2 = ch2_1; ch2_1 = ch2_0;
+ }
+ bufoff+=18;
+ sampleswritten--; /* go back to first channel's next sample */
+ }
+ if (fade_count>0) {
+ fade_count--;
+ for (i=0;i<(channels==1?32:64);i++) samples[sampleswritten-i-1]=
+ ((int32_t)samples[sampleswritten-i-1])*fade_count/fade_frames;
+ if (fade_count==0) {endofstream=1; break;}
+ }
+ }
+
+ /* 2 bytes per sample */
+ while (!ci->pcmbuf_insert((char *)samples, sampleswritten*2))
+ ci->yield();
+
+ ci->set_elapsed(
+ ((end_adr-start_adr)*loop_count + bufoff-chanstart)*
+ 1000LL/avgbytespersec);
+ }
+
+ if (ci->request_next_track())
+ goto next_track;
+
+ return CODEC_OK;
+}