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-rw-r--r--apps/enc_config.c432
-rw-r--r--apps/enc_config.h73
-rw-r--r--firmware/enc_base.c46
-rw-r--r--firmware/export/enc_base.h270
-rw-r--r--firmware/export/general.h38
-rw-r--r--firmware/export/pcm_sampr.h310
-rw-r--r--firmware/general.c77
-rw-r--r--firmware/pcm_sampr.c76
-rw-r--r--firmware/target/coldfire/pcm-coldfire.c738
9 files changed, 2060 insertions, 0 deletions
diff --git a/apps/enc_config.c b/apps/enc_config.c
new file mode 100644
index 0000000000..384e679c42
--- /dev/null
+++ b/apps/enc_config.c
@@ -0,0 +1,432 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Michael Sevakis
+ *
+ * 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 <stdio.h>
+#include <sprintf.h>
+#include <string.h>
+#include "config.h"
+#include "atoi.h"
+#include "lang.h"
+#include "misc.h"
+#include "talk.h"
+#include "general.h"
+#include "codecs.h"
+#include "menu.h"
+#include "statusbar.h"
+#include "settings.h"
+#include "audio.h"
+#include "pcm_record.h"
+#include "enc_config.h"
+
+#define MENU_ITEM_FN(fn) \
+ ((bool (*)(void))fn)
+
+#define ENC_MENU_ITEM_FN(fn) \
+ ((bool (*)(struct encoder_config *))fn)
+
+#define CALL_FN_(fn, ...) \
+ if (fn) fn(__VA_ARGS__)
+
+static bool enc_run_menu(int m, const struct menu_item items[],
+ struct encoder_config *cfg);
+static bool enc_no_config_menu(struct encoder_config *cfg);
+
+/** Function definitions for each codec - add these to enc_data
+ list following the definitions **/
+
+/** mp3_enc.codec **/
+/* mp3_enc: return encoder capabilities */
+static void mp3_enc_get_caps(const struct encoder_config *cfg,
+ struct encoder_caps *caps,
+ bool for_config)
+{
+ int i;
+ unsigned long bitr;
+
+ if (!for_config)
+ {
+ /* Overall encoder capabilities */
+ caps->samplerate_caps = MPEG1_SAMPR_CAPS | MPEG2_SAMPR_CAPS;
+ caps->channel_caps = CHN_CAP_ALL;
+ return;
+ }
+
+ /* Restrict caps based on config */
+ i = round_value_to_list32(cfg->mp3_enc.bitrate, mp3_enc_bitr,
+ MP3_ENC_NUM_BITR, false);
+ bitr = mp3_enc_bitr[i];
+
+ /* sample rate caps */
+
+ /* check if MPEG1 sample rates are available */
+ if ((bitr >= 32 && bitr <= 128) || bitr >= 160)
+ caps->samplerate_caps |= MPEG1_SAMPR_CAPS;
+
+ /* check if MPEG2 sample rates and mono are available */
+ if (bitr <= 160)
+ {
+ caps->samplerate_caps |= MPEG2_SAMPR_CAPS;
+ caps->channel_caps |= CHN_CAP_MONO;
+ }
+
+ /* check if stereo is available */
+ if (bitr >= 32)
+ caps->channel_caps |= CHN_CAP_STEREO;
+} /* mp3_enc_get_caps */
+
+/* mp3_enc: return the default configuration */
+static void mp3_enc_default_config(struct encoder_config *cfg)
+{
+ cfg->mp3_enc.bitrate = 128; /* default that works for all types */
+} /* mp3_enc_default_config */
+
+static void mp3_enc_convert_config(struct encoder_config *cfg,
+ bool to_encoder)
+{
+ if (to_encoder)
+ {
+ if ((unsigned)global_settings.mp3_enc_config.bitrate > MP3_ENC_NUM_BITR)
+ global_settings.mp3_enc_config.bitrate = MP3_ENC_BITRATE_CFG_DEFAULT;
+ cfg->mp3_enc.bitrate = mp3_enc_bitr[global_settings.mp3_enc_config.bitrate];
+ }
+ else
+ {
+ global_settings.mp3_enc_config.bitrate =
+ round_value_to_list32(cfg->mp3_enc.bitrate, mp3_enc_bitr,
+ MP3_ENC_NUM_BITR, false);
+ }
+} /* mp3_enc_convert_config */
+
+/* mp3_enc: show the bitrate setting options */
+static bool mp3_enc_bitrate(struct encoder_config *cfg)
+{
+ static const struct opt_items items[] =
+ {
+ /* Available in MPEG Version: */
+#ifdef HAVE_MPEG2_SAMPR
+#if 0
+ /* this sounds awful no matter what */
+ { "8 kBit/s", TALK_ID(8, UNIT_KBIT) }, /* 2 */
+#endif
+ /* mono only */
+ { "16 kBit/s", TALK_ID(16, UNIT_KBIT) }, /* 2 */
+ { "24 kBit/s", TALK_ID(24, UNIT_KBIT) }, /* 2 */
+#endif
+ /* stereo/mono */
+ { "32 kBit/s", TALK_ID(32, UNIT_KBIT) }, /* 1,2 */
+ { "40 kBit/s", TALK_ID(40, UNIT_KBIT) }, /* 1,2 */
+ { "48 kBit/s", TALK_ID(48, UNIT_KBIT) }, /* 1,2 */
+ { "56 kBit/s", TALK_ID(56, UNIT_KBIT) }, /* 1,2 */
+ { "64 kBit/s", TALK_ID(64, UNIT_KBIT) }, /* 1,2 */
+ { "80 kBit/s", TALK_ID(80, UNIT_KBIT) }, /* 1,2 */
+ { "96 kBit/s", TALK_ID(96, UNIT_KBIT) }, /* 1,2 */
+ { "112 kBit/s", TALK_ID(112, UNIT_KBIT) }, /* 1,2 */
+ { "128 kBit/s", TALK_ID(128, UNIT_KBIT) }, /* 1,2 */
+#if 0
+ /* oddball MPEG2-only rate stuck in the middle */
+ { "144 kBit/s", TALK_ID(144, UNIT_KBIT) }, /* 2 */
+#endif
+ { "160 kBit/s", TALK_ID(160, UNIT_KBIT) }, /* 1,2 */
+ /* stereo only */
+ { "192 kBit/s", TALK_ID(192, UNIT_KBIT) }, /* 1 */
+ { "224 kBit/s", TALK_ID(224, UNIT_KBIT) }, /* 1 */
+ { "256 kBit/s", TALK_ID(256, UNIT_KBIT) }, /* 1 */
+ { "320 kBit/s", TALK_ID(320, UNIT_KBIT) }, /* 1 */
+ };
+
+ unsigned long rate_list[ARRAYLEN(items)];
+
+ /* This is rather constant based upon the build but better than
+ storing and maintaining yet another list of numbers */
+ int n_rates = make_list_from_caps32(
+ MPEG1_BITR_CAPS | MPEG2_BITR_CAPS, mp3_enc_bitr,
+ MPEG1_BITR_CAPS
+#ifdef HAVE_MPEG2_SAMPR
+ | (MPEG2_BITR_CAPS & ~(MP3_BITR_CAP_144 | MP3_BITR_CAP_8)),
+#endif
+ rate_list);
+
+ int index = round_value_to_list32(cfg->mp3_enc.bitrate, rate_list,
+ n_rates, false);
+ bool res = set_option(str(LANG_BITRATE), &index, INT,
+ items, n_rates, NULL);
+ index = round_value_to_list32(rate_list[index], mp3_enc_bitr,
+ MP3_ENC_NUM_BITR, false);
+ cfg->mp3_enc.bitrate = mp3_enc_bitr[index];
+
+ return res;
+} /* mp3_enc_bitrate */
+
+/* mp3_enc: show the configuration menu */
+static bool mp3_enc_menu(struct encoder_config *cfg)
+{
+ static const struct menu_item items[] =
+ {
+ { ID2P(LANG_BITRATE), MENU_ITEM_FN(mp3_enc_bitrate) }
+ };
+
+ bool result;
+ int m = menu_init(items, ARRAYLEN(items), NULL, NULL, NULL, NULL);
+ result = enc_run_menu(m, items, cfg);
+ menu_exit(m);
+ return result;
+} /* mp3_enc_menu */
+
+/** wav_enc.codec **/
+/* wav_enc: show the configuration menu */
+#if 0
+static bool wav_enc_menu(struct encoder_config *cfg);
+#endif
+
+/** wavpack_enc.codec **/
+/* wavpack_enc: show the configuration menu */
+#if 0
+static bool wavpack_enc_menu(struct encoder_config *cfg);
+#endif
+
+/** config function pointers and/or data for each codec **/
+static const struct encoder_data
+{
+ void (*get_caps)(const struct encoder_config *, struct encoder_caps *,
+ bool);
+ void (*default_cfg)(struct encoder_config *);
+ void (*convert_cfg)(struct encoder_config *, bool to_encoder);
+ bool (*menu)(struct encoder_config *);
+} enc_data[REC_NUM_FORMATS] =
+{
+ /* mp3_enc.codec */
+ [REC_FORMAT_MPA_L3] = {
+ mp3_enc_get_caps,
+ mp3_enc_default_config,
+ mp3_enc_convert_config,
+ mp3_enc_menu,
+ },
+ /* wav_enc.codec */
+ [REC_FORMAT_PCM_WAV] = {
+ NULL,
+ NULL,
+ NULL,
+ enc_no_config_menu,
+ },
+ /* wavpack_enc.codec */
+ [REC_FORMAT_WAVPACK] = {
+ NULL,
+ NULL,
+ NULL,
+ enc_no_config_menu,
+ },
+};
+
+static inline bool rec_format_ok(int rec_format)
+{
+ return (unsigned)rec_format < REC_NUM_FORMATS;
+}
+
+static bool enc_run_menu(int m, const struct menu_item items[],
+ struct encoder_config *cfg)
+{
+ int selected;
+ while (1)
+ {
+ switch (selected=menu_show(m))
+ {
+ case MENU_SELECTED_EXIT:
+ return false;
+
+ case MENU_ATTACHED_USB:
+ return true;
+
+ default:
+ if (items[selected].function &&
+ ENC_MENU_ITEM_FN(items[selected].function)(cfg))
+ return true;
+ gui_syncstatusbar_draw(&statusbars, true);
+ }
+ }
+} /* enc_run_menu */
+
+/* menu created when encoder has no configuration options */
+static bool enc_no_config_menu(struct encoder_config *cfg)
+{
+ static const struct menu_item items[] =
+ {
+ { ID2P(LANG_NO_SETTINGS), NULL }
+ };
+ int m;
+ bool result;
+
+ m = menu_init(items, ARRAYLEN(items), NULL, NULL, NULL, NULL);
+ result = enc_run_menu(m, items, NULL);
+ menu_exit(m);
+
+ return result;
+ (void)cfg;
+} /* enc_no_config_menu */
+
+/* update settings dependent upon encoder settings */
+static void enc_rec_settings_changed(struct encoder_config *cfg)
+{
+ struct encoder_config enc_config;
+ struct encoder_caps caps;
+ long table[MAX(CHN_NUM_MODES, REC_NUM_FREQ)];
+ int n;
+
+ if (cfg == NULL)
+ {
+ cfg = &enc_config;
+ cfg->rec_format = global_settings.rec_format;
+ global_to_encoder_config(cfg);
+ }
+
+ /* have to sync other settings when encoder settings change */
+ if (!enc_get_caps(cfg, &caps, true))
+ return;
+
+ /* rec_channels */
+ n = make_list_from_caps32(CHN_CAP_ALL, NULL,
+ caps.channel_caps, table);
+
+ /* no zero check needed: encoder must support at least one
+ sample rate that recording supports or it shouldn't be in
+ available in the recording options */
+ n = round_value_to_list32(global_settings.rec_channels,
+ table, n, true);
+ global_settings.rec_channels = table[n];
+
+ /* rec_frequency */
+ n = make_list_from_caps32(REC_SAMPR_CAPS, rec_freq_sampr,
+ caps.samplerate_caps, table);
+
+ n = round_value_to_list32(
+ rec_freq_sampr[global_settings.rec_frequency],
+ table, n, false);
+
+ global_settings.rec_frequency = round_value_to_list32(
+ table[n], rec_freq_sampr, REC_NUM_FREQ, false);
+} /* enc_rec_settings_changed */
+
+/** public stuff **/
+void global_to_encoder_config(struct encoder_config *cfg)
+{
+ const struct encoder_data *data = &enc_data[cfg->rec_format];
+ CALL_FN_(data->convert_cfg, cfg, true);
+} /* global_to_encoder_config */
+
+void encoder_config_to_global(const struct encoder_config *cfg)
+{
+ const struct encoder_data *data = &enc_data[cfg->rec_format];
+ CALL_FN_(data->convert_cfg, (struct encoder_config *)cfg, false);
+} /* encoder_config_to_global */
+
+bool enc_get_caps(const struct encoder_config *cfg,
+ struct encoder_caps *caps,
+ bool for_config)
+{
+ /* get_caps expects caps to be zeroed first */
+ memset(caps, 0, sizeof (*caps));
+
+ if (!rec_format_ok(cfg->rec_format))
+ return false;
+
+ if (enc_data[cfg->rec_format].get_caps)
+ {
+ enc_data[cfg->rec_format].get_caps(cfg, caps, for_config);
+ }
+ else
+ {
+ /* If no function provided...defaults to all */
+ caps->samplerate_caps = SAMPR_CAP_ALL;
+ caps->channel_caps = CHN_CAP_ALL;
+ }
+
+ return true;
+} /* enc_get_caps */
+
+/* Initializes the config struct with default values */
+bool enc_init_config(struct encoder_config *cfg)
+{
+ if (!rec_format_ok(cfg->rec_format))
+ return false;
+ CALL_FN_(enc_data[cfg->rec_format].default_cfg, cfg);
+ return true;
+} /* enc_init_config */
+
+/** Encoder Menus **/
+bool enc_config_menu(struct encoder_config *cfg)
+{
+ if (!rec_format_ok(cfg->rec_format))
+ return false;
+ return enc_data[cfg->rec_format].menu(cfg);
+} /* enc_config_menu */
+
+/** Global Settings **/
+
+/* Reset all codecs to defaults */
+void enc_global_settings_reset(void)
+{
+ struct encoder_config cfg;
+ cfg.rec_format = 0;
+
+ do
+ {
+ global_to_encoder_config(&cfg);
+ enc_init_config(&cfg);
+ encoder_config_to_global(&cfg);
+ if (cfg.rec_format == global_settings.rec_format)
+ enc_rec_settings_changed(&cfg);
+ }
+ while (++cfg.rec_format < REC_NUM_FORMATS);
+} /* enc_global_settings_reset */
+
+/* Apply new settings */
+void enc_global_settings_apply(void)
+{
+ struct encoder_config cfg;
+ if (!rec_format_ok(global_settings.rec_format))
+ global_settings.rec_format = REC_FORMAT_DEFAULT;
+
+ cfg.rec_format = global_settings.rec_format;
+ global_to_encoder_config(&cfg);
+ enc_rec_settings_changed(&cfg);
+ encoder_config_to_global(&cfg);
+} /* enc_global_settings_apply */
+
+/* Show an encoder's config menu based on the global_settings.
+ Modified settings are placed in global_settings.enc_config. */
+bool enc_global_config_menu(void)
+{
+ struct encoder_config cfg;
+
+ bool res;
+
+ if (!rec_format_ok(global_settings.rec_format))
+ global_settings.rec_format = REC_FORMAT_DEFAULT;
+
+ cfg.rec_format = global_settings.rec_format;
+
+ global_to_encoder_config(&cfg);
+
+ res = enc_config_menu(&cfg);
+ if (!res)
+ {
+ enc_rec_settings_changed(&cfg);
+ encoder_config_to_global(&cfg);
+ }
+
+ return res;
+} /* enc_global_config_menu */
diff --git a/apps/enc_config.h b/apps/enc_config.h
new file mode 100644
index 0000000000..53fa7638e9
--- /dev/null
+++ b/apps/enc_config.h
@@ -0,0 +1,73 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Michael Sevakis
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#ifndef ENC_CONFIG_H
+#define ENC_CONFIG_H
+
+#include "misc.h"
+#include "enc_base.h"
+
+/** Capabilities **/
+
+/* Capabilities returned by enc_get_caps that depend upon encoder settings */
+struct encoder_caps
+{
+ unsigned long samplerate_caps; /* Mask composed of SAMPR_CAP_* flags */
+ unsigned long channel_caps; /* Mask composed of CHN_CAP_* flags */
+};
+
+/* for_config:
+ * true- the capabilities returned should be contextual based upon the
+ * settings in the config structure
+ * false- the overall capabilities are being requested
+ */
+bool enc_get_caps(const struct encoder_config *cfg,
+ struct encoder_caps *caps,
+ bool for_config);
+
+/** Configuration **/
+
+/* These translate to a back between the global format and the per-
+ instance format */
+void global_to_encoder_config(struct encoder_config *cfg);
+void encoder_config_to_global(const struct encoder_config *cfg);
+
+/* Initializes the config struct with default values.
+ set afmt member before calling. */
+bool enc_init_config(struct encoder_config *cfg);
+
+/** Encoder Menus **/
+
+/* Shows an encoder's config menu given an encoder config returned by one
+ of the enc_api functions. Modified settings are not saved to disk but
+ instead are placed in the structure. Call enc_save_config to commit
+ the data. */
+bool enc_config_menu(struct encoder_config *cfg);
+
+/** Global Settings **/
+
+/* Reset all codecs to defaults */
+void enc_global_settings_reset(void);
+
+/* Apply new settings */
+void enc_global_settings_apply(void);
+
+/* Show an encoder's config menu based on the global_settings.
+ Modified settings are placed in global_settings.enc_config. */
+bool enc_global_config_menu(void);
+#endif /* ENC_CONFIG_H */
diff --git a/firmware/enc_base.c b/firmware/enc_base.c
new file mode 100644
index 0000000000..e346064fe2
--- /dev/null
+++ b/firmware/enc_base.c
@@ -0,0 +1,46 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Michael Sevakis
+ *
+ * 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 "config.h"
+#include "audio.h"
+
+/** mp3_enc.codec **/
+
+/* These are in descending order rather than in MPEG frequency index
+ order */
+const unsigned long mp3_enc_sampr[MP3_ENC_NUM_SAMPR] =
+{
+ 48000, 44100, 32000, /* MPEG 1 */
+ 24000, 22050, 16000, /* MPEG 2 */
+#if 0
+ 12000, 11025, 8000, /* MPEG 2.5 */
+#endif
+};
+
+/* All bitrates used in the MPA L3 standard */
+const unsigned long mp3_enc_bitr[MP3_ENC_NUM_BITR] =
+{
+ 8, 16, 24, 32, 40, 48, 56, 64, 80,
+ 96, 112, 128, 144, 160, 192, 224, 256, 320
+};
+
+/** wav_enc.codec **/
+
+/** wavpack_enc.codec **/
+
+/** public functions **/
diff --git a/firmware/export/enc_base.h b/firmware/export/enc_base.h
new file mode 100644
index 0000000000..85101ac7fd
--- /dev/null
+++ b/firmware/export/enc_base.h
@@ -0,0 +1,270 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Base declarations for working with software encoders
+ *
+ * Copyright (C) 2006 Michael Sevakis
+ *
+ * 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.
+ *
+ ****************************************************************************/
+
+#ifndef ENC_BASE_H
+#define ENC_BASE_H
+
+/** encoder config structures **/
+
+/** mp3_enc.codec **/
+#define MP3_BITR_CAP_8 (1 << 0)
+#define MP3_BITR_CAP_16 (1 << 1)
+#define MP3_BITR_CAP_24 (1 << 2)
+#define MP3_BITR_CAP_32 (1 << 3)
+#define MP3_BITR_CAP_40 (1 << 4)
+#define MP3_BITR_CAP_48 (1 << 5)
+#define MP3_BITR_CAP_56 (1 << 6)
+#define MP3_BITR_CAP_64 (1 << 7)
+#define MP3_BITR_CAP_80 (1 << 8)
+#define MP3_BITR_CAP_96 (1 << 9)
+#define MP3_BITR_CAP_112 (1 << 10)
+#define MP3_BITR_CAP_128 (1 << 11)
+#define MP3_BITR_CAP_144 (1 << 12)
+#define MP3_BITR_CAP_160 (1 << 13)
+#define MP3_BITR_CAP_192 (1 << 14)
+#define MP3_BITR_CAP_224 (1 << 15)
+#define MP3_BITR_CAP_256 (1 << 16)
+#define MP3_BITR_CAP_320 (1 << 17)
+#define MP3_ENC_NUM_BITR 18
+
+/* MPEG 1 */
+#define MPEG1_SAMPR_CAPS (SAMPR_CAP_32 | SAMPR_CAP_48 | SAMPR_CAP_44)
+#define MPEG1_BITR_CAPS (MP3_BITR_CAP_32 | MP3_BITR_CAP_40 | MP3_BITR_CAP_48 | \
+ MP3_BITR_CAP_56 | MP3_BITR_CAP_64 | MP3_BITR_CAP_80 | \
+ MP3_BITR_CAP_96 | MP3_BITR_CAP_112 | MP3_BITR_CAP_128 | \
+ MP3_BITR_CAP_160 | MP3_BITR_CAP_192 | MP3_BITR_CAP_224 | \
+ MP3_BITR_CAP_256 | MP3_BITR_CAP_320)
+
+/* MPEG 2 */
+#define MPEG2_SAMPR_CAPS (SAMPR_CAP_22 | SAMPR_CAP_24 | SAMPR_CAP_16)
+#define MPEG2_BITR_CAPS (MP3_BITR_CAP_8 | MP3_BITR_CAP_16 | MP3_BITR_CAP_24 | \
+ MP3_BITR_CAP_32 | MP3_BITR_CAP_40 | MP3_BITR_CAP_48 | \
+ MP3_BITR_CAP_56 | MP3_BITR_CAP_64 | MP3_BITR_CAP_80 | \
+ MP3_BITR_CAP_96 | MP3_BITR_CAP_112 | MP3_BITR_CAP_128 | \
+ MP3_BITR_CAP_144 | MP3_BITR_CAP_160)
+
+#if 0
+/* MPEG 2.5 */
+#define MPEG2_5_SAMPR_CAPS (SAMPR_CAP_8 | SAMPR_CAP_12 | SAMPR_CAP_11)
+#define MPEG2_5_BITR_CAPS MPEG2_BITR_CAPS
+#endif
+
+/* Assume 44100 is always available and therefore MPEG1 */
+
+/* HAVE_MPEG* defines mainly apply to the bitrate menu */
+#if (REC_SAMPR_CAPS & MPEG2_SAMPR_CAPS) || defined (HAVE_SPDIF_IN)
+#define HAVE_MPEG2_SAMPR
+#endif
+
+#if 0
+#if (REC_SAMPR_CAPS & MPEG2_5_SAMPR_CAPS) || defined (HAVE_SPDIF_IN)
+#define HAVE_MPEG2_5_SAMPR
+#endif
+#endif /* 0 */
+
+#define MP3_ENC_SAMPR_CAPS (MPEG1_SAMPR_CAPS | MPEG2_SAMPR_CAPS)
+
+/* This number is count of full encoder set */
+#define MP3_ENC_NUM_SAMPR 6
+
+extern const unsigned long mp3_enc_sampr[MP3_ENC_NUM_SAMPR];
+extern const unsigned long mp3_enc_bitr[MP3_ENC_NUM_BITR];
+
+struct mp3_enc_config
+{
+ unsigned long bitrate;
+};
+
+#define MP3_ENC_BITRATE_CFG_DEFAULT 11 /* 128 */
+#define MP3_ENC_BITRATE_CFG_VALUE_LIST "8,16,24,32,40,48,56,64,80,96," \
+ "112,128,144,160,192,224,256,320"
+
+/** wav_enc.codec **/
+#define WAV_ENC_SAMPR_CAPS SAMPR_CAP_ALL
+
+struct wav_enc_config
+{
+#if 0
+ unsigned long sample_depth;
+#endif
+};
+
+/** wavpack_enc.codec **/
+#define WAVPACK_ENC_SAMPR_CAPS SAMPR_CAP_ALL
+
+struct wavpack_enc_config
+{
+#if 0
+ unsigned long sample_depth;
+#endif
+};
+
+struct encoder_config
+{
+ union
+ {
+ /* states which *_enc_config member is valid */
+ int rec_format; /* REC_FORMAT_* value */
+ int afmt; /* AFMT_* value */
+ };
+
+ union
+ {
+ struct mp3_enc_config mp3_enc;
+ struct wavpack_enc_config wavpack_enc;
+ struct wav_enc_config wav_enc;
+ };
+};
+
+/** Encoder chunk macros and definitions **/
+#define CHUNKF_START_FILE 0x0001 /* This chunk starts a new file */
+#define CHUNKF_END_FILE 0x0002 /* This chunk ends the current file */
+#define CHUNKF_PRERECORD 0x0010 /* This chunk is prerecord data,
+ a new file could start anytime */
+#define CHUNKF_ABORT 0x0020 /* Encoder should not finish this
+ chunk */
+#define CHUNKF_ERROR 0x80000000 /* An error has occured (passed to/
+ from encoder). Use the sign bit to
+ check (long)flags < 0. */
+
+/* Header at the beginning of every encoder chunk */
+struct enc_chunk_hdr
+{
+ unsigned long flags; /* in/out: flags used by encoder and file
+ writing */
+ size_t enc_size; /* out: amount of encoder data written to
+ chunk */
+ unsigned long num_pcm; /* out: number of PCM samples eaten during
+ processing
+ (<= size of allocated buffer) */
+ unsigned char *enc_data; /* out: pointer to enc_size_written bytes
+ of encoded audio data in chunk */
+ /* Encoder defined data follows header. Can be audio data + any other
+ stuff the encoder needs to handle on a per chunk basis */
+};
+
+/* Paranoia: be sure header size is 4-byte aligned */
+#define ENC_CHUNK_HDR_SIZE \
+ ALIGN_UP_P2(sizeof (struct enc_chunk_hdr), 2)
+/* Skip the chunk header and return data */
+#define ENC_CHUNK_SKIP_HDR(t, hdr) \
+ ((typeof (t))((char *)hdr + ENC_CHUNK_HDR_SIZE))
+/* Cast p to struct enc_chunk_hdr * */
+#define ENC_CHUNK_HDR(p) \
+ ((struct enc_chunk_hdr *)(p))
+
+enum enc_events
+{
+ /* File writing events - data points to enc_file_event_data */
+ ENC_START_FILE = 0, /* a new file has been opened and no data has yet
+ been written */
+ ENC_WRITE_CHUNK, /* write the current chunk to disk */
+ ENC_END_FILE, /* current file about to be closed and all valid
+ data has been written */
+ /* Encoder buffer events - data points to enc_buffer_event_data */
+ ENC_REC_NEW_STREAM, /* Take steps to finish current stream and start
+ new */
+};
+
+/**
+ * encoder can write extra data to the file such as headers or more encoded
+ * samples and must update sizes and samples accordingly.
+ */
+struct enc_file_event_data
+{
+ struct enc_chunk_hdr *chunk; /* Current chunk */
+ size_t new_enc_size; /* New size of chunk */
+ unsigned long new_num_pcm; /* New number of pcm in chunk */
+ const char *filename; /* filename to open if ENC_START_FILE */
+ int rec_file; /* Current file or < 0 if none */
+ unsigned long num_pcm_samples; /* Current pcm sample count written to
+ file so far. */
+};
+
+/**
+ * encoder may add some data to the end of the last and start of the next
+ * but must never yield when called so any encoding done should be absolutely
+ * minimal.
+ */
+struct enc_buffer_event_data
+{
+ unsigned long flags; /* in: One or more of:
+ * CHUNKF_PRERECORD
+ * CHUNKF_END_FILE
+ * CHUNKF_START_FILE
+ */
+ struct enc_chunk_hdr *pre_chunk; /* in: pointer to first prerecord
+ * chunk
+ */
+ struct enc_chunk_hdr *chunk; /* in,out: chunk were split occurs -
+ * first chunk of start
+ */
+};
+
+/** Callbacks called by encoder codec **/
+
+/* parameters passed to encoder by enc_get_inputs */
+struct enc_inputs
+{
+ unsigned long sample_rate; /* out - pcm frequency */
+ int num_channels; /* out - number of audio channels */
+ struct encoder_config *config; /* out - encoder settings */
+};
+
+void enc_get_inputs(struct enc_inputs *inputs);
+
+/* parameters pass from encoder to enc_set_parameters */
+struct enc_parameters
+{
+ /* IN parameters */
+ int afmt; /* AFMT_* id - sanity checker */
+ size_t chunk_size; /* max chunk size required */
+ unsigned long enc_sample_rate; /* actual sample rate used by encoder
+ (for recorded time calculation) */
+ size_t reserve_bytes; /* number of bytes to reserve immediately
+ following chunks */
+ void (*events_callback)(enum enc_events event,
+ void *data); /* pointer to events callback */
+ /* OUT parameters */
+ unsigned char *enc_buffer; /* pointer to enc_buffer */
+ size_t buf_chunk_size; /* size of chunks in enc_buffer */
+ int num_chunks; /* number of chunks allotted to encoder */
+ unsigned char *reserve_buffer; /* pointer to reserve_bytes bytes */
+};
+
+/* set the encoder dimensions - called by encoder codec at initialization
+ and termination */
+void enc_set_parameters(struct enc_parameters *params);
+/* returns pointer to next write chunk in circular buffer */
+struct enc_chunk_hdr * enc_get_chunk(void);
+/* releases the current chunk into the available chunks */
+void enc_finish_chunk(void);
+/* checks near empty state on pcm input buffer */
+int enc_pcm_buf_near_empty(void);
+
+#define PCM_MAX_FEED_SIZE 20000 /* max pcm size passed to encoder */
+
+/* passes a pointer to next chunk of unprocessed wav data */
+unsigned char * enc_get_pcm_data(size_t size);
+/* puts some pcm data back in the queue */
+size_t enc_unget_pcm_data(size_t size);
+
+#endif /* ENC_BASE_H */
diff --git a/firmware/export/general.h b/firmware/export/general.h
new file mode 100644
index 0000000000..427e2773b8
--- /dev/null
+++ b/firmware/export/general.h
@@ -0,0 +1,38 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 by Michael Sevakis
+ *
+ * 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.
+ *
+ ****************************************************************************/
+
+#ifndef GENERAL_H
+#define GENERAL_H
+
+#include <stdbool.h>
+
+/* round a signed/unsigned 32bit value to the closest of a list of values */
+/* returns the index of the closest value */
+int round_value_to_list32(unsigned long value,
+ const unsigned long list[],
+ int count,
+ bool signd);
+
+int make_list_from_caps32(unsigned long src_mask,
+ const unsigned long *src_list,
+ unsigned long caps_mask,
+ unsigned long *caps_list);
+
+
+#endif /* GENERAL_H */
diff --git a/firmware/export/pcm_sampr.h b/firmware/export/pcm_sampr.h
new file mode 100644
index 0000000000..c4a399b62f
--- /dev/null
+++ b/firmware/export/pcm_sampr.h
@@ -0,0 +1,310 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 by Michael Sevakis
+ *
+ * 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.
+ *
+ ****************************************************************************/
+
+#ifndef PCM_SAMPR_H
+#define PCM_SAMPR_H
+
+/* These must be macros for comparison with SAMPR_CAP_* flags by the
+ preprocessor. Add samplerate index in descending order renumbering
+ the ones later in the list if any */
+#define FREQ_96 0
+#define FREQ_88 1
+#define FREQ_64 2
+#define FREQ_48 3
+#define FREQ_44 4
+#define FREQ_32 5
+#define FREQ_24 6
+#define FREQ_22 7
+#define FREQ_16 8
+#define FREQ_12 9
+#define FREQ_11 10
+#define FREQ_8 11
+#define SAMPR_NUM_FREQ 12
+
+/* sample rate values in HZ */
+#define SAMPR_96 96000
+#define SAMPR_88 88200
+#define SAMPR_64 64000
+#define SAMPR_48 48000
+#define SAMPR_44 44100
+#define SAMPR_32 32000
+#define SAMPR_24 24000
+#define SAMPR_22 22050
+#define SAMPR_16 16000
+#define SAMPR_12 12000
+#define SAMPR_11 11025
+#define SAMPR_8 8000
+
+/* sample rate capability bits */
+#define SAMPR_CAP_96 (1 << FREQ_96)
+#define SAMPR_CAP_88 (1 << FREQ_88)
+#define SAMPR_CAP_64 (1 << FREQ_64)
+#define SAMPR_CAP_48 (1 << FREQ_48)
+#define SAMPR_CAP_44 (1 << FREQ_44)
+#define SAMPR_CAP_32 (1 << FREQ_32)
+#define SAMPR_CAP_24 (1 << FREQ_24)
+#define SAMPR_CAP_22 (1 << FREQ_22)
+#define SAMPR_CAP_16 (1 << FREQ_16)
+#define SAMPR_CAP_12 (1 << FREQ_12)
+#define SAMPR_CAP_11 (1 << FREQ_11)
+#define SAMPR_CAP_8 (1 << FREQ_8)
+#define SAMPR_CAP_ALL (SAMPR_CAP_96 | SAMPR_CAP_88 | SAMPR_CAP_64 | \
+ SAMPR_CAP_48 | SAMPR_CAP_44 | SAMPR_CAP_32 | \
+ SAMPR_CAP_24 | SAMPR_CAP_22 | SAMPR_CAP_16 | \
+ SAMPR_CAP_12 | SAMPR_CAP_11 | SAMPR_CAP_8)
+
+/* Master list of all "standard" rates supported. */
+extern const unsigned long audio_master_sampr_list[SAMPR_NUM_FREQ];
+
+/** Hardware sample rates **/
+
+/* Enumeration of supported frequencies where 0 is the highest rate
+ supported and REC_NUM_FREQUENCIES is the number available */
+enum hw_freq_indexes
+{
+ __HW_FREQ_START_INDEX = -1, /* Make sure first in list is 0 */
+
+/* 96000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_96) /* Macros and enums for each FREQ: */
+ HW_FREQ_96, /* Index in enumeration */
+#define HW_HAVE_96 /* Defined if this FREQ is defined */
+#define HW_HAVE_96_(...) __VA_ARGS__ /* Output its parameters for this FREQ */
+#else
+#define HW_HAVE_96_(...) /* Discards its parameters for this FREQ */
+#endif
+/* 88200 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_88)
+ HW_FREQ_88,
+#define HW_HAVE_88
+#define HW_HAVE_88_(...) __VA_ARGS__
+#else
+#define HW_HAVE_88_(...)
+#endif
+/* 64000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_64)
+ HW_FREQ_64,
+#define HW_HAVE_64
+#define HW_HAVE_64_(...) __VA_ARGS__
+#else
+#define HW_HAVE_64_(...)
+#endif
+/* 48000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_48)
+ HW_FREQ_48,
+#define HW_HAVE_48
+#define HW_HAVE_48_(...) __VA_ARGS__
+#else
+#define HW_HAVE_48_(...)
+#endif
+/* 44100 */
+ HW_FREQ_44,
+#define HW_HAVE_44
+#define HW_HAVE_44_(...) __VA_ARGS__
+/* 32000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_32)
+ HW_FREQ_32,
+#define HW_HAVE_32
+#define HW_HAVE_32_(...) __VA_ARGS__
+#else
+#define HW_HAVE_32_(...)
+#endif
+/* 24000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_24)
+ HW_FREQ_24,
+#define HW_HAVE_24
+#define HW_HAVE_24_(...) __VA_ARGS__
+#else
+#define HW_HAVE_24_(...)
+#endif
+/* 22050 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_22)
+ HW_FREQ_22,
+#define HW_HAVE_22
+#define HW_HAVE_22_(...) __VA_ARGS__
+#else
+#define HW_HAVE_22_(...)
+#endif
+/* 16000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_16)
+ HW_FREQ_16,
+#define HW_HAVE_16
+#define HW_HAVE_16_(...) __VA_ARGS__
+#else
+#define HW_HAVE_16_(...)
+#endif
+/* 12000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_12)
+ HW_FREQ_12,
+#define HW_HAVE_12
+#define HW_HAVE_12_(...) __VA_ARGS__
+#else
+#define HW_HAVE_12_(...)
+#endif
+/* 11025 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_11)
+ HW_FREQ_11,
+#define HW_HAVE_11
+#define HW_HAVE_11_(...) __VA_ARGS__
+#else
+#define HW_HAVE_11_(...)
+#endif
+/* 8000 */
+#if (HW_SAMPR_CAPS & SAMPR_CAP_8 )
+ HW_FREQ_8,
+#define HW_HAVE_8
+#define HW_HAVE_8_(...) __VA_ARGS__
+#else
+#define HW_HAVE_8_(...)
+#endif
+ HW_NUM_FREQ,
+ HW_FREQ_DEFAULT = HW_FREQ_44,
+ HW_SAMPR_DEFAULT = SAMPR_44,
+}; /* enum hw_freq_indexes */
+
+/* list of hardware sample rates */
+extern const unsigned long hw_freq_sampr[HW_NUM_FREQ];
+
+#ifdef HAVE_RECORDING
+/* Enumeration of supported frequencies where 0 is the highest rate
+ supported and REC_NUM_FREQUENCIES is the number available */
+enum rec_freq_indexes
+{
+ __REC_FREQ_START_INDEX = -1, /* Make sure first in list is 0 */
+
+/* 96000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_96) /* Macros and enums for each FREQ: */
+ REC_FREQ_96, /* Index in enumeration */
+#define REC_HAVE_96 /* Defined if this FREQ is defined */
+#define REC_HAVE_96_(...) __VA_ARGS__ /* Output its parameters for this FREQ */
+#else
+#define REC_HAVE_96_(...) /* Discards its parameters for this FREQ */
+#endif
+/* 88200 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_88)
+ REC_FREQ_88,
+#define REC_HAVE_88
+#define REC_HAVE_88_(...) __VA_ARGS__
+#else
+#define REC_HAVE_88_(...)
+#endif
+/* 64000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_64)
+ REC_FREQ_64,
+#define REC_HAVE_64
+#define REC_HAVE_64_(...) __VA_ARGS__
+#else
+#define REC_HAVE_64_(...)
+#endif
+/* 48000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_48)
+ REC_FREQ_48,
+#define REC_HAVE_48
+#define REC_HAVE_48_(...) __VA_ARGS__
+#else
+#define REC_HAVE_48_(...)
+#endif
+/* 44100 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_44)
+ REC_FREQ_44,
+#define REC_HAVE_44
+#define REC_HAVE_44_(...) __VA_ARGS__
+#else
+#define REC_HAVE_44_(...)
+#endif
+/* 32000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_32)
+ REC_FREQ_32,
+#define REC_HAVE_32
+#define REC_HAVE_32_(...) __VA_ARGS__
+#else
+#define REC_HAVE_32_(...)
+#endif
+/* 24000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_24)
+ REC_FREQ_24,
+#define REC_HAVE_24
+#define REC_HAVE_24_(...) __VA_ARGS__
+#else
+#define REC_HAVE_24_(...)
+#endif
+/* 22050 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_22)
+ REC_FREQ_22,
+#define REC_HAVE_22
+#define REC_HAVE_22_(...) __VA_ARGS__
+#else
+#define REC_HAVE_22_(...)
+#endif
+/* 16000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_16)
+ REC_FREQ_16,
+#define REC_HAVE_16
+#define REC_HAVE_16_(...) __VA_ARGS__
+#else
+#define REC_HAVE_16_(...)
+#endif
+/* 12000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_12)
+ REC_FREQ_12,
+#define REC_HAVE_12
+#define REC_HAVE_12_(...) __VA_ARGS__
+#else
+#define REC_HAVE_12_(...)
+#endif
+/* 11025 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_11)
+ REC_FREQ_11,
+#define REC_HAVE_11
+#define REC_HAVE_11_(...) __VA_ARGS__
+#else
+#define REC_HAVE_11_(...)
+#endif
+/* 8000 */
+#if (REC_SAMPR_CAPS & SAMPR_CAP_8 )
+ REC_FREQ_8,
+#define REC_HAVE_8
+#define REC_HAVE_8_(...) __VA_ARGS__
+#else
+#define REC_HAVE_8_(...)
+#endif
+ REC_NUM_FREQ,
+ /* This should always come out I reckon */
+ REC_FREQ_DEFAULT = REC_FREQ_44,
+ /* Get the minimum bitcount needed to save the range of values */
+ REC_FREQ_CFG_NUM_BITS = (REC_NUM_FREQ > 8 ?
+ 4 : (REC_NUM_FREQ > 4 ?
+ 3 : (REC_NUM_FREQ > 2 ?
+ 2 : 1
+ )
+ )
+ ),
+}; /* enum rec_freq_indexes */
+
+#define REC_FREQ_CFG_VAL_LIST &REC_HAVE_96_(",96") REC_HAVE_88_(",88") \
+ REC_HAVE_64_(",64") REC_HAVE_48_(",48") \
+ REC_HAVE_44_(",44") REC_HAVE_32_(",32") \
+ REC_HAVE_24_(",24") REC_HAVE_22_(",22") \
+ REC_HAVE_16_(",16") REC_HAVE_12_(",12") \
+ REC_HAVE_11_(",11") REC_HAVE_8_(",8")[1]
+
+/* List of recording supported sample rates (set or subset of master list) */
+extern const unsigned long rec_freq_sampr[REC_NUM_FREQ];
+#endif /* HAVE_RECORDING */
+
+#endif /* PCM_SAMPR_H */
diff --git a/firmware/general.c b/firmware/general.c
new file mode 100644
index 0000000000..7f4348046c
--- /dev/null
+++ b/firmware/general.c
@@ -0,0 +1,77 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 by Michael Sevakis
+ *
+ * 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 <limits.h>
+#include "config.h"
+#include "general.h"
+
+int round_value_to_list32(unsigned long value,
+ const unsigned long list[],
+ int count,
+ bool signd)
+{
+ unsigned long dmin = ULONG_MAX;
+ int idmin = -1, i;
+
+ for (i = 0; i < count; i++)
+ {
+ unsigned long diff;
+
+ if (list[i] == value)
+ {
+ idmin = i;
+ break;
+ }
+
+ if (signd ? ((long)list[i] < (long)value) : (list[i] < value))
+ diff = value - list[i];
+ else
+ diff = list[i] - value;
+
+ if (diff < dmin)
+ {
+ dmin = diff;
+ idmin = i;
+ }
+ }
+
+ return idmin;
+} /* round_value_to_list32 */
+
+/* Number of bits set in src_mask should equal src_list length */
+int make_list_from_caps32(unsigned long src_mask,
+ const unsigned long *src_list,
+ unsigned long caps_mask,
+ unsigned long *caps_list)
+{
+ int i, count;
+ unsigned long mask;
+
+ for (mask = src_mask, count = 0, i = 0;
+ mask != 0;
+ src_mask = mask, i++)
+ {
+ unsigned long test_bit;
+ mask &= mask - 1; /* Zero lowest bit set */
+ test_bit = mask ^ src_mask; /* Isolate the bit */
+ if (test_bit & caps_mask) /* Add item if caps has test bit set */
+ caps_list[count++] = src_list ? src_list[i] : (unsigned long)i;
+ }
+
+ return count;
+} /* make_list_from_caps32 */
diff --git a/firmware/pcm_sampr.c b/firmware/pcm_sampr.c
new file mode 100644
index 0000000000..cceb4b7399
--- /dev/null
+++ b/firmware/pcm_sampr.c
@@ -0,0 +1,76 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Michael Sevakis
+ *
+ * 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 "config.h"
+#include "pcm_sampr.h"
+
+/* Master list of all "standard" rates supported. */
+const unsigned long audio_master_sampr_list[SAMPR_NUM_FREQ] =
+{
+ [0 ... SAMPR_NUM_FREQ-1] = -1, /* any gaps set to -1 */
+ [FREQ_96] = SAMPR_96,
+ [FREQ_88] = SAMPR_88,
+ [FREQ_64] = SAMPR_64,
+ [FREQ_48] = SAMPR_48,
+ [FREQ_44] = SAMPR_44,
+ [FREQ_32] = SAMPR_32,
+ [FREQ_24] = SAMPR_24,
+ [FREQ_22] = SAMPR_22,
+ [FREQ_16] = SAMPR_16,
+ [FREQ_12] = SAMPR_12,
+ [FREQ_11] = SAMPR_11,
+ [FREQ_8 ] = SAMPR_8,
+};
+
+/* List of all hardware rates supported (set or subset of master list) */
+const unsigned long hw_freq_sampr[HW_NUM_FREQ] =
+{
+ [0 ... HW_NUM_FREQ-1] = -1,
+ HW_HAVE_96_([HW_FREQ_96] = SAMPR_96,)
+ HW_HAVE_88_([HW_FREQ_88] = SAMPR_88,)
+ HW_HAVE_64_([HW_FREQ_64] = SAMPR_64,)
+ HW_HAVE_48_([HW_FREQ_48] = SAMPR_48,)
+ HW_HAVE_44_([HW_FREQ_44] = SAMPR_44,)
+ HW_HAVE_32_([HW_FREQ_32] = SAMPR_32,)
+ HW_HAVE_24_([HW_FREQ_24] = SAMPR_24,)
+ HW_HAVE_22_([HW_FREQ_22] = SAMPR_22,)
+ HW_HAVE_16_([HW_FREQ_16] = SAMPR_16,)
+ HW_HAVE_12_([HW_FREQ_12] = SAMPR_12,)
+ HW_HAVE_11_([HW_FREQ_11] = SAMPR_11,)
+ HW_HAVE_8_( [HW_FREQ_8 ] = SAMPR_8 ,)
+};
+
+#ifdef HAVE_RECORDING
+/* List of recording supported sample rates (set or subset of master list) */
+const unsigned long rec_freq_sampr[REC_NUM_FREQ] =
+{
+ [0 ... REC_NUM_FREQ-1] = -1,
+ REC_HAVE_96_([REC_FREQ_96] = SAMPR_96,)
+ REC_HAVE_88_([REC_FREQ_88] = SAMPR_88,)
+ REC_HAVE_64_([REC_FREQ_64] = SAMPR_64,)
+ REC_HAVE_48_([REC_FREQ_48] = SAMPR_48,)
+ REC_HAVE_44_([REC_FREQ_44] = SAMPR_44,)
+ REC_HAVE_32_([REC_FREQ_32] = SAMPR_32,)
+ REC_HAVE_24_([REC_FREQ_24] = SAMPR_24,)
+ REC_HAVE_22_([REC_FREQ_22] = SAMPR_22,)
+ REC_HAVE_16_([REC_FREQ_16] = SAMPR_16,)
+ REC_HAVE_12_([REC_FREQ_12] = SAMPR_12,)
+ REC_HAVE_11_([REC_FREQ_11] = SAMPR_11,)
+ REC_HAVE_8_( [REC_FREQ_8 ] = SAMPR_8 ,)
+};
+#endif /* HAVE_RECORDING */
diff --git a/firmware/target/coldfire/pcm-coldfire.c b/firmware/target/coldfire/pcm-coldfire.c
new file mode 100644
index 0000000000..6b92f9cc14
--- /dev/null
+++ b/firmware/target/coldfire/pcm-coldfire.c
@@ -0,0 +1,738 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006 by Michael Sevakis
+ *
+ * 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 <stdlib.h>
+#include "system.h"
+#include "kernel.h"
+#include "logf.h"
+#include "audio.h"
+#if defined(HAVE_UDA1380)
+#include "uda1380.h"
+#elif defined(HAVE_TLV320)
+#include "tlv320.h"
+#endif
+
+/* Avoid further #ifdef's for some codec functions */
+#if defined(HAVE_UDA1380)
+#define ac_init uda1380_init
+#define ac_mute uda1380_mute
+#define ac_set_frequency uda1380_set_frequency
+#elif defined(HAVE_TLV320)
+#define ac_init tlv320_init
+#define ac_mute tlv320_mute
+#define ac_set_frequency tlv320_set_frequency
+#endif
+
+/** Semi-private shared symbols **/
+
+/* the registered callback function to ask for more pcm data */
+extern pcm_more_callback_type pcm_callback_for_more;
+extern bool pcm_playing;
+extern bool pcm_paused;
+
+/* the registered callback function for when more data is available */
+extern pcm_more_callback_type pcm_callback_more_ready;
+extern bool pcm_recording;
+
+/* peaks */
+static int play_peak_left, play_peak_right;
+static unsigned long *rec_peak_addr;
+static int rec_peak_left, rec_peak_right;
+
+#define IIS_DEFPARM ( (freq_ent[FPARM_CLOCKSEL] << 12) | \
+ (pcm_txsrc_select[pcm_monitor+1] << 8) | \
+ (4 << 2) ) /* 64 bit clocks / word clock */
+#define IIS_RESET 0x800
+
+#ifdef IAUDIO_X5
+#define SET_IIS_CONFIG(x) IIS1CONFIG = (x);
+#define IIS_CONFIG IIS1CONFIG
+#define PLLCR_SET_AUDIO_BITS_DEFPARM \
+ ((freq_ent[FPARM_CLSEL] << 28) | (1 << 22))
+#else
+#define SET_IIS_CONFIG(x) IIS2CONFIG = (x);
+#define IIS_CONFIG IIS2CONFIG
+#define PLLCR_SET_AUDIO_BITS_DEFPARM \
+ ((freq_ent[FPARM_CLSEL] << 28) | (3 << 22))
+
+#ifdef HAVE_SPDIF_OUT
+#define EBU_DEFPARM ((7 << 12) | (3 << 8) | (1 << 5) | (5 << 2))
+#endif
+#endif
+
+/** Sample rates **/
+#define FPARM_CLOCKSEL 0
+#define FPARM_CLSEL 1
+#define FPARM_FSEL 2
+#if CONFIG_CPU == MCF5249 && defined(HAVE_UDA1380)
+static const unsigned char pcm_freq_parms[HW_NUM_FREQ][3] =
+{
+ [HW_FREQ_88] = { 0x0c, 0x01, 0x03 },
+ [HW_FREQ_44] = { 0x06, 0x01, 0x02 },
+ [HW_FREQ_22] = { 0x04, 0x02, 0x01 },
+ [HW_FREQ_11] = { 0x02, 0x02, 0x00 },
+};
+#endif
+
+#if CONFIG_CPU == MCF5250 && defined(HAVE_TLV320)
+static const unsigned char pcm_freq_parms[HW_NUM_FREQ][3] =
+{
+ [HW_FREQ_88] = { 0x0c, 0x01, 0x02 },
+ [HW_FREQ_44] = { 0x06, 0x01, 0x01 },
+ [HW_FREQ_22] = { 0x04, 0x01, 0x00 },
+ [HW_FREQ_11] = { 0x02, 0x02, 0x00 },
+};
+#endif
+
+static int pcm_freq = HW_SAMPR_DEFAULT; /* 44.1 is default */
+static const unsigned char *freq_ent = pcm_freq_parms[HW_FREQ_DEFAULT];
+
+/* set frequency used by the audio hardware */
+void pcm_set_frequency(unsigned int frequency)
+{
+ int index;
+
+ switch(frequency)
+ {
+ case SAMPR_11:
+ index = HW_FREQ_11;
+ break;
+ case SAMPR_22:
+ index = HW_FREQ_22;
+ break;
+ default:
+ case SAMPR_44:
+ index = HW_FREQ_44;
+ break;
+ case SAMPR_88:
+ index = HW_FREQ_88;
+ break;
+ }
+
+ /* remember table entry and rate */
+ freq_ent = pcm_freq_parms[index];
+ pcm_freq = hw_freq_sampr[index];
+} /* pcm_set_frequency */
+
+/** monitoring/source selection **/
+static int pcm_monitor = AUDIO_SRC_PLAYBACK;
+
+static const unsigned char pcm_txsrc_select[AUDIO_NUM_SOURCES+1] =
+{
+ [AUDIO_SRC_PLAYBACK+1] = 3, /* PDOR3 */
+ [AUDIO_SRC_MIC+1] = 4, /* IIS1 RcvData */
+ [AUDIO_SRC_LINEIN+1] = 4, /* IIS1 RcvData */
+#ifdef HAVE_FMRADIO_IN
+ [AUDIO_SRC_FMRADIO+1] = 4, /* IIS1 RcvData */
+#endif
+#ifdef HAVE_SPDIF_IN
+ [AUDIO_SRC_SPDIF+1] = 7, /* EBU1 RcvData */
+#endif
+};
+
+static const unsigned short pcm_dataincontrol[AUDIO_NUM_SOURCES+1] =
+{
+ [AUDIO_SRC_PLAYBACK+1] = 0x0200, /* Reset PDIR2 data flow */
+ [AUDIO_SRC_MIC+1] = 0xc020, /* Int. when 6 samples in FIFO,
+ PDIR2 src = ebu1RcvData */
+ [AUDIO_SRC_LINEIN+1] = 0xc020, /* Int. when 6 samples in FIFO,
+ PDIR2 src = ebu1RcvData */
+#ifdef HAVE_FMRADIO_IN
+ [AUDIO_SRC_FMRADIO+1] = 0xc020, /* Int. when 6 samples in FIFO,
+ PDIR2 src = ebu1RcvData */
+#endif
+#ifdef HAVE_SPDIF_IN
+ [AUDIO_SRC_SPDIF+1] = 0xc038, /* Int. when 6 samples in FIFO,
+ PDIR2 src = ebu1RcvData */
+#endif
+};
+
+static int pcm_rec_src = AUDIO_SRC_PLAYBACK;
+
+void pcm_set_monitor(int monitor)
+{
+ if ((unsigned)monitor >= AUDIO_NUM_SOURCES)
+ monitor = AUDIO_SRC_PLAYBACK;
+ pcm_monitor = monitor;
+} /* pcm_set_monitor */
+
+void pcm_set_rec_source(int source)
+{
+ if ((unsigned)source >= AUDIO_NUM_SOURCES)
+ source = AUDIO_SRC_PLAYBACK;
+ pcm_rec_src = source;
+} /* pcm_set_rec_source */
+
+/* apply audio settings */
+void pcm_apply_settings(bool reset)
+{
+ static int last_pcm_freq = HW_SAMPR_DEFAULT;
+#if 0
+ static int last_pcm_monitor = AUDIO_SRC_PLAYBACK;
+#endif
+ static int last_pcm_rec_src = AUDIO_SRC_PLAYBACK;
+
+ /* Playback must prevent pops and record monitoring won't work at all
+ adding IIS_RESET when setting IIS_CONFIG. Use a different method for
+ each. */
+ if (reset && (pcm_monitor != AUDIO_SRC_PLAYBACK))
+ {
+ /* Not playback - reset first */
+ SET_IIS_CONFIG(IIS_RESET);
+ reset = false;
+ }
+
+ if (pcm_rec_src != last_pcm_rec_src)
+ {
+ last_pcm_rec_src = pcm_rec_src;
+ DATAINCONTROL = pcm_dataincontrol[pcm_rec_src+1];
+ }
+
+ if (pcm_freq != last_pcm_freq)
+ {
+ last_pcm_freq = pcm_freq;
+ ac_set_frequency(freq_ent[FPARM_FSEL]);
+ coldfire_set_pllcr_audio_bits(PLLCR_SET_AUDIO_BITS_DEFPARM);
+ }
+
+ SET_IIS_CONFIG(IIS_DEFPARM | (reset ? IIS_RESET : 0));
+} /* pcm_apply_settings */
+
+/** DMA **/
+
+/****************************************************************************
+ ** Playback DMA transfer
+ **/
+
+/* Set up the DMA transfer that kicks in when the audio FIFO gets empty */
+void pcm_play_dma_start(const void *addr, size_t size)
+{
+ logf("pcm_play_dma_start");
+
+ addr = (void *)((unsigned long)addr & ~3); /* Align data */
+ size &= ~3; /* Size must be multiple of 4 */
+
+ pcm_playing = true;
+
+ /* Reset the audio FIFO */
+#ifdef HAVE_SPDIF_OUT
+ EBU1CONFIG = IIS_RESET | EBU_DEFPARM;
+#endif
+
+ /* Set up DMA transfer */
+ SAR0 = (unsigned long)addr; /* Source address */
+ DAR0 = (unsigned long)&PDOR3; /* Destination address */
+ BCR0 = size; /* Bytes to transfer */
+
+ /* Enable the FIFO and force one write to it */
+ pcm_apply_settings(false);
+
+ /* Also send the audio to S/PDIF */
+#ifdef HAVE_SPDIF_OUT
+ EBU1CONFIG = EBU_DEFPARM;
+#endif
+
+ DCR0 = DMA_INT | DMA_EEXT | DMA_CS | DMA_AA |
+ DMA_SINC | DMA_SSIZE(3) | DMA_START;
+} /* pcm_play_dma_start */
+
+/* Stops the DMA transfer and interrupt */
+void pcm_play_dma_stop(void)
+{
+ logf("pcm_play_dma_stop");
+
+ pcm_playing = false;
+
+ DCR0 = 0;
+ DSR0 = 1;
+
+ /* Reset the FIFO */
+ pcm_apply_settings(false);
+
+#ifdef HAVE_SPDIF_OUT
+ EBU1CONFIG = IIS_RESET | EBU_DEFPARM;
+#endif
+} /* pcm_play_dma_stop */
+
+void pcm_init(void)
+{
+ logf("pcm_init");
+
+ pcm_playing = false;
+ pcm_paused = false;
+ pcm_callback_for_more = NULL;
+
+ MPARK = 0x81; /* PARK[1,0]=10 + BCR24BIT */
+ DIVR0 = 54; /* DMA0 is mapped into vector 54 in system.c */
+ DMAROUTE = (DMAROUTE & 0xffffff00) | DMA0_REQ_AUDIO_1;
+ DMACONFIG = 1; /* DMA0Req = PDOR3, DMA1Req = PDIR2 */
+
+ /* Reset the audio FIFO */
+ SET_IIS_CONFIG(IIS_RESET);
+
+ pcm_set_frequency(-1);
+ pcm_set_monitor(-1);
+
+ /* Prevent pops (resets DAC to zero point) */
+ SET_IIS_CONFIG(IIS_DEFPARM | IIS_RESET);
+
+ /* Initialize default register values. */
+ ac_init();
+
+#if defined(HAVE_UDA1380)
+ /* Sleep a while so the power can stabilize (especially a long
+ delay is needed for the line out connector). */
+ sleep(HZ);
+ /* Power on FSDAC and HP amp. */
+ uda1380_enable_output(true);
+#elif defined(HAVE_TLV320)
+ sleep(HZ/4);
+#endif
+
+ /* UDA1380: Unmute the master channel
+ (DAC should be at zero point now). */
+ ac_mute(false);
+
+ /* Call pcm_play_dma_stop to initialize everything. */
+ pcm_play_dma_stop();
+
+ /* Enable interrupt at level 7, priority 0 */
+ ICR6 = (7 << 2);
+ IMR &= ~(1 << 14); /* bit 14 is DMA0 */
+} /* pcm_init */
+
+size_t pcm_get_bytes_waiting(void)
+{
+ return BCR0 & 0xffffff;
+} /* pcm_get_bytes_waiting */
+
+/* DMA0 Interrupt is called when the DMA has finished transfering a chunk
+ from the caller's buffer */
+void DMA0(void) __attribute__ ((interrupt_handler, section(".icode")));
+void DMA0(void)
+{
+ int res = DSR0;
+
+ DSR0 = 1; /* Clear interrupt */
+ DCR0 &= ~DMA_EEXT;
+
+ /* Stop on error */
+ if ((res & 0x70) == 0)
+ {
+ pcm_more_callback_type get_more = pcm_callback_for_more;
+ unsigned char *next_start;
+ size_t next_size = 0;
+
+ if (get_more)
+ get_more(&next_start, &next_size);
+
+ if (next_size > 0)
+ {
+ SAR0 = (unsigned long)next_start; /* Source address */
+ BCR0 = next_size; /* Bytes to transfer */
+ DCR0 |= DMA_EEXT;
+ return;
+ }
+ else
+ {
+ /* Finished playing */
+#if 0
+ /* int. logfs can trash the display */
+ logf("DMA0 No Data:0x%04x", res);
+#endif
+ }
+ }
+ else
+ {
+ logf("DMA Error:0x%04x", res);
+ }
+
+ pcm_play_dma_stop();
+} /* DMA0 */
+
+/****************************************************************************
+ ** Recording DMA transfer
+ **/
+void pcm_rec_dma_start(const void *addr, size_t size)
+{
+ logf("pcm_rec_dma_start");
+
+ addr = (void *)((unsigned long)addr & ~3); /* Align data */
+ size &= ~3; /* Size must be multiple of 4 */
+
+ pcm_recording = true;
+
+ DAR1 = (unsigned long)addr; /* Destination address */
+ SAR1 = (unsigned long)&PDIR2; /* Source address */
+ BCR1 = size; /* Bytes to transfer */
+
+ rec_peak_addr = (unsigned long *)addr;
+
+ pcm_apply_settings(false);
+
+ /* Start the DMA transfer.. */
+#ifdef HAVE_SPDIF_IN
+ INTERRUPTCLEAR = 0x03c00000;
+#endif
+
+ DCR1 = DMA_INT | DMA_EEXT | DMA_CS | DMA_DINC |
+ DMA_DSIZE(3) | DMA_START;
+} /* pcm_dma_start */
+
+void pcm_rec_dma_stop(void)
+{
+ logf("pcm_rec_dma_stop");
+
+ pcm_recording = false;
+
+ DCR1 = 0;
+ DSR1 = 1; /* Clear interrupt */
+} /* pcm_dma_stop */
+
+void pcm_init_recording(void)
+{
+ logf("pcm_init_recording");
+
+ pcm_recording = false;
+ pcm_callback_more_ready = NULL;
+
+ AUDIOGLOB |= 0x180; /* IIS1 fifo auto sync = on, PDIR2 auto sync = on */
+
+ DIVR1 = 55; /* DMA1 is mapped into vector 55 in system.c */
+ DMACONFIG = 1; /* DMA0Req = PDOR3, DMA1Req = PDIR2 */
+ DMAROUTE = (DMAROUTE & 0xffff00ff) | DMA1_REQ_AUDIO_2;
+
+#ifdef HAVE_SPDIF_IN
+ /* PHASECONFIG setup: gain = 3*2^13, source = EBUIN */
+ PHASECONFIG = (6 << 3) | (4 << 0);
+#endif
+
+ pcm_rec_dma_stop();
+
+ ICR7 = (7 << 2); /* Enable interrupt at level 7, priority 0 */
+ IMR &= ~(1 << 15); /* bit 15 is DMA1 */
+} /* pcm_init_recording */
+
+void pcm_close_recording(void)
+{
+ logf("pcm_close_recording");
+
+ pcm_rec_dma_stop();
+
+ DMAROUTE &= 0xffff00ff;
+ ICR7 = 0x00; /* Disable interrupt */
+ IMR |= (1 << 15); /* bit 15 is DMA1 */
+} /* pcm_close_recording */
+
+/* DMA1 Interrupt is called when the DMA has finished transfering a chunk
+ into the caller's buffer */
+void DMA1(void) __attribute__ ((interrupt_handler, section(".icode")));
+void DMA1(void)
+{
+ int res = DSR1;
+ pcm_more_callback_type more_ready;
+ unsigned char *next_start;
+ ssize_t next_size = 0; /* passing <> 0 is indicates
+ an error condition */
+
+ DSR1 = 1; /* Clear interrupt */
+ DCR1 &= ~DMA_EEXT;
+
+ if (res & 0x70)
+ {
+ next_size = DMA_REC_ERROR_DMA;
+ logf("DMA1 err: 0x%x", res);
+ }
+#ifdef HAVE_SPDIF_IN
+ else if (pcm_rec_src == AUDIO_SRC_SPDIF &&
+ (INTERRUPTSTAT & 0x01c00000)) /* valnogood, symbolerr, parityerr */
+ {
+ INTERRUPTCLEAR = 0x03c00000;
+ next_size = DMA_REC_ERROR_SPDIF;
+ logf("spdif err");
+ }
+#endif
+
+ more_ready = pcm_callback_more_ready;
+
+ if (more_ready)
+ more_ready(&next_start, &next_size);
+
+ if (next_size > 0)
+ {
+ /* Start peaking at dest */
+ rec_peak_addr = (unsigned long *)next_start;
+ DAR1 = (unsigned long)next_start; /* Destination address */
+ BCR1 = (unsigned long)next_size; /* Bytes to transfer */
+ DCR1 |= DMA_EEXT;
+ return;
+ }
+ else
+ {
+#if 0
+ /* int. logfs can trash the display */
+ logf("DMA1 No Data:0x%04x", res);
+#endif
+ }
+
+ /* Finished recording */
+ pcm_rec_dma_stop();
+} /* DMA1 */
+
+void pcm_mute(bool mute)
+{
+ ac_mute(mute);
+ if (mute)
+ sleep(HZ/16);
+} /* pcm_mute */
+
+void pcm_play_pause_pause(void)
+{
+ /* Disable DMA peripheral request. */
+ DCR0 &= ~DMA_EEXT;
+ pcm_apply_settings(true);
+#ifdef HAVE_SPDIF_OUT
+ EBU1CONFIG = EBU_DEFPARM;
+#endif
+} /* pcm_play_pause_pause */
+
+void pcm_play_pause_unpause(void)
+{
+ /* Enable the FIFO and force one write to it */
+ pcm_apply_settings(false);
+#ifdef HAVE_SPDIF_OUT
+ EBU1CONFIG = EBU_DEFPARM;
+#endif
+ DCR0 |= DMA_EEXT | DMA_START;
+} /* pcm_play_pause_unpause */
+
+/**
+ * Return playback peaks - Peaks ahead in the DMA buffer based upon the
+ * calling period to attempt to compensate for
+ * delay.
+ */
+void pcm_calculate_peaks(int *left, int *right)
+{
+ unsigned long samples;
+ unsigned long *addr, *end;
+ long peak_p, peak_n;
+ int level;
+
+ static unsigned long last_peak_tick = 0;
+ static unsigned long frame_period = 0;
+
+ /* Throttled peak ahead based on calling period */
+ unsigned long period = current_tick - last_peak_tick;
+
+ /* Keep reasonable limits on period */
+ if (period < 1)
+ period = 1;
+ else if (period > HZ/5)
+ period = HZ/5;
+
+ frame_period = (3*frame_period + period) >> 2;
+
+ last_peak_tick = current_tick;
+
+ if (!pcm_playing || pcm_paused)
+ {
+ play_peak_left = play_peak_right = 0;
+ goto peak_done;
+ }
+
+ /* prevent interrupt from setting up next transfer and
+ be sure SAR0 and BCR0 refer to current transfer */
+ level = set_irq_level(HIGHEST_IRQ_LEVEL);
+
+ addr = (long *)(SAR0 & ~3);
+ samples = (BCR0 & 0xffffff) >> 2;
+
+ set_irq_level(level);
+
+ samples = MIN(frame_period*pcm_freq/HZ, samples);
+ end = addr + samples;
+ peak_p = peak_n = 0;
+
+ if (left && right)
+ {
+ if (samples > 0)
+ {
+ long peak_rp = 0, peak_rn = 0;
+
+ do
+ {
+ long value = *addr;
+ long ch;
+
+ ch = value >> 16;
+ if (ch > peak_p) peak_p = ch;
+ else if (ch < peak_n) peak_n = ch;
+
+ ch = (short)value;
+ if (ch > peak_rp) peak_rp = ch;
+ else if (ch < peak_rn) peak_rn = ch;
+
+ addr += 4;
+ }
+ while (addr < end);
+
+ play_peak_left = MAX(peak_p, -peak_n);
+ play_peak_right = MAX(peak_rp, -peak_rn);
+ }
+ }
+ else if (left || right)
+ {
+ if (samples > 0)
+ {
+ if (left)
+ {
+ /* Put left channel in low word */
+ addr = (long *)((short *)addr - 1);
+ end = (long *)((short *)end - 1);
+ }
+
+ do
+ {
+ long value = *(short *)addr;
+
+ if (value > peak_p) peak_p = value;
+ else if (value < peak_n) peak_n = value;
+
+ addr += 4;
+ }
+ while (addr < end);
+
+ if (left)
+ play_peak_left = MAX(peak_p, -peak_n);
+ else
+ play_peak_right = MAX(peak_p, -peak_n);
+ }
+ }
+
+peak_done:
+ if (left)
+ *left = play_peak_left;
+
+ if (right)
+ *right = play_peak_right;
+} /* pcm_calculate_peaks */
+
+/**
+ * Return recording peaks - Looks at every 4th sample from last peak up to
+ * current write position.
+ */
+void pcm_calculate_rec_peaks(int *left, int *right)
+{
+ unsigned long *pkaddr, *addr, *end;
+ long peak_lp, peak_ln; /* L +,- */
+ long peak_rp, peak_rn; /* R +,- */
+ int level;
+
+ if (!pcm_recording)
+ {
+ rec_peak_left = rec_peak_right = 0;
+ goto peak_done;
+ }
+
+ /* read these atomically or each value may not refer to the
+ same data transfer */
+ level = set_irq_level(HIGHEST_IRQ_LEVEL);
+
+ pkaddr = rec_peak_addr;
+ addr = pkaddr;
+ end = (unsigned long *)(DAR1 & ~3);
+
+ set_irq_level(level);
+
+ if (addr < end)
+ {
+ peak_lp = peak_ln =
+ peak_rp = peak_rn = 0;
+
+ /* peak one sample per line */
+ do
+ {
+ long value = *addr;
+ long ch;
+
+ ch = value >> 16;
+ if (ch < peak_ln)
+ peak_ln = ch;
+ else if (ch > peak_lp)
+ peak_lp = ch;
+
+ ch = (short)value;
+ if (ch > peak_rp)
+ peak_rp = ch;
+ else if (ch < peak_rn)
+ peak_rn = ch;
+
+ addr += 4;
+ }
+ while (addr < end);
+
+ /* only update rec_peak_addr if a DMA interrupt hasn't already
+ done so */
+ level = set_irq_level(HIGHEST_IRQ_LEVEL);
+
+ if (pkaddr == rec_peak_addr)
+ rec_peak_addr = end;
+
+ set_irq_level(level);
+
+ /* save peaks */
+ rec_peak_left = MAX(peak_lp, -peak_ln);
+ rec_peak_right = MAX(peak_rp, -peak_rn);
+ }
+
+peak_done:
+ if (left)
+ *left = rec_peak_left;
+
+ if (right)
+ *right = rec_peak_right;
+} /* pcm_calculate_rec_peaks */
+
+/**
+ * Select VINL & VINR source: 0=Line-in, 1=FM Radio
+ */
+/* All use GPIO */
+#if defined(IAUDIO_X5)
+ #define REC_MUX_BIT (1 << 29)
+ #define REC_MUX_SET_LINE() or_l(REC_MUX_BIT, &GPIO_OUT)
+ #define REC_MUX_SET_FM() and_l(~REC_MUX_BIT, &GPIO_OUT)
+#else
+#if defined(IRIVER_H100_SERIES)
+ #define REC_MUX_BIT (1 << 23)
+#elif defined(IRIVER_H300_SERIES)
+ #define REC_MUX_BIT (1 << 30)
+#endif
+ #define REC_MUX_SET_LINE() and_l(~REC_MUX_BIT, &GPIO_OUT)
+ #define REC_MUX_SET_FM() or_l(REC_MUX_BIT, &GPIO_OUT)
+#endif
+
+void pcm_rec_mux(int source)
+{
+ if (source == 0)
+ REC_MUX_SET_LINE(); /* Line In */
+ else
+ REC_MUX_SET_FM(); /* FM radio */
+
+ or_l(REC_MUX_BIT, &GPIO_ENABLE);
+ or_l(REC_MUX_BIT, &GPIO_FUNCTION);
+} /* pcm_rec_mux */
generated by cgit (git 2.34.1) at 2024-04-16 16:07:36 +0000