Forgot to use cvs add on a few new files in the fresh checkout I used. woops.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11453 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis 2006-11-06 18:18:05 +00:00
parent 0f5cb94aa4
commit 5efee7c94a
9 changed files with 2060 additions and 0 deletions

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/***************************************************************************
* __________ __ ___.
* 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 */

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/***************************************************************************
* __________ __ ___.
* 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 */

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/***************************************************************************
* __________ __ ___.
* 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 **/

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/***************************************************************************
* __________ __ ___.
* 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 */

38
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/***************************************************************************
* __________ __ ___.
* 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 */

310
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/***************************************************************************
* __________ __ ___.
* 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 */

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/***************************************************************************
* __________ __ ___.
* 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 */

76
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/***************************************************************************
* __________ __ ___.
* 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 */

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/***************************************************************************
* __________ __ ___.
* 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 */