asterisk/main/codec_builtin.c

997 lines
23 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2014, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Built-in supported codecs
*
* \author Joshua Colp <jcolp@digium.com>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/ilbc.h"
#include "asterisk/logger.h"
#include "asterisk/astobj2.h"
#include "asterisk/codec.h"
#include "asterisk/format.h"
#include "asterisk/format_cache.h"
#include "asterisk/frame.h"
#include "asterisk/smoother.h"
int __ast_codec_register_with_format(struct ast_codec *codec, const char *format_name,
struct ast_module *mod);
enum frame_type {
TYPE_HIGH, /* 0x0 */
TYPE_LOW, /* 0x1 */
TYPE_SILENCE, /* 0x2 */
TYPE_DONTSEND /* 0x3 */
};
#define TYPE_MASK 0x3
static int g723_len(unsigned char buf)
{
enum frame_type type = buf & TYPE_MASK;
switch(type) {
case TYPE_DONTSEND:
return 0;
break;
case TYPE_SILENCE:
return 4;
break;
case TYPE_HIGH:
return 24;
break;
case TYPE_LOW:
return 20;
break;
default:
ast_log(LOG_WARNING, "Badly encoded frame (%u)\n", type);
}
return -1;
}
static int g723_samples(struct ast_frame *frame)
{
unsigned char *buf = frame->data.ptr;
int pos = 0, samples = 0, res;
while(pos < frame->datalen) {
res = g723_len(buf[pos]);
if (res <= 0)
break;
samples += 240;
pos += res;
}
return samples;
}
static int g723_length(unsigned int samples)
{
return (samples / 240) * 20;
}
static struct ast_codec g723 = {
.name = "g723",
.description = "G.723.1",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 30,
.maximum_ms = 300,
.default_ms = 30,
.minimum_bytes = 20,
.samples_count = g723_samples,
.get_length = g723_length,
};
static int codec2_samples(struct ast_frame *frame)
{
return 160 * (frame->datalen / 6);
}
static int codec2_length(unsigned int samples)
{
return (samples / 160) * 6;
}
static struct ast_codec codec2 = {
.name = "codec2",
.description = "Codec 2",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 6,
.samples_count = codec2_samples,
.get_length = codec2_length,
.smooth = 1,
};
static int none_samples(struct ast_frame *frame)
{
return frame->datalen;
}
static int none_length(unsigned int samples) {
return samples;
}
static struct ast_codec none = {
.name = "none",
.description = "<Null> codec",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000, /* This must have some sample rate to prevent divide by 0 */
.minimum_ms = 10,
.maximum_ms = 150,
.default_ms = 20,
.minimum_bytes = 20,
.samples_count = none_samples,
.get_length = none_length,
};
static int ulaw_samples(struct ast_frame *frame)
{
return frame->datalen;
}
static int ulaw_length(unsigned int samples)
{
return samples;
}
static struct ast_codec ulaw = {
.name = "ulaw",
.description = "G.711 u-law",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 150,
.default_ms = 20,
.minimum_bytes = 80,
.samples_count = ulaw_samples,
.get_length = ulaw_length,
.smooth = 1,
};
static struct ast_codec alaw = {
.name = "alaw",
.description = "G.711 a-law",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 150,
.default_ms = 20,
.minimum_bytes = 80,
.samples_count = ulaw_samples,
.get_length = ulaw_length,
.smooth = 1,
};
static int gsm_samples(struct ast_frame *frame)
{
return 160 * (frame->datalen / 33);
}
static int gsm_length(unsigned int samples)
{
return (samples / 160) * 33;
}
static struct ast_codec gsm = {
.name = "gsm",
.description = "GSM",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 33,
.samples_count = gsm_samples,
.get_length = gsm_length,
.smooth = 1,
};
static int g726_samples(struct ast_frame *frame)
{
return frame->datalen * 2;
}
static int g726_length(unsigned int samples)
{
return samples / 2;
}
static struct ast_codec g726rfc3551 = {
.name = "g726",
.description = "G.726 RFC3551",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 40,
.samples_count = g726_samples,
.get_length = g726_length,
.smooth = 1,
};
static struct ast_codec g726aal2 = {
.name = "g726aal2",
.description = "G.726 AAL2",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 40,
.samples_count = g726_samples,
.get_length = g726_length,
.smooth = 1,
};
static struct ast_codec adpcm = {
.name = "adpcm",
.description = "Dialogic ADPCM",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 40,
.samples_count = g726_samples,
.get_length = g726_length,
.smooth = 1,
};
static int slin_samples(struct ast_frame *frame)
{
return frame->datalen / 2;
}
static int slin_length(unsigned int samples)
{
return samples * 2;
}
static struct ast_codec slin8 = {
.name = "slin",
.description = "16 bit Signed Linear PCM",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 160,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin12 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (12kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 12000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 240,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin16 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (16kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 320,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin24 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (24kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 24000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 480,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin32 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (32kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 32000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 640,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin44 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (44kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 44100,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 882,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin48 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (48kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 48000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 960,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin96 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (96kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 96000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 1920,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin192 = {
.name = "slin",
.description = "16 bit Signed Linear PCM (192kHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 192000,
.minimum_ms = 10,
.maximum_ms = 70,
.default_ms = 20,
.minimum_bytes = 3840,
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static int lpc10_samples(struct ast_frame *frame)
{
int samples = 22 * 8;
/* assumes that the RTP packet contains one LPC10 frame */
samples += (((char *)(frame->data.ptr))[7] & 0x1) * 8;
return samples;
}
static struct ast_codec lpc10 = {
.name = "lpc10",
.description = "LPC10",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 20,
.default_ms = 20,
.minimum_bytes = 7,
.samples_count = lpc10_samples,
.smooth = 1,
};
static int g729_samples(struct ast_frame *frame)
{
return frame->datalen * 8;
}
static int g729_length(unsigned int samples)
{
return samples / 8;
}
static struct ast_codec g729a = {
.name = "g729",
.description = "G.729A",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 230,
.default_ms = 20,
.minimum_bytes = 10,
.samples_count = g729_samples,
.get_length = g729_length,
.smooth = 1,
.smoother_flags = AST_SMOOTHER_FLAG_G729,
};
static unsigned char get_n_bits_at(unsigned char *data, int n, int bit)
{
int byte = bit / 8; /* byte containing first bit */
int rem = 8 - (bit % 8); /* remaining bits in first byte */
unsigned char ret = 0;
if (n <= 0 || n > 8)
return 0;
if (rem < n) {
ret = (data[byte] << (n - rem));
ret |= (data[byte + 1] >> (8 - n + rem));
} else {
ret = (data[byte] >> (rem - n));
}
return (ret & (0xff >> (8 - n)));
}
static int speex_get_wb_sz_at(unsigned char *data, int len, int bit)
{
static const int SpeexWBSubModeSz[] = {
4, 36, 112, 192,
352, 0, 0, 0 };
int off = bit;
unsigned char c;
/* skip up to two wideband frames */
if (((len * 8 - off) >= 5) &&
get_n_bits_at(data, 1, off)) {
c = get_n_bits_at(data, 3, off + 1);
off += SpeexWBSubModeSz[c];
if (((len * 8 - off) >= 5) &&
get_n_bits_at(data, 1, off)) {
c = get_n_bits_at(data, 3, off + 1);
off += SpeexWBSubModeSz[c];
if (((len * 8 - off) >= 5) &&
get_n_bits_at(data, 1, off)) {
ast_log(LOG_WARNING, "Encountered corrupt speex frame; too many wideband frames in a row.\n");
return -1;
}
}
}
return off - bit;
}
static int speex_samples(unsigned char *data, int len)
{
static const int SpeexSubModeSz[] = {
5, 43, 119, 160,
220, 300, 364, 492,
79, 0, 0, 0,
0, 0, 0, 0 };
static const int SpeexInBandSz[] = {
1, 1, 4, 4,
4, 4, 4, 4,
8, 8, 16, 16,
32, 32, 64, 64 };
int bit = 0;
int cnt = 0;
int off;
unsigned char c;
while ((len * 8 - bit) >= 5) {
/* skip wideband frames */
off = speex_get_wb_sz_at(data, len, bit);
if (off < 0) {
ast_log(LOG_WARNING, "Had error while reading wideband frames for speex samples\n");
break;
}
bit += off;
if ((len * 8 - bit) < 5)
break;
/* get control bits */
c = get_n_bits_at(data, 5, bit);
bit += 5;
if (c == 15) {
/* terminator */
break;
} else if (c == 14) {
/* in-band signal; next 4 bits contain signal id */
c = get_n_bits_at(data, 4, bit);
bit += 4;
bit += SpeexInBandSz[c];
} else if (c == 13) {
/* user in-band; next 4 bits contain msg len */
c = get_n_bits_at(data, 4, bit);
bit += 4;
/* after which it's 5-bit signal id + c bytes of data */
bit += 5 + c * 8;
} else if (c > 8) {
/* unknown */
ast_log(LOG_WARNING, "Unknown speex control frame %d\n", c);
break;
} else {
/* skip number bits for submode (less the 5 control bits) */
bit += SpeexSubModeSz[c] - 5;
cnt += 160; /* new frame */
}
}
return cnt;
}
static int speex8_samples(struct ast_frame *frame)
{
return speex_samples(frame->data.ptr, frame->datalen);
}
static struct ast_codec speex8 = {
.name = "speex",
.description = "SpeeX",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 60,
.default_ms = 20,
.minimum_bytes = 10,
.samples_count = speex8_samples,
};
static int speex16_samples(struct ast_frame *frame)
{
return 2 * speex_samples(frame->data.ptr, frame->datalen);
}
static struct ast_codec speex16 = {
.name = "speex",
.description = "SpeeX 16khz",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 10,
.maximum_ms = 60,
.default_ms = 20,
.minimum_bytes = 10,
.samples_count = speex16_samples,
};
static int speex32_samples(struct ast_frame *frame)
{
return 4 * speex_samples(frame->data.ptr, frame->datalen);
}
static struct ast_codec speex32 = {
.name = "speex",
.description = "SpeeX 32khz",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 32000,
.minimum_ms = 10,
.maximum_ms = 60,
.default_ms = 20,
.minimum_bytes = 10,
.samples_count = speex32_samples,
};
static int ilbc_samples(struct ast_frame *frame)
{
struct ilbc_attr *attr = ast_format_get_attribute_data(frame->subclass.format);
const unsigned int mode = attr ? attr->mode : 30;
const unsigned int samples_per_frame = mode * ast_format_get_sample_rate(frame->subclass.format) / 1000;
const unsigned int octets_per_frame = (mode == 20) ? 38 : 50;
return samples_per_frame * frame->datalen / octets_per_frame;
}
static struct ast_codec ilbc = {
.name = "ilbc",
.description = "iLBC",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 300,
.default_ms = 20,
.minimum_bytes = 38,
.samples_count = ilbc_samples,
.smooth = 0,
};
static struct ast_codec g722 = {
.name = "g722",
.description = "G722",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 10,
.maximum_ms = 150,
.default_ms = 20,
.minimum_bytes = 80,
.samples_count = g726_samples,
.get_length = g726_length,
.smooth = 1,
};
static int siren7_samples(struct ast_frame *frame)
{
return frame->datalen * (16000 / 4000);
}
static int siren7_length(unsigned int samples)
{
return samples / (16000 / 4000);
}
static struct ast_codec siren7 = {
.name = "siren7",
.description = "ITU G.722.1 (Siren7, licensed from Polycom)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 20,
.maximum_ms = 80,
.default_ms = 20,
.minimum_bytes = 80,
.samples_count = siren7_samples,
.get_length = siren7_length,
};
static int siren14_samples(struct ast_frame *frame)
{
return (int) frame->datalen * ((float) 32000 / 6000);
}
static int siren14_length(unsigned int samples)
{
return (int) samples / ((float) 32000 / 6000);;
}
static struct ast_codec siren14 = {
.name = "siren14",
.description = "ITU G.722.1 Annex C, (Siren14, licensed from Polycom)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 32000,
.minimum_ms = 20,
.maximum_ms = 80,
.default_ms = 20,
.minimum_bytes = 120,
.samples_count = siren14_samples,
.get_length = siren14_length,
};
static struct ast_codec testlaw = {
.name = "testlaw",
.description = "G.711 test-law",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 10,
.maximum_ms = 150,
.default_ms = 20,
.minimum_bytes = 80,
.samples_count = ulaw_samples,
.get_length = ulaw_length,
.smooth = 1,
};
static int g719_samples(struct ast_frame *frame)
{
return (int) frame->datalen * ((float) 48000 / 8000);
}
static int g719_length(unsigned int samples)
{
return (int) samples / ((float) 48000 / 8000);
}
static struct ast_codec g719 = {
.name = "g719",
.description = "ITU G.719",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 48000,
.minimum_ms = 20,
.maximum_ms = 80,
.default_ms = 20,
.minimum_bytes = 160,
.samples_count = g719_samples,
.get_length = g719_length,
};
static int opus_samples(struct ast_frame *frame)
{
/*
* XXX This is likely not at all what's intended from this
* callback. If you have codec_opus.so loaded then this
* function is overridden anyway. However, since opus is
* variable bit rate and I cannot extract the calculation code
* from the opus library, I am going to punt and assume 20ms
* worth of samples. In testing, this has worked just fine.
* Pass through support doesn't seem to care about the value
* returned anyway.
*/
return ast_format_get_sample_rate(frame->subclass.format) / 50;
}
static struct ast_codec opus = {
.name = "opus",
.description = "Opus Codec",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 48000,
.minimum_ms = 20,
.maximum_ms = 60,
.default_ms = 20,
.samples_count = opus_samples,
.minimum_bytes = 10,
};
static struct ast_codec jpeg = {
.name = "jpeg",
.description = "JPEG image",
.type = AST_MEDIA_TYPE_IMAGE,
};
static struct ast_codec png = {
.name = "png",
.description = "PNG Image",
.type = AST_MEDIA_TYPE_IMAGE,
};
static struct ast_codec h261 = {
.name = "h261",
.description = "H.261 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec h263 = {
.name = "h263",
.description = "H.263 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec h263p = {
.name = "h263p",
.description = "H.263+ video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec h264 = {
.name = "h264",
.description = "H.264 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec h265 = {
.name = "h265",
.description = "H.265 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec mpeg4 = {
.name = "mpeg4",
.description = "MPEG4 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec vp8 = {
.name = "vp8",
.description = "VP8 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec vp9 = {
.name = "vp9",
.description = "VP9 video",
.type = AST_MEDIA_TYPE_VIDEO,
.sample_rate = 1000,
};
static struct ast_codec t140red = {
.name = "red",
.description = "T.140 Realtime Text with redundancy",
.type = AST_MEDIA_TYPE_TEXT,
};
static struct ast_codec t140 = {
.name = "t140",
.description = "Passthrough T.140 Realtime Text",
.type = AST_MEDIA_TYPE_TEXT,
};
static struct ast_codec t38 = {
.name = "t38",
.description = "T.38 UDPTL Fax",
.type = AST_MEDIA_TYPE_IMAGE,
};
static int silk_samples(struct ast_frame *frame)
{
/* XXX This is likely not at all what's intended from this callback. However,
* since SILK is variable bit rate, I have no idea how to take a frame of data
* and determine the number of samples present. Instead, we base this on the
* sample rate of the codec and the expected number of samples to receive in 20ms.
* In testing, this has worked just fine.
*/
return ast_format_get_sample_rate(frame->subclass.format) / 50;
}
static struct ast_codec silk8 = {
.name = "silk",
.description = "SILK Codec (8 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 160,
.samples_count = silk_samples
};
static struct ast_codec silk12 = {
.name = "silk",
.description = "SILK Codec (12 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 12000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 240,
.samples_count = silk_samples
};
static struct ast_codec silk16 = {
.name = "silk",
.description = "SILK Codec (16 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 320,
.samples_count = silk_samples
};
static struct ast_codec silk24 = {
.name = "silk",
.description = "SILK Codec (24 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 24000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 480,
.samples_count = silk_samples
};
#define CODEC_REGISTER_AND_CACHE(codec) \
({ \
int __res_ ## __LINE__ = 0; \
struct ast_format *__fmt_ ## __LINE__; \
struct ast_codec *__codec_ ## __LINE__; \
res |= __ast_codec_register_with_format(&(codec), (codec).name, NULL); \
__codec_ ## __LINE__ = ast_codec_get((codec).name, (codec).type, (codec).sample_rate); \
__fmt_ ## __LINE__ = __codec_ ## __LINE__ ? ast_format_create(__codec_ ## __LINE__) : NULL; \
res |= ast_format_cache_set(__fmt_ ## __LINE__); \
ao2_ref(__fmt_ ## __LINE__, -1); \
ao2_ref(__codec_ ## __LINE__, -1); \
__res_ ## __LINE__; \
})
#define CODEC_REGISTER_AND_CACHE_NAMED(fmt_name, codec) \
({ \
int __res_ ## __LINE__ = 0; \
struct ast_format *__fmt_ ## __LINE__; \
struct ast_codec *__codec_ ## __LINE__; \
res |= __ast_codec_register_with_format(&(codec), fmt_name, NULL); \
__codec_ ## __LINE__ = ast_codec_get((codec).name, (codec).type, (codec).sample_rate); \
__fmt_ ## __LINE__ = ast_format_create_named((fmt_name), __codec_ ## __LINE__); \
res |= ast_format_cache_set(__fmt_ ## __LINE__); \
ao2_ref(__fmt_ ## __LINE__, -1); \
ao2_ref(__codec_ ## __LINE__, -1); \
__res_ ## __LINE__; \
})
int ast_codec_builtin_init(void)
{
int res = 0;
res |= CODEC_REGISTER_AND_CACHE(codec2);
res |= CODEC_REGISTER_AND_CACHE(g723);
res |= CODEC_REGISTER_AND_CACHE(ulaw);
res |= CODEC_REGISTER_AND_CACHE(alaw);
res |= CODEC_REGISTER_AND_CACHE(gsm);
res |= CODEC_REGISTER_AND_CACHE(g726rfc3551);
res |= CODEC_REGISTER_AND_CACHE(g726aal2);
res |= CODEC_REGISTER_AND_CACHE(adpcm);
res |= CODEC_REGISTER_AND_CACHE(slin8);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin12", slin12);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin16", slin16);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin24", slin24);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin32", slin32);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin44", slin44);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin48", slin48);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin96", slin96);
res |= CODEC_REGISTER_AND_CACHE_NAMED("slin192", slin192);
res |= CODEC_REGISTER_AND_CACHE(lpc10);
res |= CODEC_REGISTER_AND_CACHE(g729a);
res |= CODEC_REGISTER_AND_CACHE(speex8);
res |= CODEC_REGISTER_AND_CACHE_NAMED("speex16", speex16);
res |= CODEC_REGISTER_AND_CACHE_NAMED("speex32", speex32);
res |= CODEC_REGISTER_AND_CACHE(ilbc);
res |= CODEC_REGISTER_AND_CACHE(g722);
res |= CODEC_REGISTER_AND_CACHE(siren7);
res |= CODEC_REGISTER_AND_CACHE(siren14);
res |= CODEC_REGISTER_AND_CACHE(testlaw);
res |= CODEC_REGISTER_AND_CACHE(g719);
res |= CODEC_REGISTER_AND_CACHE(opus);
res |= CODEC_REGISTER_AND_CACHE(jpeg);
res |= CODEC_REGISTER_AND_CACHE(png);
res |= CODEC_REGISTER_AND_CACHE(h261);
res |= CODEC_REGISTER_AND_CACHE(h263);
res |= CODEC_REGISTER_AND_CACHE(h263p);
res |= CODEC_REGISTER_AND_CACHE(h264);
res |= CODEC_REGISTER_AND_CACHE(h265);
res |= CODEC_REGISTER_AND_CACHE(mpeg4);
res |= CODEC_REGISTER_AND_CACHE(vp8);
res |= CODEC_REGISTER_AND_CACHE(vp9);
res |= CODEC_REGISTER_AND_CACHE(t140red);
res |= CODEC_REGISTER_AND_CACHE(t140);
res |= CODEC_REGISTER_AND_CACHE(t38);
res |= CODEC_REGISTER_AND_CACHE(none);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk8", silk8);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
return res;
}