codec_builtin.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2014, Digium, Inc.
 *
 * Joshua Colp <[email protected]>
 *
 * 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 <[email protected]>
 */

/*** 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,
};

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;
}