videolan
/
ffmpeg
mirror of https://git.videolan.org/git/ffmpeg.git
1
Fork 0
Code Releases Activity

g726: use bits_per_coded_sample instead of bitrate to determine mode 

This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.

This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
This commit is contained in:
Justin Ruggles 2011-10-27 20:16:45 -04:00
parent d405237bae
commit 6ac34eed54
2 changed files with 29 additions and 28 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
libavcodec/g726.c
View File

@ -301,29 +301,29 @@ static int16_t g726_encode(G726Context* c, int16_t sig)
static av_cold int g726_encode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
unsigned int index;
if (avctx->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Samplerate is invalid\n");
return -1;
}
index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
if (avctx->bit_rate % avctx->sample_rate) {
av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
return -1;
}
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return -1;
}
if(index>3){
av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
return -1;
if (avctx->bit_rate % avctx->sample_rate) {
av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
return AVERROR(EINVAL);
}
g726_reset(c, index);
c->code_size = index+2;
c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
if (c->code_size < 2 || c->code_size > 5) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
return AVERROR(EINVAL);
}
avctx->bits_per_coded_sample = c->code_size;
g726_reset(c, c->code_size - 2);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
@ -332,7 +332,7 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[index];
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
return 0;
}
@ -365,25 +365,23 @@ static int g726_encode_frame(AVCodecContext *avctx,
static av_cold int g726_decode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
unsigned int index;
if (avctx->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Samplerate is invalid\n");
return -1;
}
index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return -1;
}
if(index>3){
av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
return -1;
c->code_size = avctx->bits_per_coded_sample;
if (c->code_size < 2 || c->code_size > 5) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
return AVERROR(EINVAL);
}
g726_reset(c, index);
c->code_size = index+2;
g726_reset(c, c->code_size - 2);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;

17
libavformat/riff.c
View File

@ -385,11 +385,13 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_GSM_MS) {
bps = 0;
} else if (enc->codec_id == CODEC_ID_ADPCM_G726) {
bps = 4;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id)))
bps = 16; // default to 16
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if(bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample){
av_log(enc, AV_LOG_WARNING, "requested bits_per_coded_sample (%d) and actually stored (%d) differ\n", enc->bits_per_coded_sample, bps);
@ -400,12 +402,10 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
//blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->codec_id == CODEC_ID_AC3) {
blkalign = 3840; //maximum bytes per frame
} else if (enc->codec_id == CODEC_ID_ADPCM_G726) { //
blkalign = 1;
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = enc->channels*bps >> 3;
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == CODEC_ID_PCM_U8 ||
enc->codec_id == CODEC_ID_PCM_S24LE ||
enc->codec_id == CODEC_ID_PCM_S32LE ||
@ -545,6 +545,9 @@ int ff_get_wav_header(AVIOContext *pb, AVCodecContext *codec, int size)
codec->channels = 0;
codec->sample_rate = 0;
}
/* override bits_per_coded_sample for G.726 */
if (codec->codec_id == CODEC_ID_ADPCM_G726)
codec->bits_per_coded_sample = codec->bit_rate / codec->sample_rate;
return 0;
}