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riffenc: use av_get_audio_frame_duration()

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For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
This commit is contained in:
Justin Ruggles 2012-02-27 02:34:14 -05:00
parent 9524cf79df
commit c019070fda
1 changed files with 23 additions and 5 deletions
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28
libavformat/riff.c
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@ -387,7 +387,7 @@ void ff_end_tag(AVIOContext *pb, int64_t start)
/* returns the size or -1 on error */
int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
{
int bps, blkalign, bytespersec;
int bps, blkalign, bytespersec, frame_size;
int hdrsize = 18;
int waveformatextensible;
uint8_t temp[256];
@ -396,6 +396,14 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
if(!enc->codec_tag || enc->codec_tag > 0xffff)
return -1;
/* We use the known constant frame size for the codec if known, otherwise
fallback to using AVCodecContext.frame_size, which is not as reliable
for indicating packet duration */
frame_size = av_get_audio_frame_duration(enc, 0);
if (!frame_size)
frame_size = enc->frame_size;
waveformatextensible = (enc->channels > 2 && enc->channel_layout)
|| enc->sample_rate > 48000
|| av_get_bits_per_sample(enc->codec_id) > 16;
@ -422,7 +430,9 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
}
if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
blkalign = enc->frame_size; //this is wrong, but it seems many demuxers do not work if this is set correctly
/* this is wrong, but it seems many demuxers do not work if this is set
correctly */
blkalign = frame_size;
//blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->codec_id == CODEC_ID_AC3) {
blkalign = 3840; //maximum bytes per frame
@ -462,7 +472,7 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
bytestream_put_le32(&riff_extradata, 0); /* dwPTSHigh */
} else if (enc->codec_id == CODEC_ID_GSM_MS || enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) {
hdrsize += 2;
bytestream_put_le16(&riff_extradata, enc->frame_size); /* wSamplesPerBlock */
bytestream_put_le16(&riff_extradata, frame_size); /* wSamplesPerBlock */
} else if(enc->extradata_size){
riff_extradata_start= enc->extradata;
riff_extradata= enc->extradata + enc->extradata_size;
@ -618,10 +628,18 @@ int ff_get_bmp_header(AVIOContext *pb, AVStream *st)
void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
{
int gcd;
int audio_frame_size;
/* We use the known constant frame size for the codec if known, otherwise
fallback to using AVCodecContext.frame_size, which is not as reliable
for indicating packet duration */
audio_frame_size = av_get_audio_frame_duration(stream, 0);
if (!audio_frame_size)
audio_frame_size = stream->frame_size;
*au_ssize= stream->block_align;
if(stream->frame_size && stream->sample_rate){
*au_scale=stream->frame_size;
if (audio_frame_size && stream->sample_rate) {
*au_scale = audio_frame_size;
*au_rate= stream->sample_rate;
}else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
stream->codec_type == AVMEDIA_TYPE_DATA ||