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https://gitee.com/openharmony/third_party_ffmpeg
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cbr audio muxing fix
Originally committed as revision 3661 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
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70122f2902
commit
359fa0febc
@ -127,6 +127,23 @@ static int pcm_encode_init(AVCodecContext *avctx)
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break;
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}
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_S16LE:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_U16BE:
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avctx->block_align = 2 * avctx->channels;
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break;
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case CODEC_ID_PCM_S8:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_MULAW:
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case CODEC_ID_PCM_ALAW:
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avctx->block_align = avctx->channels;
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break;
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default:
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break;
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}
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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@ -251,33 +251,24 @@ void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const CodecTag *tags
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put_byte(pb, 0);
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}
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static void parse_specific_params(AVCodecContext *stream, int *au_byterate, int *au_ssize, int *au_scale)
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static void parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
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{
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switch(stream->codec_id) {
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case CODEC_ID_PCM_S16LE:
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*au_scale = *au_ssize = 2*stream->channels;
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*au_byterate = *au_ssize * stream->sample_rate;
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break;
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_MULAW:
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*au_scale = *au_ssize = stream->channels;
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*au_byterate = *au_ssize * stream->sample_rate;
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break;
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case CODEC_ID_MP2:
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*au_ssize = 1;
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*au_scale = 1;
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*au_byterate = stream->bit_rate / 8;
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case CODEC_ID_MP3:
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*au_ssize = 1;
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*au_scale = 1;
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*au_byterate = stream->bit_rate / 8;
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default:
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*au_ssize = 1;
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*au_scale = 1;
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*au_byterate = stream->bit_rate / 8;
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break;
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int gcd;
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*au_ssize= stream->block_align;
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if(stream->frame_size && stream->sample_rate){
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*au_scale=stream->frame_size;
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*au_rate= stream->sample_rate;
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}else if(stream->codec_type == CODEC_TYPE_VIDEO){
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*au_scale= stream->frame_rate_base;
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*au_rate = stream->frame_rate;
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}else{
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*au_scale= stream->block_align ? stream->block_align*8 : 8;
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*au_rate = stream->bit_rate;
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}
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gcd= ff_gcd(*au_scale, *au_rate);
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*au_scale /= gcd;
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*au_rate /= gcd;
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}
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static offset_t avi_start_new_riff(AVIContext *avi, ByteIOContext *pb,
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@ -595,18 +586,12 @@ static int avi_write_idx1(AVFormatContext *s)
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if (avi->frames_hdr_strm[n] != 0) {
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stream = &s->streams[n]->codec;
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url_fseek(pb, avi->frames_hdr_strm[n], SEEK_SET);
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if (stream->codec_type == CODEC_TYPE_VIDEO) {
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put_le32(pb, stream->frame_number);
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if (nb_frames < stream->frame_number)
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nb_frames = stream->frame_number;
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parse_specific_params(stream, &au_byterate, &au_ssize, &au_scale);
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if (au_ssize == 0) {
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put_le32(pb, stream->frame_number);
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nb_frames += stream->frame_number;
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} else {
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if (stream->codec_id == CODEC_ID_MP2 || stream->codec_id == CODEC_ID_MP3) {
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put_le32(pb, stream->frame_number);
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nb_frames += stream->frame_number;
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} else {
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parse_specific_params(stream, &au_byterate, &au_ssize, &au_scale);
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put_le32(pb, avi->audio_strm_length[n] / au_ssize);
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}
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put_le32(pb, avi->audio_strm_length[n] / au_ssize);
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}
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}
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}
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@ -71,7 +71,7 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
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}
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if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
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blkalign = 1;
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blkalign = enc->frame_size; //this is wrong, but seems many demuxers dont work if this is set correctly
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//blkalign = 144 * enc->bit_rate/enc->sample_rate;
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} else if (enc->block_align != 0) { /* specified by the codec */
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blkalign = enc->block_align;
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@ -1,8 +1,8 @@
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ffmpeg regression test
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580b53ca26eb2184b928967b8e7d18a7 *./data/b-libav.avi
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8517e7c83227074b8d632477fda310d5 *./data/b-libav.avi
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342282 ./data/b-libav.avi
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./data/b-libav.avi CRC=001a3415
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80f85b6dce1da8310a49bb4607a34f06 *./data/b-libav.asf
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c83444a0e8ef47b6af1d868d1bb21696 *./data/b-libav.asf
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342967 ./data/b-libav.asf
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./data/b-libav.asf CRC=750f18c7
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1cbf838e659d7fc3d3e33f4187b91f6c *./data/b-libav.rm
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