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-rw-r--r--libavformat/flvdec.c25
1 files changed, 0 insertions, 25 deletions
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index 060d817c6b..56b6fa5cfc 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -218,35 +218,10 @@ static int amf_parse_object(AVFormatContext *s, AVStream *astream, AVStream *vst
vcodec = vstream ? vstream->codec : NULL;
if(amf_type == AMF_DATA_TYPE_BOOL) {
- if(!strcmp(key, "stereo") && acodec) acodec->channels = num_val > 0 ? 2 : 1;
} else if(amf_type == AMF_DATA_TYPE_NUMBER) {
if(!strcmp(key, "duration")) s->duration = num_val * AV_TIME_BASE;
-// else if(!strcmp(key, "width") && vcodec && num_val > 0) vcodec->width = num_val;
-// else if(!strcmp(key, "height") && vcodec && num_val > 0) vcodec->height = num_val;
else if(!strcmp(key, "videodatarate") && vcodec && 0 <= (int)(num_val * 1024.0))
vcodec->bit_rate = num_val * 1024.0;
- else if(!strcmp(key, "audiocodecid") && acodec && 0 <= (int)num_val)
- flv_set_audio_codec(s, astream, (int)num_val << FLV_AUDIO_CODECID_OFFSET);
- else if(!strcmp(key, "videocodecid") && vcodec && 0 <= (int)num_val)
- flv_set_video_codec(s, vstream, (int)num_val);
- else if(!strcmp(key, "audiosamplesize") && acodec && 0 < (int)num_val) {
- acodec->bits_per_coded_sample = num_val;
- //we may have to rewrite a previously read codecid because FLV only marks PCM endianness.
- if(num_val == 8 && (acodec->codec_id == CODEC_ID_PCM_S16BE || acodec->codec_id == CODEC_ID_PCM_S16LE))
- acodec->codec_id = CODEC_ID_PCM_S8;
- }
- else if(!strcmp(key, "audiosamplerate") && acodec && num_val >= 0) {
- //some tools, like FLVTool2, write consistently approximate metadata sample rates
- if (!acodec->sample_rate) {
- switch((int)num_val) {
- case 44000: acodec->sample_rate = 44100 ; break;
- case 22000: acodec->sample_rate = 22050 ; break;
- case 11000: acodec->sample_rate = 11025 ; break;
- case 5000 : acodec->sample_rate = 5512 ; break;
- default : acodec->sample_rate = num_val;
- }
- }
- }
}
}