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| author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-01-15 13:38:03 -0500 |
|---|---|---|
| committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-03-03 16:43:11 -0500 |
| commit | 5602a464c9f9e3c0922f5cfeccaf2fa1c40b2401 (patch) | |
| tree | 783511b998940d65fe61a2da3f81e46bbf34cf6d /libavformat | |
| parent | 737ca4482bd68d6dc52a1811f5de8e4952ad7036 (diff) | |
| download | ffmpeg-5602a464c9f9e3c0922f5cfeccaf2fa1c40b2401.tar.gz | |
avcodec: add a Vorbis parser to get packet duration
This also allows for removing some of the Vorbis-related hacks.
Diffstat (limited to 'libavformat')
| -rw-r--r-- | libavformat/oggparsevorbis.c | 1 | ||||
| -rw-r--r-- | libavformat/rtpdec.c | 3 | ||||
| -rw-r--r-- | libavformat/utils.c | 6 |
3 files changed, 5 insertions, 5 deletions
diff --git a/libavformat/oggparsevorbis.c b/libavformat/oggparsevorbis.c index ba9b348456..b9d9f575e0 100644 --- a/libavformat/oggparsevorbis.c +++ b/libavformat/oggparsevorbis.c @@ -251,6 +251,7 @@ vorbis_header (AVFormatContext * s, int idx) st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_VORBIS; + st->need_parsing = AVSTREAM_PARSE_HEADERS; if (srate > 0) { st->codec->sample_rate = srate; diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index a8c5c3ff4c..61653f7b39 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -396,6 +396,9 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext case CODEC_ID_H264: st->need_parsing = AVSTREAM_PARSE_FULL; break; + case CODEC_ID_VORBIS: + st->need_parsing = AVSTREAM_PARSE_HEADERS; + break; case CODEC_ID_ADPCM_G722: /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ diff --git a/libavformat/utils.c b/libavformat/utils.c index cf4392b581..123bc8bf9a 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -701,9 +701,6 @@ static int get_audio_frame_size(AVCodecContext *enc, int size) { int frame_size; - if(enc->codec_id == CODEC_ID_VORBIS) - return -1; - if (enc->frame_size <= 1) { int bits_per_sample = av_get_bits_per_sample(enc->codec_id); @@ -1995,8 +1992,7 @@ static int has_codec_parameters(AVCodecContext *avctx) case AVMEDIA_TYPE_AUDIO: val = avctx->sample_rate && avctx->channels && avctx->sample_fmt != AV_SAMPLE_FMT_NONE; if (!avctx->frame_size && - (avctx->codec_id == CODEC_ID_VORBIS || - avctx->codec_id == CODEC_ID_AAC || + (avctx->codec_id == CODEC_ID_AAC || avctx->codec_id == CODEC_ID_MP1 || avctx->codec_id == CODEC_ID_MP2 || avctx->codec_id == CODEC_ID_MP3 || |