diff options
| author | Luca Abeni <lucabe72@email.it> | 2008-09-08 14:24:59 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2008-09-08 14:24:59 +0000 |
| commit | dd1c8f3e6e5380f993c86750bb09fd42e130143f (patch) | |
| tree | 2ae5b2bbda1b685069f4db6ed408097288590dde /libavformat/flvdec.c | |
| parent | 71375e05006e68fecdeb8d5fa80c3cce52a5cf86 (diff) | |
Bump Major version, this commit is almost just renaming bits_per_sample to
bits_per_coded_sample but that cannot be done seperately.
Patch by Luca Abeni
Also reset the minor version and fix the forgotton change to libfaad.
Note: The API/ABI should not be considered stable yet, there still may
be a change done here or there if some developer has some cleanup ideas and
patches!
Originally committed as revision 15262 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/flvdec.c')
| -rw-r--r-- | libavformat/flvdec.c | 10 |
1 files changed, 5 insertions, 5 deletions
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c index ff6f98106d..e548fb4a85 100644 --- a/libavformat/flvdec.c +++ b/libavformat/flvdec.c @@ -42,7 +42,7 @@ static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_c switch(flv_codecid) { //no distinction between S16 and S8 PCM codec flags case FLV_CODECID_PCM: - acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 : + acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : #ifdef WORDS_BIGENDIAN CODEC_ID_PCM_S16BE; #else @@ -50,7 +50,7 @@ static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_c #endif break; case FLV_CODECID_PCM_LE: - acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break; + acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break; case FLV_CODECID_AAC : acodec->codec_id = CODEC_ID_AAC; break; case FLV_CODECID_ADPCM: acodec->codec_id = CODEC_ID_ADPCM_SWF; break; case FLV_CODECID_SPEEX: acodec->codec_id = CODEC_ID_SPEEX; break; @@ -185,7 +185,7 @@ static int amf_parse_object(AVFormatContext *s, AVStream *astream, AVStream *vst 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_sample = 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; @@ -382,13 +382,13 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt) } if(is_audio){ - if(!st->codec->sample_rate || !st->codec->bits_per_sample || (!st->codec->codec_id && !st->codec->codec_tag)) { + if(!st->codec->sample_rate || !st->codec->bits_per_coded_sample || (!st->codec->codec_id && !st->codec->codec_tag)) { st->codec->channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1; if((flags & FLV_AUDIO_CODECID_MASK) == FLV_CODECID_NELLYMOSER_8HZ_MONO) st->codec->sample_rate= 8000; else st->codec->sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3); - st->codec->bits_per_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8; + st->codec->bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8; flv_set_audio_codec(s, st, flags & FLV_AUDIO_CODECID_MASK); } }else{ |