diff options
| author | Stefano Sabatini <stefano.sabatini-lala@poste.it> | 2010-11-12 11:04:40 +0000 |
|---|---|---|
| committer | Stefano Sabatini <stefano.sabatini-lala@poste.it> | 2010-11-12 11:04:40 +0000 |
| commit | 5d6e4c160a4a0d71c17e8428123027c747ff0fb3 (patch) | |
| tree | d3132b2b615fe19f3f4b5ad43b095c076320c780 /libavformat | |
| parent | 09f47fa44ebf3f18651397517b49e6f8c5a0e374 (diff) | |
Replace deprecated symbols SAMPLE_FMT_* with AV_SAMPLE_FMT_*, and enum
SampleFormat with AVSampleFormat.
Originally committed as revision 25730 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat')
| -rw-r--r-- | libavformat/flic.c | 2 | ||||
| -rw-r--r-- | libavformat/output-example.c | 2 | ||||
| -rw-r--r-- | libavformat/utils.c | 2 |
3 files changed, 3 insertions, 3 deletions
diff --git a/libavformat/flic.c b/libavformat/flic.c index fbdf931517..51320c9da3 100644 --- a/libavformat/flic.c +++ b/libavformat/flic.c @@ -157,7 +157,7 @@ static int flic_read_header(AVFormatContext *s, ast->codec->codec_tag = 0; ast->codec->sample_rate = FLIC_TFTD_SAMPLE_RATE; ast->codec->channels = 1; - ast->codec->sample_fmt = SAMPLE_FMT_U8; + ast->codec->sample_fmt = AV_SAMPLE_FMT_U8; ast->codec->bit_rate = st->codec->sample_rate * 8; ast->codec->bits_per_coded_sample = 8; ast->codec->channel_layout = CH_LAYOUT_MONO; diff --git a/libavformat/output-example.c b/libavformat/output-example.c index 15e2d9ac98..06207eddfc 100644 --- a/libavformat/output-example.c +++ b/libavformat/output-example.c @@ -68,7 +68,7 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id) c->codec_type = AVMEDIA_TYPE_AUDIO; /* put sample parameters */ - c->sample_fmt = SAMPLE_FMT_S16; + c->sample_fmt = AV_SAMPLE_FMT_S16; c->bit_rate = 64000; c->sample_rate = 44100; c->channels = 2; diff --git a/libavformat/utils.c b/libavformat/utils.c index 23d0a3b0d5..03dfbb04c3 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -2015,7 +2015,7 @@ static int has_codec_parameters(AVCodecContext *enc) int val; switch(enc->codec_type) { case AVMEDIA_TYPE_AUDIO: - val = enc->sample_rate && enc->channels && enc->sample_fmt != SAMPLE_FMT_NONE; + val = enc->sample_rate && enc->channels && enc->sample_fmt != AV_SAMPLE_FMT_NONE; if(!enc->frame_size && (enc->codec_id == CODEC_ID_VORBIS || enc->codec_id == CODEC_ID_AAC || |