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authorStefano Sabatini <stefano.sabatini-lala@poste.it>2010-11-12 11:04:40 +0000
committerStefano Sabatini <stefano.sabatini-lala@poste.it>2010-11-12 11:04:40 +0000
commit5d6e4c160a4a0d71c17e8428123027c747ff0fb3 (patch)
treed3132b2b615fe19f3f4b5ad43b095c076320c780 /libavformat
parent09f47fa44ebf3f18651397517b49e6f8c5a0e374 (diff)
downloadffmpeg-5d6e4c160a4a0d71c17e8428123027c747ff0fb3.tar.gz
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.c2
-rw-r--r--libavformat/output-example.c2
-rw-r--r--libavformat/utils.c2
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 ||