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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility

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Originally committed as revision 4453 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer
2005-07-17 22:24:36 +00:00
parent 4c1a012e9c
commit 01f4895c68
56 changed files with 1023 additions and 1024 deletions
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32
libavformat/swf.c
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@ -324,7 +324,7 @@ static int swf_write_header(AVFormatContext *s)
video_enc = NULL;
audio_enc = NULL;
for(i=0;i<s->nb_streams;i++) {
enc = &s->streams[i]->codec;
enc = s->streams[i]->codec;
if (enc->codec_type == CODEC_TYPE_AUDIO)
audio_enc = enc;
else {
@ -652,7 +652,7 @@ static int swf_write_audio(AVFormatContext *s,
static int swf_write_packet(AVFormatContext *s, AVPacket *pkt)
{
AVCodecContext *codec = &s->streams[pkt->stream_index]->codec;
AVCodecContext *codec = s->streams[pkt->stream_index]->codec;
if (codec->codec_type == CODEC_TYPE_AUDIO)
return swf_write_audio(s, codec, pkt->data, pkt->size);
else
@ -668,7 +668,7 @@ static int swf_write_trailer(AVFormatContext *s)
video_enc = NULL;
for(i=0;i<s->nb_streams;i++) {
enc = &s->streams[i]->codec;
enc = s->streams[i]->codec;
if (enc->codec_type == CODEC_TYPE_VIDEO)
video_enc = enc;
}
@ -775,8 +775,8 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
if (tag < 0) {
if ( ast || vst ) {
if ( vst && ast ) {
vst->codec.time_base.den = ast->codec.sample_rate / swf->samples_per_frame;
vst->codec.time_base.num = 1;
vst->codec->time_base.den = ast->codec->sample_rate / swf->samples_per_frame;
vst->codec->time_base.num = 1;
}
break;
}
@ -794,11 +794,11 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
vst = av_new_stream(s, 0);
av_set_pts_info(vst, 24, 1, 1000); /* 24 bit pts in ms */
vst->codec.codec_type = CODEC_TYPE_VIDEO;
vst->codec.codec_id = CODEC_ID_FLV1;
vst->codec->codec_type = CODEC_TYPE_VIDEO;
vst->codec->codec_id = CODEC_ID_FLV1;
if ( swf->samples_per_frame ) {
vst->codec.time_base.den = 1000. / swf->ms_per_frame;
vst->codec.time_base.num = 1;
vst->codec->time_base.den = 1000. / swf->ms_per_frame;
vst->codec->time_base.num = 1;
}
}
} else if ( ( tag == TAG_STREAMHEAD || tag == TAG_STREAMHEAD2 ) && !ast) {
@ -819,26 +819,26 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
return -ENOMEM;
if (v & 0x01)
ast->codec.channels = 2;
ast->codec->channels = 2;
else
ast->codec.channels = 1;
ast->codec->channels = 1;
switch((v>> 2) & 0x03) {
case 1:
ast->codec.sample_rate = 11025;
ast->codec->sample_rate = 11025;
break;
case 2:
ast->codec.sample_rate = 22050;
ast->codec->sample_rate = 22050;
break;
case 3:
ast->codec.sample_rate = 44100;
ast->codec->sample_rate = 44100;
break;
default:
av_free(ast);
return AVERROR_IO;
}
ast->codec.codec_type = CODEC_TYPE_AUDIO;
ast->codec.codec_id = CODEC_ID_MP3;
ast->codec->codec_type = CODEC_TYPE_AUDIO;
ast->codec->codec_id = CODEC_ID_MP3;
}
} else {
url_fskip(pb, len);