lavf: replace AVStream.codec with AVStream.codecpar

Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.

In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.

There are multiple important problems with this approach:
    - the fields in AVCodecContext are in general one of
        * stream parameters
        * codec options
        * codec state
      However, it's not clear which ones are which. It is consequently
      unclear which fields are a demuxer allowed to set or a muxer allowed to
      read. This leads to erratic behaviour depending on whether decoding or
      encoding is being performed or not (and whether it uses the AVStream
      embedded codec context).
    - various synchronization issues arising from the fact that the same
      context is used by several different APIs (muxers/demuxers,
      parsers, bitstream filters and encoders/decoders) simultaneously, with
      there being no clear rules for who can modify what and the different
      processes being typically delayed with respect to each other.
    - avformat_find_stream_info() making it necessary to support opening
      and closing a single codec context multiple times, thus
      complicating the semantics of freeing various allocated objects in the
      codec context.

Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.

1 files changed, 17 insertions, 17 deletions

diff --git a/libavformat/voc_packet.c b/libavformat/voc_packet.c
index 9e1c7469da..ff48d502f2 100644
--- a/libavformat/voc_packet.c
+++ b/libavformat/voc_packet.c
@@ -24,7 +24,7 @@ int
 ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 {
     VocDecContext *voc = s->priv_data;
-    AVCodecContext *dec = st->codec;
+    AVCodecParameters *par = st->codecpar;
     AVIOContext *pb = s->pb;
     VocType type;
     int size, tmp_codec=-1;
@@ -45,13 +45,13 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 
         switch (type) {
         case VOC_TYPE_VOICE_DATA:
-            if (!dec->sample_rate) {
-                dec->sample_rate = 1000000 / (256 - avio_r8(pb));
+            if (!par->sample_rate) {
+                par->sample_rate = 1000000 / (256 - avio_r8(pb));
                 if (sample_rate)
-                    dec->sample_rate = sample_rate;
-                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
-                dec->channels = channels;
-                dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
+                    par->sample_rate = sample_rate;
+                avpriv_set_pts_info(st, 64, 1, par->sample_rate);
+                par->channels = channels;
+                par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
             } else
                 avio_skip(pb, 1);
             tmp_codec = avio_r8(pb);
@@ -73,11 +73,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
             break;
 
         case VOC_TYPE_NEW_VOICE_DATA:
-            if (!dec->sample_rate) {
-                dec->sample_rate = avio_rl32(pb);
-                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
-                dec->bits_per_coded_sample = avio_r8(pb);
-                dec->channels = avio_r8(pb);
+            if (!par->sample_rate) {
+                par->sample_rate = avio_rl32(pb);
+                avpriv_set_pts_info(st, 64, 1, par->sample_rate);
+                par->bits_per_coded_sample = avio_r8(pb);
+                par->channels = avio_r8(pb);
             } else
                 avio_skip(pb, 6);
             tmp_codec = avio_rl16(pb);
@@ -96,11 +96,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 
     if (tmp_codec >= 0) {
         tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
-        if (dec->codec_id == AV_CODEC_ID_NONE)
-            dec->codec_id = tmp_codec;
-        else if (dec->codec_id != tmp_codec)
+        if (par->codec_id == AV_CODEC_ID_NONE)
+            par->codec_id = tmp_codec;
+        else if (par->codec_id != tmp_codec)
             av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
-        if (dec->codec_id == AV_CODEC_ID_NONE) {
+        if (par->codec_id == AV_CODEC_ID_NONE) {
             if (s->audio_codec_id == AV_CODEC_ID_NONE) {
                 av_log(s, AV_LOG_ERROR, "unknown codec tag\n");
                 return AVERROR(EINVAL);
@@ -109,7 +109,7 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
         }
     }
 
-    dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample;
+    par->bit_rate = par->sample_rate * par->bits_per_coded_sample;
 
     if (max_size <= 0)
         max_size = 2048;