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, 22 insertions, 22 deletions

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 6158934900..c7733a6862 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -97,8 +97,8 @@ static int rtp_write_header(AVFormatContext *s1)
         return AVERROR(EINVAL);
     }
     st = s1->streams[0];
-    if (!is_supported(st->codec->codec_id)) {
-        av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
+    if (!is_supported(st->codecpar->codec_id)) {
+        av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codecpar->codec_id);
 
         return -1;
     }
@@ -106,7 +106,7 @@ static int rtp_write_header(AVFormatContext *s1)
     if (s->payload_type < 0) {
         /* Re-validate non-dynamic payload types */
         if (st->id < RTP_PT_PRIVATE)
-            st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
+            st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
 
         s->payload_type = st->id;
     } else {
@@ -149,13 +149,13 @@ static int rtp_write_header(AVFormatContext *s1)
     }
     s->max_payload_size = s1->packet_size - 12;
 
-    if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
-        avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
+    if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+        avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
     } else {
         avpriv_set_pts_info(st, 32, 1, 90000);
     }
     s->buf_ptr = s->buf;
-    switch(st->codec->codec_id) {
+    switch(st->codecpar->codec_id) {
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         s->buf_ptr = s->buf + 4;
@@ -183,8 +183,8 @@ static int rtp_write_header(AVFormatContext *s1)
         break;
     case AV_CODEC_ID_H264:
         /* check for H.264 MP4 syntax */
-        if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
-            s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
+        if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
+            s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
         }
         break;
     case AV_CODEC_ID_HEVC:
@@ -192,8 +192,8 @@ static int rtp_write_header(AVFormatContext *s1)
          * things simple and similar to the avcC/H264 case above, instead
          * of trying to handle the pre-standardization versions (as in
          * libavcodec/hevc.c). */
-        if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
-            s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
+        if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
+            s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
         }
         break;
     case AV_CODEC_ID_VORBIS:
@@ -206,7 +206,7 @@ static int rtp_write_header(AVFormatContext *s1)
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
     case AV_CODEC_ID_OPUS:
-        if (st->codec->channels > 2) {
+        if (st->codecpar->channels > 2) {
             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
             goto fail;
         }
@@ -216,16 +216,16 @@ static int rtp_write_header(AVFormatContext *s1)
         avpriv_set_pts_info(st, 32, 1, 48000);
         break;
     case AV_CODEC_ID_ILBC:
-        if (st->codec->block_align != 38 && st->codec->block_align != 50) {
+        if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
             goto fail;
         }
-        s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
+        s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
         break;
     case AV_CODEC_ID_AMR_NB:
     case AV_CODEC_ID_AMR_WB:
         s->max_frames_per_packet = 50;
-        if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
+        if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
             n = 31;
         else
             n = 61;
@@ -234,7 +234,7 @@ static int rtp_write_header(AVFormatContext *s1)
             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
             goto fail;
         }
-        if (st->codec->channels != 1) {
+        if (st->codecpar->channels != 1) {
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
             goto fail;
         }
@@ -454,8 +454,8 @@ static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
 {
     RTPMuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
-    int frame_duration = av_get_audio_frame_duration(st->codec, 0);
-    int frame_size = st->codec->block_align;
+    int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
+    int frame_size = st->codecpar->block_align;
     int frames = size / frame_size;
 
     while (frames > 0) {
@@ -505,26 +505,26 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     }
     s->cur_timestamp = s->base_timestamp + pkt->pts;
 
-    switch(st->codec->codec_id) {
+    switch(st->codecpar->codec_id) {
     case AV_CODEC_ID_PCM_MULAW:
     case AV_CODEC_ID_PCM_ALAW:
     case AV_CODEC_ID_PCM_U8:
     case AV_CODEC_ID_PCM_S8:
-        return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
+        return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
     case AV_CODEC_ID_PCM_U16BE:
     case AV_CODEC_ID_PCM_U16LE:
     case AV_CODEC_ID_PCM_S16BE:
     case AV_CODEC_ID_PCM_S16LE:
-        return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
+        return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
     case AV_CODEC_ID_ADPCM_G722:
         /* The actual sample size is half a byte per sample, but since the
          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
          * the correct parameter for send_samples_bits is 8 bits per stream
          * clock. */
-        return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
+        return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
     case AV_CODEC_ID_ADPCM_G726:
         return rtp_send_samples(s1, pkt->data, size,
-                                st->codec->bits_per_coded_sample * st->codec->channels);
+                                st->codecpar->bits_per_coded_sample * st->codecpar->channels);
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         rtp_send_mpegaudio(s1, pkt->data, size);