Merge commit '9200514ad8717c63f82101dc394f4378854325bf'

* commit '9200514ad8717c63f82101dc394f4378854325bf':
  lavf: replace AVStream.codec with AVStream.codecpar

This has been a HUGE effort from:
    - Derek Buitenhuis <derek.buitenhuis@gmail.com>
    - Hendrik Leppkes <h.leppkes@gmail.com>
    - wm4 <nfxjfg@googlemail.com>
    - Clément Bœsch <clement@stupeflix.com>
    - James Almer <jamrial@gmail.com>
    - Michael Niedermayer <michael@niedermayer.cc>
    - Rostislav Pehlivanov <atomnuker@gmail.com>

Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

1 files changed, 18 insertions, 18 deletions

diff --git a/libavformat/voc_packet.c b/libavformat/voc_packet.c
index f4b8231..5833a79 100644
--- a/libavformat/voc_packet.c
+++ b/libavformat/voc_packet.c
@@ -27,7 +27,7 @@ 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;
@@ -57,13 +57,13 @@ 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);
@@ -85,11 +85,11 @@ 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);
@@ -108,11 +108,11 @@ 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);
@@ -121,7 +121,7 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
         }
     }
 
-    dec->bit_rate = dec->sample_rate * dec->channels * dec->bits_per_coded_sample;
+    par->bit_rate = par->sample_rate * par->channels * par->bits_per_coded_sample;
 
     if (max_size <= 0)
         max_size = 2048;
@@ -131,7 +131,7 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
     ret = av_get_packet(pb, pkt, size);
     pkt->dts = pkt->pts = voc->pts;
 
-    duration = av_get_audio_frame_duration(st->codec, size);
+    duration = av_get_audio_frame_duration2(st->codecpar, size);
     if (duration > 0 && voc->pts != AV_NOPTS_VALUE)
         voc->pts += duration;
     else