[FFmpeg-cvslog] bethsoftvid: add audio stream only after getting the first audio packet

[Justin Ruggles]

ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Wed Jan 18 19:50:37 2012 -0500| [773ff823dac3da733c251b4cb1cc95b3b7e03001] | committer: Justin Ruggles

bethsoftvid: add audio stream only after getting the first audio packet

This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=773ff823dac3da733c251b4cb1cc95b3b7e03001
---

 libavformat/bethsoftvid.c |   42 +++++++++++++++++++++++++++---------------
 1 files changed, 27 insertions(+), 15 deletions(-)

diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c
index 10be7d2..1092299 100644
--- a/libavformat/bethsoftvid.c
+++ b/libavformat/bethsoftvid.c
@@ -34,14 +34,18 @@
 
 #define BVID_PALETTE_SIZE 3 * 256
 
+#define DEFAULT_SAMPLE_RATE 11111
+
 typedef struct BVID_DemuxContext
 {
     int nframes;
+    int sample_rate;        /**< audio sample rate */
     /** delay value between frames, added to individual frame delay.
      * custom units, which will be added to other custom units (~=16ms according
      * to free, unofficial documentation) */
     int bethsoft_global_delay;
-
+    int video_index;        /**< video stream index */
+    int audio_index;        /**< audio stream index */
     uint8_t *palette;
 
     int is_finished;
@@ -73,6 +77,7 @@ static int vid_read_header(AVFormatContext *s)
     stream = avformat_new_stream(s, NULL);
     if (!stream)
         return AVERROR(ENOMEM);
+    vid->video_index = stream->index;
     stream->start_time = 0;
     avpriv_set_pts_info(stream, 32, 1, 60);     // 16 ms increments, i.e. 60 fps
     stream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
@@ -83,16 +88,9 @@ static int vid_read_header(AVFormatContext *s)
     vid->bethsoft_global_delay = avio_rl16(pb);
     avio_rl16(pb);
 
-    // done with video codec, set up audio codec
-    stream = avformat_new_stream(s, NULL);
-    if (!stream)
-        return AVERROR(ENOMEM);
-    stream->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    stream->codec->codec_id = CODEC_ID_PCM_U8;
-    stream->codec->channels = 1;
-    stream->codec->sample_rate = 11025;
-    stream->codec->bits_per_coded_sample = 8;
-    stream->codec->bit_rate = stream->codec->channels * stream->codec->sample_rate * stream->codec->bits_per_coded_sample;
+    // wait until the first audio packet to create the audio stream
+    vid->audio_index = -1;
+    s->ctx_flags |= AVFMTCTX_NOHEADER;
 
     return 0;
 }
@@ -168,7 +166,7 @@ static int read_frame(BVID_DemuxContext *vid, AVIOContext *pb, AVPacket *pkt,
     av_free(vidbuf_start);
 
     pkt->pos = position;
-    pkt->stream_index = 0;  // use the video decoder, which was initialized as the first stream
+    pkt->stream_index = vid->video_index;
     pkt->duration = duration;
     if (block_type == VIDEO_I_FRAME)
         pkt->flags |= AV_PKT_FLAG_KEY;
@@ -219,9 +217,22 @@ static int vid_read_packet(AVFormatContext *s,
         case FIRST_AUDIO_BLOCK:
             avio_rl16(pb);
             // soundblaster DAC used for sample rate, as on specification page (link above)
-            s->streams[1]->codec->sample_rate = 1000000 / (256 - avio_r8(pb));
-            s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_coded_sample;
+            vid->sample_rate = 1000000 / (256 - avio_r8(pb));
         case AUDIO_BLOCK:
+            if (vid->audio_index < 0) {
+                AVStream *st = avformat_new_stream(s, NULL);
+                if (!st)
+                    return AVERROR(ENOMEM);
+                vid->audio_index                 = st->index;
+                st->codec->codec_type            = AVMEDIA_TYPE_AUDIO;
+                st->codec->codec_id              = CODEC_ID_PCM_U8;
+                st->codec->channels              = 1;
+                st->codec->bits_per_coded_sample = 8;
+                st->codec->sample_rate           = vid->sample_rate;
+                st->codec->bit_rate              = 8 * st->codec->sample_rate;
+                st->start_time                   = 0;
+                avpriv_set_pts_info(st, 64, 1, vid->sample_rate);
+            }
             audio_length = avio_rl16(pb);
             if ((ret_value = av_get_packet(pb, pkt, audio_length)) != audio_length) {
                 if (ret_value < 0)
@@ -229,7 +240,8 @@ static int vid_read_packet(AVFormatContext *s,
                 av_log(s, AV_LOG_ERROR, "incomplete audio block\n");
                 return AVERROR(EIO);
             }
-            pkt->stream_index = 1;
+            pkt->stream_index = vid->audio_index;
+            pkt->duration     = audio_length;
             pkt->flags |= AV_PKT_FLAG_KEY;
             return 0;