This avoids initializing a stream with dummy values or when the file does not
contain audio.
---
libavformat/bethsoftvid.c | 39 +++++++++++++++++++++++----------------
1 files changed, 23 insertions(+), 16 deletions(-)
diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c
index 6a73f62..dc48d7c 100644
--- a/libavformat/bethsoftvid.c
+++ b/libavformat/bethsoftvid.c
@@ -43,7 +43,9 @@ typedef struct BVID_DemuxContext
int bethsoft_global_delay;
int video_pts; /**< video time stamp */
+ int video_index; /**< video stream index */
int audio_pts; /**< audio time stamp */
+ int audio_index; /**< audio stream index */
uint8_t *palette;
@@ -77,6 +79,7 @@ static int vid_read_header(AVFormatContext *s,
stream = avformat_new_stream(s, NULL);
if (!stream)
return AVERROR(ENOMEM);
+ vid->video_index = stream->index;
avpriv_set_pts_info(stream, 32, 1, 60); // 16 ms increments, i.e. 60
fps
stream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
stream->codec->codec_id = CODEC_ID_BETHSOFTVID;
@@ -86,16 +89,9 @@ 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;
}
@@ -172,7 +168,7 @@ 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;
pkt->pts = vid->video_pts;
vid->video_pts += duration;
@@ -200,7 +196,7 @@ static int vid_read_packet(AVFormatContext *s,
BVID_DemuxContext *vid = s->priv_data;
AVIOContext *pb = s->pb;
unsigned char block_type;
- int audio_length;
+ int audio_length, sample_rate = 11111;
int ret_value;
if(vid->is_finished || pb->eof_reached)
@@ -225,10 +221,21 @@ 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;
- avpriv_set_pts_info(s->streams[1], 64, 1,
s->streams[1]->codec->sample_rate);
+ 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 = sample_rate;
+ st->codec->bit_rate = 8 * st->codec->sample_rate;
+ avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+ }
audio_length = avio_rl16(pb);
if ((ret_value = av_get_packet(pb, pkt, audio_length)) !=
audio_length) {
if (ret_value < 0)
@@ -236,7 +243,7 @@ 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->pts = vid->audio_pts;
vid->audio_pts += audio_length;
--
1.7.1
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