Commit 773ff823 authored by Justin Ruggles's avatar 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.
parent 9546f331
......@@ -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;
......
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