提交 abdff646 编写于 作者: P Peter Ross

Simplify AVCodecContext usage within FFplay audio_decode_frame()

Originally committed as revision 14503 to svn://svn.ffmpeg.org/ffmpeg/trunk
上级 a78d882d
......@@ -1571,6 +1571,7 @@ static int synchronize_audio(VideoState *is, short *samples,
static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, double *pts_ptr)
{
AVPacket *pkt = &is->audio_pkt;
AVCodecContext *dec= is->audio_st->codec;
int n, len1, data_size;
double pts;
......@@ -1578,7 +1579,7 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
/* NOTE: the audio packet can contain several frames */
while (is->audio_pkt_size > 0) {
data_size = buf_size;
len1 = avcodec_decode_audio2(is->audio_st->codec,
len1 = avcodec_decode_audio2(dec,
(int16_t *)audio_buf, &data_size,
is->audio_pkt_data, is->audio_pkt_size);
if (len1 < 0) {
......@@ -1594,9 +1595,9 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
/* if no pts, then compute it */
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
n = 2 * dec->channels;
is->audio_clock += (double)data_size /
(double)(n * is->audio_st->codec->sample_rate);
(double)(n * dec->sample_rate);
#if defined(DEBUG_SYNC)
{
static double last_clock;
......@@ -1621,7 +1622,7 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
if (packet_queue_get(&is->audioq, pkt, 1) < 0)
return -1;
if(pkt->data == flush_pkt.data){
avcodec_flush_buffers(is->audio_st->codec);
avcodec_flush_buffers(dec);
continue;
}
......
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