Simplify AVCodecContext usage within FFplay audio_decode_frame()
Originally committed as revision 14503 to svn://svn.ffmpeg.org/ffmpeg/trunk
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a78d882d76
commit
abdff64682
9
ffplay.c
9
ffplay.c
@ -1571,6 +1571,7 @@ static int synchronize_audio(VideoState *is, short *samples,
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static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, double *pts_ptr)
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{
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AVPacket *pkt = &is->audio_pkt;
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AVCodecContext *dec= is->audio_st->codec;
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int n, len1, data_size;
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double pts;
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@ -1578,7 +1579,7 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
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/* NOTE: the audio packet can contain several frames */
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while (is->audio_pkt_size > 0) {
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data_size = buf_size;
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len1 = avcodec_decode_audio2(is->audio_st->codec,
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len1 = avcodec_decode_audio2(dec,
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(int16_t *)audio_buf, &data_size,
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is->audio_pkt_data, is->audio_pkt_size);
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if (len1 < 0) {
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@ -1594,9 +1595,9 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
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/* if no pts, then compute it */
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pts = is->audio_clock;
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*pts_ptr = pts;
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n = 2 * is->audio_st->codec->channels;
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n = 2 * dec->channels;
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is->audio_clock += (double)data_size /
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(double)(n * is->audio_st->codec->sample_rate);
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(double)(n * dec->sample_rate);
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#if defined(DEBUG_SYNC)
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{
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static double last_clock;
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@ -1621,7 +1622,7 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
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if (packet_queue_get(&is->audioq, pkt, 1) < 0)
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return -1;
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if(pkt->data == flush_pkt.data){
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avcodec_flush_buffers(is->audio_st->codec);
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avcodec_flush_buffers(dec);
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continue;
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}
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