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Merge remote-tracking branch 'cus/stable'

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* cus/stable:
  ffplay: assume 0 stream start time if start time is unset in duration check
  ffplay: do not update audio clock with old audio pts if current frame is AV_NOPTS_VALUE

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2013-07-10 00:57:00 +02:00
commit 9e214900ad
1 changed files with 11 additions and 6 deletions
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17
ffplay.c
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@ -2297,10 +2297,11 @@ static int audio_decode_frame(VideoState *is)
audio_clock0 = is->audio_clock; audio_clock0 = is->audio_clock;
/* update the audio clock with the pts */ /* update the audio clock with the pts */
if (is->frame->pts != AV_NOPTS_VALUE) { if (is->frame->pts != AV_NOPTS_VALUE)
is->audio_clock = is->frame->pts * av_q2d(tb) + (double) is->frame->nb_samples / is->frame->sample_rate; is->audio_clock = is->frame->pts * av_q2d(tb) + (double) is->frame->nb_samples / is->frame->sample_rate;
is->audio_clock_serial = is->audio_pkt_temp_serial; else
} is->audio_clock = NAN;
is->audio_clock_serial = is->audio_pkt_temp_serial;
#ifdef DEBUG #ifdef DEBUG
{ {
static double last_clock; static double last_clock;
@ -2374,8 +2375,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt); bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index; is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
/* Let's assume the audio driver that is used by SDL has two periods. */ /* Let's assume the audio driver that is used by SDL has two periods. */
set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0); if (!isnan(is->audio_clock)) {
sync_clock_to_slave(&is->extclk, &is->audclk); set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
sync_clock_to_slave(&is->extclk, &is->audclk);
}
} }
static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params) static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params)
@ -2682,6 +2685,7 @@ static int read_thread(void *arg)
int st_index[AVMEDIA_TYPE_NB]; int st_index[AVMEDIA_TYPE_NB];
AVPacket pkt1, *pkt = &pkt1; AVPacket pkt1, *pkt = &pkt1;
int eof = 0; int eof = 0;
int64_t stream_start_time;
int pkt_in_play_range = 0; int pkt_in_play_range = 0;
AVDictionaryEntry *t; AVDictionaryEntry *t;
AVDictionary **opts; AVDictionary **opts;
@ -2923,8 +2927,9 @@ static int read_thread(void *arg)
continue; continue;
} }
/* check if packet is in play range specified by user, then queue, otherwise discard */ /* check if packet is in play range specified by user, then queue, otherwise discard */
stream_start_time = ic->streams[pkt->stream_index]->start_time;
pkt_in_play_range = duration == AV_NOPTS_VALUE || pkt_in_play_range = duration == AV_NOPTS_VALUE ||
(pkt->pts - ic->streams[pkt->stream_index]->start_time) * (pkt->pts - (stream_start_time != AV_NOPTS_VALUE ? stream_start_time : 0)) *
av_q2d(ic->streams[pkt->stream_index]->time_base) - av_q2d(ic->streams[pkt->stream_index]->time_base) -
(double)(start_time != AV_NOPTS_VALUE ? start_time : 0) / 1000000 (double)(start_time != AV_NOPTS_VALUE ? start_time : 0) / 1000000
<= ((double)duration / 1000000); <= ((double)duration / 1000000);