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ffplay: handle audio buffersink output properly with buffering filters

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Fixes cases when the audio filter generates less or more frames than the input.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2013-04-01 20:26:22 +02:00
parent 325846aac0
commit 0b24e341ed
1 changed files with 18 additions and 8 deletions
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26
ffplay.c
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@ -188,6 +188,7 @@ typedef struct VideoState {
unsigned int audio_buf1_size;
int audio_buf_index; /* in bytes */
int audio_write_buf_size;
int audio_buf_frames_pending;
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
int audio_pkt_temp_serial;
@ -2153,10 +2154,12 @@ static int audio_decode_frame(VideoState *is)
int flush_complete = 0;
int wanted_nb_samples;
AVRational tb;
int ret;
int reconfigure;
for (;;) {
/* NOTE: the audio packet can contain several frames */
while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet) || is->audio_buf_frames_pending) {
if (!is->frame) {
if (!(is->frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
@ -2171,6 +2174,8 @@ static int audio_decode_frame(VideoState *is)
if (is->paused)
return -1;
if (!is->audio_buf_frames_pending) {
if (flush_complete)
break;
new_packet = 0;
@ -2200,10 +2205,6 @@ static int audio_decode_frame(VideoState *is)
pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
#if CONFIG_AVFILTER
{
int ret;
int reconfigure;
dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
reconfigure =
@ -2235,10 +2236,18 @@ static int audio_decode_frame(VideoState *is)
if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
return ret;
av_frame_unref(is->frame);
if ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, is->frame, 0)) < 0)
return ret;
tb = is->out_audio_filter->inputs[0]->time_base;
#endif
}
#if CONFIG_AVFILTER
if ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, is->frame, 0)) < 0) {
if (ret == AVERROR(EAGAIN)) {
is->audio_buf_frames_pending = 0;
continue;
}
return ret;
}
is->audio_buf_frames_pending = 1;
tb = is->out_audio_filter->inputs[0]->time_base;
#endif
data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(is->frame),
@ -2340,6 +2349,7 @@ static int audio_decode_frame(VideoState *is)
if (pkt->data == flush_pkt.data) {
avcodec_flush_buffers(dec);
flush_complete = 0;
is->audio_buf_frames_pending = 0;
}
*pkt_temp = *pkt;