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ffplay: use swr_set_compensation for audio synchronization

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Also change synchronize_audio to only calculate the wanted number of samples
instead of doing the actual synchronization, and make swr_convert handle the
sample addition or deletion.

This new method replaces the old buggy way which simply deleted or
multiplied samples.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2011-12-29 21:05:38 +01:00
parent 5387f9917f
commit 6f06545b26
1 changed files with 26 additions and 47 deletions
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73
ffplay.c
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@ -1970,25 +1970,19 @@ static void update_sample_display(VideoState *is, short *samples, int samples_si
}
}
/* return the new audio buffer size (samples can be added or deleted
to get better sync if video or external master clock) */
static int synchronize_audio(VideoState *is, short *samples,
int samples_size1, double pts)
/* return the wanted number of samples to get better sync if sync_type is video
* or external master clock */
static int synchronize_audio(VideoState *is, int nb_samples)
{
int n, samples_size;
double ref_clock;
n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels;
samples_size = samples_size1;
int wanted_nb_samples = nb_samples;
/* if not master, then we try to remove or add samples to correct the clock */
if (((is->av_sync_type == AV_SYNC_VIDEO_MASTER && is->video_st) ||
is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) {
double diff, avg_diff;
int wanted_size, min_size, max_size, nb_samples;
int min_nb_samples, max_nb_samples;
ref_clock = get_master_clock(is);
diff = get_audio_clock(is) - ref_clock;
diff = get_audio_clock(is) - get_master_clock(is);
if (diff < AV_NOSYNC_THRESHOLD) {
is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum;
@ -2000,38 +1994,13 @@ static int synchronize_audio(VideoState *is, short *samples,
avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
if (fabs(avg_diff) >= is->audio_diff_threshold) {
wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n);
nb_samples = samples_size / n;
min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
if (wanted_size < min_size)
wanted_size = min_size;
else if (wanted_size > FFMIN3(max_size, samples_size, sizeof(is->audio_buf2)))
wanted_size = FFMIN3(max_size, samples_size, sizeof(is->audio_buf2));
/* add or remove samples to correction the synchro */
if (wanted_size < samples_size) {
/* remove samples */
samples_size = wanted_size;
} else if (wanted_size > samples_size) {
uint8_t *samples_end, *q;
int nb;
/* add samples */
nb = (samples_size - wanted_size);
samples_end = (uint8_t *)samples + samples_size - n;
q = samples_end + n;
while (nb > 0) {
memcpy(q, samples_end, n);
q += n;
nb -= n;
}
samples_size = wanted_size;
}
wanted_nb_samples = nb_samples + (int)(diff * is->audio_src_freq);
min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
}
av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n",
diff, avg_diff, samples_size - samples_size1,
diff, avg_diff, wanted_nb_samples - nb_samples,
is->audio_clock, is->video_clock, is->audio_diff_threshold);
}
} else {
@ -2042,7 +2011,7 @@ static int synchronize_audio(VideoState *is, short *samples,
}
}
return samples_size;
return wanted_nb_samples;
}
/* decode one audio frame and returns its uncompressed size */
@ -2057,6 +2026,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
double pts;
int new_packet = 0;
int flush_complete = 0;
int wanted_nb_samples;
for (;;) {
/* NOTE: the audio packet can contain several frames */
@ -2091,8 +2061,12 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
dec->sample_fmt, 1);
dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
wanted_nb_samples = synchronize_audio(is, is->frame->nb_samples);
if (dec->sample_fmt != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || dec->sample_rate != is->audio_src_freq) {
if (dec->sample_fmt != is->audio_src_fmt ||
dec_channel_layout != is->audio_src_channel_layout ||
dec->sample_rate != is->audio_src_freq ||
(wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) {
if (is->swr_ctx)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
@ -2119,8 +2093,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (is->swr_ctx) {
const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2};
if (wanted_nb_samples != is->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt_freq / dec->sample_rate,
wanted_nb_samples * is->audio_tgt_freq / dec->sample_rate) < 0) {
fprintf(stderr, "swr_set_compensation() failed\n");
break;
}
}
len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
in, is->frame->nb_samples);
if (len2 < 0) {
fprintf(stderr, "audio_resample() failed\n");
break;
@ -2196,8 +2177,6 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
audio_size = synchronize_audio(is, (int16_t *)is->audio_buf, audio_size,
pts);
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;