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Merge remote-tracking branch 'cus/stable' 

* cus/stable:
  ffplay: silence buffer size must be a multiple of frame size
  ffplay: use swr_set_compensation for audio synchronization

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-01-14 16:40:49 +01:00
parent 4057505e28 1348420b8a
commit 0b4f1e0d3d
1 changed files with 28 additions and 48 deletions
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76
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 /* return the wanted number of samples to get better sync if sync_type is video
to get better sync if video or external master clock) */ * or external master clock */
static int synchronize_audio(VideoState *is, short *samples, static int synchronize_audio(VideoState *is, int nb_samples)
int samples_size1, double pts)
{ {
int n, samples_size; int wanted_nb_samples = nb_samples;
double ref_clock;
n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels;
samples_size = samples_size1;
/* if not master, then we try to remove or add samples to correct the clock */ /* 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) || if (((is->av_sync_type == AV_SYNC_VIDEO_MASTER && is->video_st) ||
is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) { is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) {
double diff, avg_diff; 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) - get_master_clock(is);
diff = get_audio_clock(is) - ref_clock;
if (diff < AV_NOSYNC_THRESHOLD) { if (diff < AV_NOSYNC_THRESHOLD) {
is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum; 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); avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
if (fabs(avg_diff) >= is->audio_diff_threshold) { if (fabs(avg_diff) >= is->audio_diff_threshold) {
wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n); wanted_nb_samples = nb_samples + (int)(diff * is->audio_src_freq);
nb_samples = samples_size / n; min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
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;
}
} }
av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n", 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); is->audio_clock, is->video_clock, is->audio_diff_threshold);
} }
} else { } 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 */ /* 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; double pts;
int new_packet = 0; int new_packet = 0;
int flush_complete = 0; int flush_complete = 0;
int wanted_nb_samples;
for (;;) { for (;;) {
/* NOTE: the audio packet can contain several frames */ /* 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->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); 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) if (is->swr_ctx)
swr_free(&is->swr_ctx); swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL, 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) { if (is->swr_ctx) {
const uint8_t *in[] = { is->frame->data[0] }; const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2}; 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), 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) { if (len2 < 0) {
fprintf(stderr, "audio_resample() failed\n"); fprintf(stderr, "audio_resample() failed\n");
break; break;
@ -2182,6 +2163,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
VideoState *is = opaque; VideoState *is = opaque;
int audio_size, len1; int audio_size, len1;
int bytes_per_sec; int bytes_per_sec;
int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt_channels, 1, is->audio_tgt_fmt, 1);
double pts; double pts;
audio_callback_time = av_gettime(); audio_callback_time = av_gettime();
@ -2192,12 +2174,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
if (audio_size < 0) { if (audio_size < 0) {
/* if error, just output silence */ /* if error, just output silence */
is->audio_buf = is->silence_buf; is->audio_buf = is->silence_buf;
is->audio_buf_size = sizeof(is->silence_buf); is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size;
} else { } else {
if (is->show_mode != SHOW_MODE_VIDEO) if (is->show_mode != SHOW_MODE_VIDEO)
update_sample_display(is, (int16_t *)is->audio_buf, audio_size); 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_size = audio_size;
} }
is->audio_buf_index = 0; is->audio_buf_index = 0;