RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity

ffplay: use audio parameters from the decoded frame instead of AVCodecContext 

Based on commit by Justin Ruggles (the changed code is too different to apply as is)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-10-13 16:45:01 +02:00
parent 15ef1cfe64
commit d0707677fa
1 changed files with 16 additions and 16 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

32
ffplay.c
View File

@ -1974,34 +1974,34 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
flush_complete = 1;
continue;
}
data_size = av_samples_get_buffer_size(NULL, dec->channels,
data_size = av_samples_get_buffer_size(NULL, is->frame->channels,
is->frame->nb_samples,
dec->sample_fmt, 1);
is->frame->format, 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);
(is->frame->channel_layout && is->frame->channels == av_get_channel_layout_nb_channels(is->frame->channel_layout)) ?
is->frame->channel_layout : av_get_default_channel_layout(is->frame->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 ||
(wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) {
if (is->frame->format != is->audio_src.fmt ||
dec_channel_layout != is->audio_src.channel_layout ||
is->frame->sample_rate != is->audio_src.freq ||
(wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) {
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
dec_channel_layout, dec->sample_fmt, dec->sample_rate,
dec_channel_layout, is->frame->format, is->frame->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels,
is->frame->sample_rate, av_get_sample_fmt_name(is->frame->format), (int)is->frame->channels,
is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
break;
}
is->audio_src.channel_layout = dec_channel_layout;
is->audio_src.channels = dec->channels;
is->audio_src.freq = dec->sample_rate;
is->audio_src.fmt = dec->sample_fmt;
is->audio_src.channels = is->frame->channels;
is->audio_src.freq = is->frame->sample_rate;
is->audio_src.fmt = is->frame->format;
}
if (is->swr_ctx) {
@ -2009,8 +2009,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
uint8_t *out[] = {is->audio_buf2};
int out_count = sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt);
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) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / is->frame->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate) < 0) {
fprintf(stderr, "swr_set_compensation() failed\n");
break;
}
@ -2035,7 +2035,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
pts = is->audio_clock;
*pts_ptr = pts;
is->audio_clock += (double)data_size /
(dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt));
(is->frame->channels * is->frame->sample_rate * av_get_bytes_per_sample(is->frame->format));
#ifdef DEBUG
{
static double last_clock;