doc/examples/muxing: Always use swr, simplifies code slightly

Idea-from: 56f98e340f
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2014-07-27 01:32:19 +02:00
parent 22e9fe06eb
commit fbd46e2f1c
1 changed files with 0 additions and 6 deletions

View File

@ -241,7 +241,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
c->sample_rate, ost->frame->nb_samples);
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
@ -261,7 +260,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@ -318,7 +316,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
if (ost->swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
@ -333,9 +330,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
exit(1);
}
frame = ost->tmp_frame;
} else {
dst_nb_samples = frame->nb_samples;
}
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;