mirror of https://git.ffmpeg.org/ffmpeg.git
doc/examples/muxing: Always use swr, simplifies code slightly
Idea-from: 56f98e340f
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -241,7 +241,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
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c->sample_rate, ost->frame->nb_samples);
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/* create resampler context */
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if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
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ost->swr_ctx = swr_alloc();
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if (!ost->swr_ctx) {
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fprintf(stderr, "Could not allocate resampler context\n");
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@ -261,7 +260,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
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fprintf(stderr, "Failed to initialize the resampling context\n");
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exit(1);
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}
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}
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}
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/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
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@ -318,7 +316,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
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if (frame) {
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/* convert samples from native format to destination codec format, using the resampler */
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if (ost->swr_ctx) {
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/* compute destination number of samples */
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dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
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c->sample_rate, c->sample_rate, AV_ROUND_UP);
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@ -333,9 +330,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
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exit(1);
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}
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frame = ost->tmp_frame;
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} else {
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dst_nb_samples = frame->nb_samples;
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}
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frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
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ost->samples_count += dst_nb_samples;
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