examples/muxing: reindent after previous commit
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@ -265,41 +265,41 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
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c = st->codec;
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if (!flush) {
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get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
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get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
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/* convert samples from native format to destination codec format, using the resampler */
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if (swr_ctx) {
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/* compute destination number of samples */
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
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c->sample_rate, c->sample_rate, AV_ROUND_UP);
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if (dst_nb_samples > max_dst_nb_samples) {
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av_free(dst_samples_data[0]);
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ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
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dst_nb_samples, c->sample_fmt, 0);
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if (ret < 0)
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/* convert samples from native format to destination codec format, using the resampler */
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if (swr_ctx) {
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/* compute destination number of samples */
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
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c->sample_rate, c->sample_rate, AV_ROUND_UP);
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if (dst_nb_samples > max_dst_nb_samples) {
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av_free(dst_samples_data[0]);
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ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
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dst_nb_samples, c->sample_fmt, 0);
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if (ret < 0)
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exit(1);
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max_dst_nb_samples = dst_nb_samples;
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dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
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c->sample_fmt, 0);
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}
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/* convert to destination format */
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ret = swr_convert(swr_ctx,
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dst_samples_data, dst_nb_samples,
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(const uint8_t **)src_samples_data, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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exit(1);
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max_dst_nb_samples = dst_nb_samples;
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dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
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c->sample_fmt, 0);
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}
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} else {
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dst_nb_samples = src_nb_samples;
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}
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/* convert to destination format */
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ret = swr_convert(swr_ctx,
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dst_samples_data, dst_nb_samples,
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(const uint8_t **)src_samples_data, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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exit(1);
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}
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} else {
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dst_nb_samples = src_nb_samples;
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}
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audio_frame->nb_samples = dst_nb_samples;
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audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
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avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
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dst_samples_data[0], dst_samples_size, 0);
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samples_count += dst_nb_samples;
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audio_frame->nb_samples = dst_nb_samples;
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audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
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avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
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dst_samples_data[0], dst_samples_size, 0);
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samples_count += dst_nb_samples;
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}
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ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
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