static void write_audio_frame(AVFormatContext *oc, AVStream *st) { AVCodecContext *c; AVPacket pkt = { 0 }; // data and size must be 0; int got_packet, ret, dst_nb_samples; av_init_packet(&pkt); c = st->codec; get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels); /* convert samples from native format to destination codec format, using the resampler */ if (swr_ctx) { /* compute destination number of samples */ dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples, c->sample_rate, c->sample_rate, AV_ROUND_UP); if (dst_nb_samples > max_dst_nb_samples) { av_free(dst_samples_data[0]); ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels, dst_nb_samples, c->sample_fmt, 0); if (ret < 0) exit(1); max_dst_nb_samples = dst_nb_samples; dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples, c->sample_fmt, 0); } /* convert to destination format */ ret = swr_convert(swr_ctx, dst_samples_data, dst_nb_samples, (const uint8_t **)src_samples_data, src_nb_samples); if (ret < 0) { fprintf(stderr, "Error while converting\n"); exit(1); } } else { dst_nb_samples = src_nb_samples; } audio_frame->nb_samples = dst_nb_samples; audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt, dst_samples_data[0], dst_samples_size, 0); samples_count += dst_nb_samples; ret = avcodec_encode_audio2(c, &pkt, audio_frame, &got_packet); if (ret < 0) { fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret)); exit(1); } if (!got_packet) return; ret = write_frame(oc, &c->time_base, st, &pkt); if (ret != 0) { fprintf(stderr, "Error while writing audio frame: %s\n", av_err2str(ret)); exit(1); } }