Dana 14. 7. 2015. 20:49 osoba "Ruurd Adema" <[email protected]> napisala je: > > I'm trying to write live incoming audiopackets into a mov file with AAC encoding using the FFmpeg api. > > When using no encoding (AV_CODEC_ID_PCM_S16LE) it works well, when using AAC encoding (AV_CODEC_ID_AAC) it fails. The resulting audiofile plays too fast and sounds distorted. > > I’m new to the FFmpeg api, (and quite a beginner in programming anyway), so big chance I forgot something or doing something wrong. Is there anyone willing to help me with this one? > > audiopacket_sample_count = audiopacket->GetSampleFrameCount(); > audiopacket_channel_count = decklink_config()->audio_channel_count; > audiopacket_size = audiopacket_sample_count * (decklink_config()->audio_sampletype/8) * audiopacket_channel_count; > > audiopacket->GetBytes(&audiopacket_data); > > av_init_packet(&pkt); > > if (AUDIO_TYPE == AV_CODEC_ID_PCM_S16LE) > { > audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den); > > pkt.pts = audio_pts; > pkt.dts = pkt.pts; > pkt.flags |= AV_PKT_FLAG_KEY; > pkt.stream_index = audio_stream->index; > pkt.data = (uint8_t *)audiopacket_data; > pkt.size = audiopacket_size; > > av_interleaved_write_frame(output_fmt_ctx, &pkt); > } > else if (AUDIO_TYPE == AV_CODEC_ID_AAC) > { > frame = av_frame_alloc(); > frame->format = audio_stream->codec->sample_fmt; > frame->channel_layout = audio_stream->codec->channel_layout; > frame->sample_rate = audio_stream->codec->sample_rate; > frame->nb_samples = audiopacket_sample_count; > > audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den); > > frame->pts = audio_pts; > > if (avcodec_fill_audio_frame(frame, audiopacket_channel_count, audio_stream->codec->sample_fmt, (const uint8_t *)audiopacket_data, audiopacket_size, 0) < 0) > { > fprintf(stderr, "[ERROR] Filling audioframe failed!\n"); > exit(-1); > } > > if (avcodec_encode_audio2(audio_stream->codec, &pkt, frame, &got_packet) != 0) > { > fprintf(stderr, "[ERROR] Encoding audio failed\n"); > } > > if (got_packet) > { > pkt.stream_index = audio_stream->index; > pkt.flags |= AV_PKT_FLAG_KEY; > > av_interleaved_write_frame(output_fmt_ctx, &pkt); > } > av_frame_free(&frame); > } > av_free_packet(&pkt); > > > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user >
Do you send exact same number of samples that aac encoder request? You need to buffer samples....
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