FFMPEG AAC encoding causes audio to be lower in pitch - ffmpeg

Question

I built a sample application that encodes AAC (from PortAudio) into a MP4 container (no video stream).
The resulting audio is lower in pitch.
#include "stdafx.h"
#include "TestRecording.h"
#include "libffmpeg.h"
TestRecording::TestRecording()
{
}
TestRecording::~TestRecording()
{
}
struct RecordingContext
{
RecordingContext()
{
formatContext = NULL;
audioStream = NULL;
audioFrame = NULL;
audioFrameframeNumber = 0;
}
libffmpeg::AVFormatContext* formatContext;
libffmpeg::AVStream* audioStream;
libffmpeg::AVFrame* audioFrame;
int audioFrameframeNumber;
};
static int AudioRecordCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData)
{
RecordingContext* recordingContext = (RecordingContext*)userData;
libffmpeg::avcodec_fill_audio_frame(recordingContext->audioFrame,
recordingContext->audioFrame->channels,
recordingContext->audioStream->codec->sample_fmt,
static_cast<const unsigned char*>(inputBuffer),
(framesPerBuffer * sizeof(float) * recordingContext->audioFrame->channels),
0);
libffmpeg::AVPacket pkt;
libffmpeg::av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
int gotpacket;
int result = avcodec_encode_audio2(recordingContext->audioStream->codec, &pkt, recordingContext->audioFrame, &gotpacket);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't encode the audio frame to acc");
return paContinue;
}
if (gotpacket)
{
pkt.stream_index = recordingContext->audioStream->index;
recordingContext->audioFrameframeNumber++;
// this codec requires no bitstream filter, just send it to the muxer!
result = libffmpeg::av_write_frame(recordingContext->formatContext, &pkt);
if (result < 0)
{
LOG(ERROR) << "Couldn't write the encoded audio frame";
libffmpeg::av_free_packet(&pkt);
return paContinue;
}
libffmpeg::av_free_packet(&pkt);
}
return paContinue;
}
static bool InitializeRecordingContext(RecordingContext* recordingContext)
{
int result = libffmpeg::avformat_alloc_output_context2(&recordingContext->formatContext, NULL, NULL, "C:\\Users\\Paul\\Desktop\\test.mp4");
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't create output format context");
return false;
}
libffmpeg::AVCodec *audioCodec;
audioCodec = libffmpeg::avcodec_find_encoder(libffmpeg::AV_CODEC_ID_AAC);
if (audioCodec == NULL)
{
LOG(ERROR) << "Couldn't find the encoder for AAC";
}
recordingContext->audioStream = libffmpeg::avformat_new_stream(recordingContext->formatContext, audioCodec);
if (!recordingContext->audioStream)
{
LOG(ERROR) << "Couldn't create the audio stream";
return false;
}
recordingContext->audioStream->codec->bit_rate = 64000;
recordingContext->audioStream->codec->sample_fmt = libffmpeg::AV_SAMPLE_FMT_FLTP;
recordingContext->audioStream->codec->sample_rate = 48000;
recordingContext->audioStream->codec->channel_layout = AV_CH_LAYOUT_STEREO;
recordingContext->audioStream->codec->channels = libffmpeg::av_get_channel_layout_nb_channels(recordingContext->audioStream->codec->channel_layout);
recordingContext->audioStream->codecpar->bit_rate = recordingContext->audioStream->codec->bit_rate;
recordingContext->audioStream->codecpar->format = recordingContext->audioStream->codec->sample_fmt;
recordingContext->audioStream->codecpar->sample_rate = recordingContext->audioStream->codec->sample_rate;
recordingContext->audioStream->codecpar->channel_layout = recordingContext->audioStream->codec->channel_layout;
recordingContext->audioStream->codecpar->channels = recordingContext->audioStream->codec->channels;
result = libffmpeg::avcodec_open2(recordingContext->audioStream->codec, audioCodec, NULL);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio codec");
return false;
}
// create a new frame to store the audio samples
recordingContext->audioFrame = libffmpeg::av_frame_alloc();
if (!recordingContext->audioFrame)
{
LOG(ERROR) << "Couldn't alloce the output audio frame";
return false;
}
recordingContext->audioFrame->nb_samples = recordingContext->audioStream->codec->frame_size;
recordingContext->audioFrame->channel_layout = recordingContext->audioStream->codec->channel_layout;
recordingContext->audioFrame->channels = recordingContext->audioStream->codec->channels;
recordingContext->audioFrame->format = recordingContext->audioStream->codec->sample_fmt;
recordingContext->audioFrame->sample_rate = recordingContext->audioStream->codec->sample_rate;
result = libffmpeg::av_frame_get_buffer(recordingContext->audioFrame, 0);
if (result < 0)
{
LOG(ERROR) << "Coudln't initialize the output audio frame buffer";
return false;
}
// some formats want video_stream headers to be separate
if (!strcmp(recordingContext->formatContext->oformat->name, "mp4") || !strcmp(recordingContext->formatContext->oformat->name, "mov") || !strcmp(recordingContext->formatContext->oformat->name, "3gp"))
{
recordingContext->audioStream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// open the ouput file
if (!(recordingContext->formatContext->oformat->flags & AVFMT_NOFILE))
{
result = libffmpeg::avio_open(&recordingContext->formatContext->pb, recordingContext->formatContext->filename, AVIO_FLAG_WRITE);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the output file");
return false;
}
}
// write the stream headers
result = libffmpeg::avformat_write_header(recordingContext->formatContext, NULL);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the headers to the file");
return false;
}
return true;
}
static bool FinalizeRecordingContext(RecordingContext* recordingContext)
{
int result = 0;
// write the trailing information
if (recordingContext->formatContext->pb)
{
result = libffmpeg::av_write_trailer(recordingContext->formatContext);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the trailer information");
return false;
}
}
// close all the codes
for (int i = 0; i < (int)recordingContext->formatContext->nb_streams; i++)
{
result = libffmpeg::avcodec_close(recordingContext->formatContext->streams[i]->codec);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the codec");
return false;
}
}
// close the output file
if (recordingContext->formatContext->pb)
{
if (!(recordingContext->formatContext->oformat->flags & AVFMT_NOFILE))
{
result = libffmpeg::avio_close(recordingContext->formatContext->pb);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the output file");
return false;
}
}
}
// free the format context and all of its data
libffmpeg::avformat_free_context(recordingContext->formatContext);
recordingContext->formatContext = NULL;
recordingContext->audioStream = NULL;
if (recordingContext->audioFrame)
{
libffmpeg::av_frame_free(&recordingContext->audioFrame);
recordingContext->audioFrame = NULL;
}
return true;
}
int TestRecording::Test()
{
PaError result = paNoError;
result = Pa_Initialize();
if (result != paNoError) LOGINT_WITH_MESSAGE(ERROR, result, "Error initializing audio device framework");
RecordingContext recordingContext;
if (!InitializeRecordingContext(&recordingContext))
{
LOG(ERROR) << "Couldn't start recording file";
return 0;
}
auto defaultDevice = Pa_GetDefaultInputDevice();
auto deviceInfo = Pa_GetDeviceInfo(defaultDevice);
PaStreamParameters inputParameters;
inputParameters.device = defaultDevice;
inputParameters.channelCount = 2;
inputParameters.sampleFormat = paFloat32;
inputParameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
PaStream* stream = NULL;
result = Pa_OpenStream(
&stream,
&inputParameters,
NULL,
48000,
1024,
paClipOff,
AudioRecordCallback,
&recordingContext);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio stream");
result = Pa_StartStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't start the audio stream");
Sleep(1000 * 5);
result = Pa_StopStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
if (!FinalizeRecordingContext(&recordingContext)) LOG(ERROR) << "Couldn't stop recording file";
result = Pa_CloseStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
return 0;
}
Here is the stdout, in case it helps.
https://gist.github.com/pauldotknopf/9f24a604ce1f8a081aa68da1bf169e98
Why is the audio lower in pitch? I assume I am overlooking a parameter that needs to be configured between PortAudio and FFMPEG. Is there something super obvious that I am missing?