PCM to AAC with FAAC, numberBytesWritten >> 16bits?

Next iteration of my question:

Thank you for your inputs, it has helped me to understa开发者_C百科nd a little bit more about the Frame and inputSamples utility. I’ve done modifications to my source code with the new knowledge you’ve given me. But I still have problems, so I might not have understood fully what you meant. Here is my OpenFile function, sorry for the name but I’ll refactor later; when it’ll work =)

//-----------------------------------------------------------------------------
/* 
This Function Open a File containing the Audio, Binary, Data.
*///___________________________________________________________________________
const short* OpenFile(const char* fileName, long& fileSize, WavFormat* wav)
{
// ouvre le fichier
ifstream file;
file.open((char*)fileName, ios::binary|ios::in);

if (file.good())
{
    // Read the WAV's Header
    wav = CheckWavHeader(file, wav);

    cout << "chunkID: " << wav->chunkID <<'\n';
    cout << "chunkSize: " << wav->chunkSize <<'\n';
    cout << "format: " << wav->format <<'\n';
    cout << "subChunk1ID: " << wav->subChunk1ID <<'\n';
    cout << "subChunk1Size: " << wav->subChunk1Size <<'\n';
    cout << "audioFormat: " << wav->audioFormat <<'\n'; // audioFormat == 1, alors PCM 16bits
    cout << "numChannels: " << wav->numChannels <<'\n';
    cout << "sampleRate: " << wav->sampleRate <<'\n';
    cout << "byteRate: " << wav->byteRate <<'\n';
    cout << "blockAlign: " << wav->blockAlign <<'\n';
    cout << "bitsPerSample: " << wav->bitsPerSample <<'\n';
    cout << "subChunk2ID: " << wav->subChunk2ID <<'\n';
    cout << "subChunk2Size: " << wav->subChunk2Size <<'\n';

    // Get the file’s size
    file.seekg(0L, ios::end);
    fileSize = ((long)file.tellg() - DATA_POS);

    file.seekg(DATA_POS, ios::beg); // back to the data.

    // Read the Data into the Buffer
    uint nbSamples = fileSize / sizeof(short);
    short* inputArray = new short[nbSamples];
    file.read((char*)inputArray, fileSize);

    // Close the file and return the Data
    file.close();
    return (const short*)inputArray;
}
else
{
    exit(-1);
}
}

I’m opening the file, checking its size, create a short buffer and read the wav’s data into the short buffer and finally I return it.

In the main, for now I commented the G711 decoder. When I run the application, the faacEncOpen gives me 2048 for inputSamples (it’s logic since I have 2 channels in the Wav’s file for a FRAME_LEN of 1024). So if I understood correctly, 1 Frame == 2048 samples for my application. So for each Frame I call the faacEncEncode, I give the tmpInputBuffer that is a buffer of the same size as inputSamples at the inputBuffer[i * inputSamples] index.

//-----------------------------------------------------------------------------
/*
The Main entry Point of the Application
*///_____________________________________________________________________________
int main()
{
// Get the File's Data
WavFormat* wav = new WavFormat;
long fileSize;
const short* fileInput = OpenFile("audioTest.wav", fileSize, wav);

// G711 mu-Law Decoder
//MuLawDecoder* decoder = new MuLawDecoder();
//short* inputBuffer = decoder->MuLawDecode_shortArray((byte*)fileInput, (int)nbChunk);

short* inputBuffer = (short*)fileInput;

// Info for FAAC
ulong sampleRate = wav->sampleRate;
uint numChannels = wav->numChannels;
ulong inputSamples;
ulong maxOutputBytes;

// Ouvre l'Encodeur et assigne la Configuration.
faacEncHandle hEncoder = faacEncOpen(sampleRate, numChannels, &inputSamples, &maxOutputBytes);
faacEncConfigurationPtr faacConfig = faacEncGetCurrentConfiguration(hEncoder);

faacConfig->inputFormat = FAAC_INPUT_16BIT;
faacConfig->bitRate = 64000;

int result = faacEncSetConfiguration(hEncoder, faacConfig);

/*Input Buffer and Output Buffer*/
byte* outputBuffer = new byte[maxOutputBytes];
int nbBytesWritten = 0;
Sink* sink = new Sink();

uint nbFrame = fileSize / inputSamples;
int32_t* tmpInputBuffer = new int32_t[inputSamples];

for (uint i = 0; i < nbFrame; i++)
{   
    strncpy((char*)tmpInputBuffer, (const char*)&inputBuffer[i * inputSamples], inputSamples);

    nbBytesWritten = faacEncEncode(hEncoder, tmpInputBuffer, inputSamples, outputBuffer, maxOutputBytes);

    cout << 100.0 * (float)i / nbFrame << "%\t nbBytesWritten = " << nbBytesWritten << "\n";

    if (nbBytesWritten > 0)
    {
        sink->AddAACStream(outputBuffer, nbBytesWritten);
    }
}

sink->WriteToFile("output.aac");

// Close AAC Encoder
faacEncClose(hEncoder);

// Supprimer tous les pointeurs
delete sink;
//delete decoder;
delete[] fileInput;
//delete[] inputBuffer;
delete[] outputBuffer;
delete[] tmpInputBuffer;

system("pause");

return 0;
}

When the output Data is Dumped into an .acc file (as RAW AAC), I use the application mp4muxer.exe to create an .mp4 file to listen to the final converted sound. But the sound is not good at all...

I'm wondering if there is something I'm not seeing or do not unserstand that I should.

Thank you in advance for your useful inputs.

Each call to faacEncEncode encodes inputSamples samples, not just one. Your main loop should read that many samples from the WAV file into the input buffer, then call faacEncEncode once for that buffer, and finally write the output buffer to the AAC file.

It's possible that I've misunderstood what you're doing (if so, it would be useful to know: (1) What's the OpenFile function you're calling, and does it (despite its name) actually read the file as well as opening it? (2) How is inputBuffer set up?) but:

faacEncEncode expects to be given a whole frame's worth of samples. A frame is the number of samples you got passed back in inputSamples when you called faacEncOpen. (You can give it less than a whole frame if you've reached the end of the input, of course.)

So you're getting 460 and 539 bytes for each of two frames -- not for 16 bits in each case. And it looks as if your input-data pointers are actually offset by only one sample each time, so you're handing it badly overlapping frames. (And the wrong number of them; nbChunk is not the number of frames you have.)