I am having some trouble encoding audio using LibAv*.
I am currently taking raw unformatted PCM from the mic in 1024 byte
chunks and trying to convert this to MP3 or AAC. The input format is
single channel, 8khz 16-bit PCM. I was hoping to keep the output sample
settings the same.
When encoding to MP3 I get large amounts of distortion but I can just
about make out my voice in the background. When encoding to AAC, FFProbe
states that there is invalid data in the input file.
Here is my audio encoding parameters and method:
...
encContext.bit_rate = 11025;
encContext.sample_rate = 8000;
encContext.channels = 1;
...
//I take the input byte array and marshal that to an array of short's
int AudioEncoder::EncodeAudio(array<short>^ input, array<unsigned
char>^% output)
{
//make sure the output is null
output = nullptr;
int outbuf_size = 100000;
uint8_t* outbuf = (uint8_t*)av_malloc(outbuf_size);
if (outbuf)
{
//convert the byte array to unmanaged
pin_ptr<short> unmanagedInput = &input[0];
int16_t* samples = (int16_t*)unmanagedInput;
//try and do the encoding
int out_size = avcodec_encode_audio(encContext, outbuf,
input->Length, samples);
if (out_size > 0)
{
//now copy the output back...
output = gcnew array<unsigned char>(out_size);
Marshal::Copy(IntPtr(outbuf), output, 0, output->Length);
}
av_free(samples);
samples = NULL;
av_free(outbuf);
outbuf = NULL;
return out_size;
}
return -1;
}
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