Reputation: 5178
I am receiving a byte array (int8_t*
) and I would like to use FFMPEG to encode it into FLAC. All the examples I found are reading data from files, which is not the case for me. Following the original documents (see here), I came up with the following solution:
#include "libavcodec/avcodec.h"
// ...
// params:
// audioData: original audio data
// len: length of the byte array (audio data)
// sampleRate: sample rate of the original audio data
// frameSize: frame size of the original data
uint8_t* encodeToFlac(uint8_t* audioData, int len, int sampleRate, int frameSize) {
uint8_t* convertedAudioData;
// Context information
AVCodecContext* context = avcodec_alloc_context();
context->bit_rate = 64000;
context->sample_rate = sampleRate;
context->channels = 2;
context->frame_size = frameSize;
short* samples = malloc(frameSize * 2 * context->channels);
int outAudioDataSize = len * 2;
convertedAudioData = malloc(outAudioDataSize);
int outSize = avcodec_encode_audio(c, convertedAudioData, outAudioDataSize, samples);
return convertedAudioData;
}
I have two main issues with the above solution:
I did not specify what the final encoding should be (for example, MP3, FLAC, etc), which makes me wonder if I'm using FFMPEG library correctly?
Do I have all the necessary information about the source - original audio data? I am not certain if I have all the necessary information to perform the encoding.
Upvotes: 3
Views: 1679
Reputation: 1988
You are nearly there, follow this example: https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/encode_audio.c
Answer to 1st question:
There you'll see codec = avcodec_find_encoder(AV_CODEC_ID_MP2)
.
In your case, you guessed it, it probably will be codec = avcodec_find_encoder(AV_CODEC_ID_FLAC)
and check/fix other values accordingly.
As for 2nd one... I'm sure you'll find out yourself, especially you must set this correctly (line 158) c->sample_fmt = AV_SAMPLE_FMT_S16
according to what your int8_t
array formatted.
Hope that helps.
Upvotes: 1