Reputation: 3263
I want to decode a stream of AAC frames continuously, one frame at a time.
I went through the ffmpeg examples (The correct answer doesn't need to make use of ffmpeg necessarily), and I only found examples using complete AAC files and batch algorithms. But I want to decode a continuous AAC stream. How can I do this?
UPDATE: Following the comments and Decode AAC to PCM with ffmpeg on android , I was able to decode to PCM using ffmpeg, however the output is very metallic and noisy. What am I doing wrong here when calling this method for each AAC frame:
...
/*loop that receives frame in buffer*/
while(1){
/*receive frame*/
input = receive_one_buffer();
/*decode frame*/
decodeBuffer(input,strlen(input),Outfile);
}
...
/*decode frame*/
void decodeBuffer(char * input, int numBytes, ofstream& Outfile) {
/*"input" contains one AAC-LC frame*/
//copy bytes from buffer
uint8_t inputBytes[numBytes + FF_INPUT_BUFFER_PADDING_SIZE];
memset(inputBytes, 0, numBytes + FF_INPUT_BUFFER_PADDING_SIZE);
memcpy(inputBytes, input, numBytes);
av_register_all();
AVCodec *codec = avcodec_find_decoder(CODEC_ID_AAC);
AVCodecContext *avCtx = avcodec_alloc_context();
avCtx->channels = 1;
avCtx->sample_rate = 44100;
//the input buffer
AVPacket avPacket;
av_init_packet(&avPacket);
avPacket.size = numBytes; //input buffer size
avPacket.data = inputBytes; // the input buffer
int outSize;
int len;
uint8_t *outbuf = static_cast<uint8_t *>(malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE));
while (avPacket.size > 0) {
outSize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio3(avCtx, (short *) outbuf, &outSize,
&avPacket);
Outfile.write((char*)outbuf, outSize);
avPacket.size -= len;
avPacket.data += len;
}
av_free_packet(&avPacket);
avcodec_close(avCtx);
//av_free(avCtx);
return;
}
Upvotes: 3
Views: 8955
Reputation: 8244
You have to keep the decoder alive between subsequent decode calls. The AAC decoder must decode the previous buffer to be correctly "primed".
Please check for details:
https://developer.apple.com/library/mac/technotes/tn2258/_index.html
The following code assumes that the "ReceiveBuffer" function returns exactly one complete AAC access unit.
(BTW: you can't use strlen on a binary buffer; you'll get the distance to the first zero and not the buffer length)
#include <iostream>
#include <fstream>
#include "libavcodec\avcodec.h"
#include "libavformat\avformat.h"
#include "libavdevice\avdevice.h"
#include "libavfilter\avfilter.h"
AVCodecContext * CreateContext()
{
av_register_all();
AVCodec *codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
AVCodecContext *avCtx = avcodec_alloc_context3(codec);
return avCtx;
}
int32_t DecodeBuffer
(
std::ostream & output,
uint8_t * pInput,
uint32_t cbInputSize,
AVCodecContext * pAVContext
)
{
int32_t cbDecoded = 0;
//the input buffer
AVPacket avPacket;
av_init_packet(&avPacket);
avPacket.size = cbInputSize; //input buffer size
avPacket.data = pInput; // the input bufferra
AVFrame * pDecodedFrame = av_frame_alloc();
int nGotFrame = 0;
cbDecoded = avcodec_decode_audio4( pAVContext,
pDecodedFrame,
& nGotFrame,
& avPacket);
int data_size = av_samples_get_buffer_size( NULL,
pAVContext->channels,
pDecodedFrame->nb_samples,
pAVContext->sample_fmt,
1);
output.write((const char*)pDecodedFrame->data[0],data_size);
av_frame_free(&pDecodedFrame);
return cbDecoded;
}
uint8_t * ReceiveBuffer( uint32_t * cbBufferSize)
{
// TODO implement
return NULL;
}
int main
(
int argc,
char *argv[]
)
{
int nResult = 0;
AVCodecContext * pAVContext = CreateContext();
std::ofstream myOutputFile("audio.pcm",std::ios::binary);
while(1)
{
uint32_t cbBufferSize = 0;
uint8_t *pCompressedAudio = ReceiveBuffer( &cbBufferSize);
if(cbBufferSize && pCompressedAudio)
{
DecodeBuffer( myOutputFile,
pCompressedAudio,
cbBufferSize,
pAVContext);
}
else
{
break;
}
}
avcodec_close(pAVContext);
av_free(pAVContext);
return nResult;
}
Upvotes: 4