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FFmpeg encoding aac audio, encoded file can not be played
9 décembre 2020, par cs guyI am trying to encodea aac audio. Example of MP2 here. I followed this documentation. I encode the audio with the code below by calling
startEncodingand after 2 seconds I callstopEncoding. Everything seems to work fine I get an file with some size but the problem is I can not open or play it. I dont know why. I must be doing something wrong in the code.

header :


class MediaEncoder {
public:
 MediaEncoder(char *filePath);

 void startEncoding();
 void stopEncoding();

 void encode(AVFrame *frame);
 bool isEncoding() const;

 void startEncoderWorker();

 int32_t check_sample_fmt(enum AVSampleFormat sample_fmt);
 
 bool signalExitFuture = false;
 int32_t ret;

private:
 std::future<void> encodingFuture;
 AVCodecContext *avCodecContext;
 AVFrame *avFrame;
 AVPacket *avPacket;
 AVCodec *codec;
 FILE* file;
};
</void>

cpp :


MediaEncoder::MediaEncoder(char *filePath){
 buffer = std::make_unique>(recorderBufferSize);

 /* find the encoder */
 codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
 avCodecContext = avcodec_alloc_context3(codec);

 avCodecContext->bit_rate = 64000;

 /* check that the encoder supports given sample format */
 avCodecContext->sample_fmt = AVSampleFormat::AV_SAMPLE_FMT_FLTP;
 
 /* select other audio parameters supported by the encoder */
 avCodecContext->sample_rate = defaultSampleRate;
 avCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
 avCodecContext->channels = av_get_channel_layout_nb_channels(avCodecContext->channel_layout);

 /* open it */
 avcodec_open2(avCodecContext, codec, nullptr)
 
 file = fopen(filePath, "wb");
 
 /* packet for holding encoded output */
 avPacket = av_packet_alloc();
 
 /* frame containing input raw audio */
 avFrame = av_frame_alloc();
 
 avFrame->nb_samples = avCodecContext->frame_size;
 avFrame->format = avCodecContext->sample_fmt;
 avFrame->channel_layout = avCodecContext->channel_layout;

 /* allocate the data buffers */
 av_frame_get_buffer(avFrame, 0);
}


void MediaEncoder::startEncoding() {
 // set flags for decoding thread
 signalExitFuture = false;

 encodingFuture = std::async(std::launch::async, &MediaEncoder::startEncoderWorker, this);
}

void MediaEncoder::stopEncoding() {
 signalExitFuture = true;
}

bool MediaEncoder::isEncoding() const {
 return !signalExitFuture;
}

void MediaEncoder::encode(AVFrame *frame) {
 /* send the frame for encoding */
 ret = avcodec_send_frame(avCodecContext, frame);
 if (ret < 0) {
 LOGE("Error sending the frame to the encoder %s",
 av_err2str(ret));
 *recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
 return;
 }

 /* read all the available output packets (in general there may be any
 * number of them */
 while (ret >= 0) {
 ret = avcodec_receive_packet(avCodecContext, avPacket);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
 return;
 } else if (ret < 0) {
 LOGE("Error encoding audio frame %s",
 av_err2str(ret));
 *recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
 return;
 }
 
 /* Solution begins here
 int aac_profile = 2; // AAC LC
 int frequencey_index = 4; // 44,100Hz
 int channel_configuration = 2; // stereo (left, right)

 int frame_length = avPacket->size + 7; // your frame length
 
 unsigned char adts_header[7];

 // fill in ADTS data
 adts_header[0] = (unsigned char) 0xFF;
 adts_header[1] = (unsigned char) 0xF9;
 adts_header[2] = (unsigned char) (((aac_profile -1) << 6 ) + (frequencey_index << 2) + (channel_configuration >> 2));
 adts_header[3] = (unsigned char) (((channel_configuration & 3) << 6) + (frame_length >> 11));
 adts_header[4] = (unsigned char) ((frame_length & 0x7FF) >> 3);
 adts_header[5] = (unsigned char) (((frame_length & 7) << 5) + 0x1F);
 adts_header[6] = (unsigned char) 0xFC;

 fwrite(adts_header, 1, 7, file); 
 Solution ends here */ 
 fwrite(avPacket->data, 1, avPacket->size, file);
 av_packet_unref(avPacket);
 }
}

void MediaEncoder::startEncoderWorker() {
 try {
 float *leftChannel;
 float *rightChannel;
 float val;

 while (!signalExitFuture) {
 ret = av_frame_make_writable(avFrame);
 if (ret < 0) {
 LOGE("av_frame_make_writable Can not Ensure that the frame data is writable. %s",
 av_err2str(ret));
 *recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
 return;
 }

 leftChannel = (float *) avFrame->data[0];
 rightChannel = (float *) avFrame->data[1];

 for (int32_t i = 0; i < avCodecContext->frame_size; ++i) {
 
 leftChannel[i] = 0.4;
 rightChannel[i] = 0.4;
 }

 encode(avFrame);
 }

 /* flush the encoder */
 encode(nullptr);

 fclose(file);
 LOGE("Encoding finished!");

 av_frame_free(&avFrame);
 av_packet_free(&avPacket);
 avcodec_free_context(&avCodecContext);
 } catch (std::exception &e) {
 LOGE("startEncoderWorker uncaught exception %s", e.what());
 }

 LOGE("Deleting Media Encoder!");

}


Here is an recorded 11 seconds of an aac file recorded with real float pcm data rather than 0.4 as it is in the code I posted. Google Drive Link


The code above works. Thanks to the legend @Markus-Schumann


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Revision 30148 : On pousse encore un peu plus loin la conf (toujours pas utilisable dans ...
24 juillet 2009, par kent1@… — LogOn pousse encore un peu plus loin la conf (toujours pas utilisable dans l’état)
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Revision 30775 : Installation et désinstallation plus propre
10 août 2009, par kent1@… — LogInstallation et désinstallation plus propre
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