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Wrap audio data of the pcm_alaw type into an MKA audio file using the ffmpeg API
19 septembre 2020, par bbddImagine that in my project, I receive
RTP
packets with the payload type-8, for later saving this load as the Nth part of the audio track. I extract this load from theRTP
packet and save it to a temporary buffer :

...

while ((rtp = receiveRtpPackets()).withoutErrors()) {
 payloadData.push(rtp.getPayloadData());
}

audioGenerator.setPayloadData(payloadData);
audioGenerator.recordToFile();

...



After filling a temporary buffer of a certain size with this payload, I process this buffer, namely, extract the entire payload and encode it using ffmpeg for further saving to an audio file in Matroska format. But I have a problem. Since the payload of the
RTP
packet istype 8
, I have to save the raw audio data of the pcm_alaw format tomka
audio format. But when saving raw datapcm_alaw
to an audio file, I get these messages from the library :

...

[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time

...



When you open an audio file in vlc, nothing is played (the audio track timestamp is missing).


The task of my project is to simply take pcm_alaw data and pack it in a container, in
mka
format. The best way to determine the codec is to use the av_guess_codec() function, which in turn automatically selects the desired codec ID. But how do I pack the raw data into the container correctly, I do not know.

It is important to note that I can get as raw data any format of this data (audio formats only) defined by the
RTP
packet type (All types ofRTP
packet payload). All I know is that in any case, I have to pack the audio data in anmka
container.

I also attach the code (borrowed from this resource) that I use :


audiogenerater.h


extern "C"
{
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
}

class AudioGenerater
{
public:

 AudioGenerater();
 ~AudioGenerater() = default;

 void generateAudioFileWithOptions(
 QString fileName,
 QByteArray pcmData,
 int channel,
 int bitRate,
 int sampleRate,
 AVSampleFormat format);
 
private:

 // init Format
 bool initFormat(QString audioFileName);

private:

 AVCodec *m_AudioCodec = nullptr;
 AVCodecContext *m_AudioCodecContext = nullptr;
 AVFormatContext *m_FormatContext = nullptr;
 AVOutputFormat *m_OutputFormat = nullptr;
};



audiogenerater.cpp


AudioGenerater::AudioGenerater()
{
 av_register_all();
 avcodec_register_all();
}

AudioGenerater::~AudioGenerater()
{
 // ... 
}

bool AudioGenerater::initFormat(QString audioFileName)
{
 // Create an output Format context
 int result = avformat_alloc_output_context2(&m_FormatContext, nullptr, nullptr, audioFileName.toLocal8Bit().data());
 if (result < 0) {
 return false;
 }

 m_OutputFormat = m_FormatContext->oformat;

 // Create an audio stream
 AVStream* audioStream = avformat_new_stream(m_FormatContext, m_AudioCodec);
 if (audioStream == nullptr) {
 avformat_free_context(m_FormatContext);
 return false;
 }

 // Set the parameters in the stream
 audioStream->id = m_FormatContext->nb_streams - 1;
 audioStream->time_base = { 1, 8000 };
 result = avcodec_parameters_from_context(audioStream->codecpar, m_AudioCodecContext);
 if (result < 0) {
 avformat_free_context(m_FormatContext);
 return false;
 }

 // Print FormatContext information
 av_dump_format(m_FormatContext, 0, audioFileName.toLocal8Bit().data(), 1);

 // Open file IO
 if (!(m_OutputFormat->flags & AVFMT_NOFILE)) {
 result = avio_open(&m_FormatContext->pb, audioFileName.toLocal8Bit().data(), AVIO_FLAG_WRITE);
 if (result < 0) {
 avformat_free_context(m_FormatContext);
 return false;
 }
 }

 return true;
}

void AudioGenerater::generateAudioFileWithOptions(
 QString _fileName,
 QByteArray _pcmData,
 int _channel,
 int _bitRate,
 int _sampleRate,
 AVSampleFormat _format)
{
 AVFormatContext* oc;
 if (avformat_alloc_output_context2(
 &oc, nullptr, nullptr, _fileName.toStdString().c_str())
 < 0) {
 qDebug() << "Error in line: " << __LINE__;
 return;
 }
 if (!oc) {
 printf("Could not deduce output format from file extension: using mka.\n");
 avformat_alloc_output_context2(
 &oc, nullptr, "mka", _fileName.toStdString().c_str());
 }
 if (!oc) {
 qDebug() << "Error in line: " << __LINE__;
 return;
 }
 AVOutputFormat* fmt = oc->oformat;
 if (fmt->audio_codec == AV_CODEC_ID_NONE) {
 qDebug() << "Error in line: " << __LINE__;
 return;
 }

 AVCodecID codecID = av_guess_codec(
 fmt, nullptr, _fileName.toStdString().c_str(), nullptr, AVMEDIA_TYPE_AUDIO);
 // Find Codec
 m_AudioCodec = avcodec_find_encoder(codecID);
 if (m_AudioCodec == nullptr) {
 qDebug() << "Error in line: " << __LINE__;
 return;
 }
 // Create an encoder context
 m_AudioCodecContext = avcodec_alloc_context3(m_AudioCodec);
 if (m_AudioCodecContext == nullptr) {
 qDebug() << "Error in line: " << __LINE__;
 return;
 }

 // Setting parameters
 m_AudioCodecContext->bit_rate = _bitRate;
 m_AudioCodecContext->sample_rate = _sampleRate;
 m_AudioCodecContext->sample_fmt = _format;
 m_AudioCodecContext->channels = _channel;

 m_AudioCodecContext->channel_layout = av_get_default_channel_layout(_channel);
 m_AudioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

 // Turn on the encoder
 int result = avcodec_open2(m_AudioCodecContext, m_AudioCodec, nullptr);
 if (result < 0) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // Create a package
 if (!initFormat(_fileName)) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // write to the file header
 result = avformat_write_header(m_FormatContext, nullptr);
 if (result < 0) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // Create Frame
 AVFrame* frame = av_frame_alloc();
 if (frame == nullptr) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 int nb_samples = 0;
 if (m_AudioCodecContext->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) {
 nb_samples = 10000;
 }
 else {
 nb_samples = m_AudioCodecContext->frame_size;
 }

 // Set the parameters of the Frame
 frame->nb_samples = nb_samples;
 frame->format = m_AudioCodecContext->sample_fmt;
 frame->channel_layout = m_AudioCodecContext->channel_layout;

 // Apply for data memory
 result = av_frame_get_buffer(frame, 0);
 if (result < 0) {
 av_frame_free(&frame);
 {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 }

 // Set the Frame to be writable
 result = av_frame_make_writable(frame);
 if (result < 0) {
 av_frame_free(&frame);
 {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 }

 int perFrameDataSize = frame->linesize[0];
 int count = _pcmData.size() / perFrameDataSize;
 bool needAddOne = false;
 if (_pcmData.size() % perFrameDataSize != 0) {
 count++;
 needAddOne = true;
 }

 int frameCount = 0;
 for (int i = 0; i < count; ++i) {
 // Create a Packet
 AVPacket* pkt = av_packet_alloc();
 if (pkt == nullptr) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 av_init_packet(pkt);

 if (i == count - 1)
 perFrameDataSize = _pcmData.size() % perFrameDataSize;

 // Synthesize WAV files
 memset(frame->data[0], 0, perFrameDataSize);
 memcpy(frame->data[0], &(_pcmData.data()[perFrameDataSize * i]), perFrameDataSize);

 frame->pts = frameCount++;
 // send Frame
 result = avcodec_send_frame(m_AudioCodecContext, frame);
 if (result < 0)
 continue;

 // Receive the encoded Packet
 result = avcodec_receive_packet(m_AudioCodecContext, pkt);
 if (result < 0) {
 av_packet_free(&pkt);
 continue;
 }

 // write to file
 av_packet_rescale_ts(pkt, m_AudioCodecContext->time_base, m_FormatContext->streams[0]->time_base);
 pkt->stream_index = 0;
 result = av_interleaved_write_frame(m_FormatContext, pkt);
 if (result < 0)
 continue;

 av_packet_free(&pkt);
 }

 // write to the end of the file
 av_write_trailer(m_FormatContext);
 // Close file IO
 avio_closep(&m_FormatContext->pb);
 // Release Frame memory
 av_frame_free(&frame);

 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
}



main.cpp


int main(int argc, char **argv)
{
 av_log_set_level(AV_LOG_TRACE);

 QFile file("rawDataOfPcmAlawType.bin");
 if (!file.open(QIODevice::ReadOnly)) {
 return EXIT_FAILURE;
 }
 QByteArray rawData(file.readAll());

 AudioGenerater generator;
 generator.generateAudioFileWithOptions(
 "test.mka",
 rawData,
 1, 
 64000, 
 8000,
 AV_SAMPLE_FMT_S16);

 return 0;
}



It is IMPORTANT you help me find the most appropriate way to record
pcm_alaw
or a different data format in anMKA
audio file.

I ask everyone who knows anything to help (there is too little time left to implement this project)


-
using ffmpeg headers with a c program
15 avril 2014, par orpgoli'm trying to learn ffmpeg from drangers guide (for school), i have a mac so the first thing i did was to use Macports and get ffmpeg and sdl...
now when i'm tried to compile drangers code, the compiler didnt recognize the headers...
so i used -I on gnu when compiling, and also gave the "whole/path/name..." to the headers,
but i always get an error on some header missing...below i will put the code with my corrections.
i also tried including all the headers that i get an error about but there is always another header the compiler can't find
i tried both gnu (on console) and on xcode.
// tutorial01.c
// Code based on a tutorial by Martin Bohme (boehme@inb.uni-luebeckREMOVETHIS.de)
// Tested on Gentoo, CVS version 5/01/07 compiled with GCC 4.1.1
// A small sample program that shows how to use libavformat and libavcodec to
// read video from a file.
//
// Use
//
// gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lz
//
// to build (assuming libavformat and libavcodec are correctly installed
// your system).
//
// Run using
//
// tutorial01 myvideofile.mpg
//
// to write the first five frames from "myvideofile.mpg" to disk in PPM
// format.
#include "/opt/local/include/libavcodec/avcodec.h"
#include "/opt/local/include/libavformat/avformat.h"
#include
void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame) {
FILE *pFile;
char szFilename[32];
int y;
// Open file
sprintf(szFilename, "frame%d.ppm", iFrame);
pFile=fopen(szFilename, "wb");
if(pFile==NULL)
return;
// Write header
fprintf(pFile, "P6\n%d %d\n255\n", width, height);
// Write pixel data
for(y=0; ydata[0]+y*pFrame->linesize[0], 1, width*3, pFile);
// Close file
fclose(pFile);
}
int main(int argc, char *argv[]) {
AVFormatContext *pFormatCtx;
int i, videoStream;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
AVFrame *pFrame;
AVFrame *pFrameRGB;
AVPacket packet;
int frameFinished;
int numBytes;
uint8_t *buffer;
if(argc < 2) {
printf("Please provide a movie file\n");
return -1;
}
// Register all formats and codecs
av_register_all();
// Open video file
if(av_open_input_file(&pFormatCtx, argv[1], NULL, 0, NULL)!=0)
return -1; // Couldn't open file
// Retrieve stream information
if(av_find_stream_info(pFormatCtx)<0)
return -1; // Couldn't find stream information
// Dump information about file onto standard error
dump_format(pFormatCtx, 0, argv[1], 0);
// Find the first video stream
videoStream=-1;
for(i=0; inb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_VIDEO) {
videoStream=i;
break;
}
if(videoStream==-1)
return -1; // Didn't find a video stream
// Get a pointer to the codec context for the video stream
pCodecCtx=pFormatCtx->streams[videoStream]->codec;
// Find the decoder for the video stream
pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec==NULL) {
fprintf(stderr, "Unsupported codec!\n");
return -1; // Codec not found
}
// Open codec
if(avcodec_open(pCodecCtx, pCodec)<0)
return -1; // Could not open codec
// Allocate video frame
pFrame=avcodec_alloc_frame();
// Allocate an AVFrame structure
pFrameRGB=avcodec_alloc_frame();
if(pFrameRGB==NULL)
return -1;
// Determine required buffer size and allocate buffer
numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
pCodecCtx->height);
buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
// Assign appropriate parts of buffer to image planes in pFrameRGB
// Note that pFrameRGB is an AVFrame, but AVFrame is a superset
// of AVPicture
avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
pCodecCtx->width, pCodecCtx->height);
// Read frames and save first five frames to disk
i=0;
while(av_read_frame(pFormatCtx, &packet)>=0) {
// Is this a packet from the video stream?
if(packet.stream_index==videoStream) {
// Decode video frame
avcodec_decode_video(pCodecCtx, pFrame, &frameFinished,
packet.data, packet.size);
// Did we get a video frame?
if(frameFinished) {
// Convert the image from its native format to RGB
img_convert((AVPicture *)pFrameRGB, PIX_FMT_RGB24,
(AVPicture*)pFrame, pCodecCtx->pix_fmt, pCodecCtx->width,
pCodecCtx->height);
// Save the frame to disk
if(++i<=5)
SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height,
i);
}
}
// Free the packet that was allocated by av_read_frame
av_free_packet(&packet);
}
// Free the RGB image
av_free(buffer);
av_free(pFrameRGB);
// Free the YUV frame
av_free(pFrame);
// Close the codec
avcodec_close(pCodecCtx);
// Close the video file
av_close_input_file(pFormatCtx);
return 0;
}