问题描述
想象一下，在我的项目中，我收到了RTP个数据包，其中包含有效载荷type-8，以便稍后将此负载保存为音轨的第N部分。我从RTP数据包中提取了此负载，并将其保存到临时缓冲区：
...
while ((rtp = receiveRtpPackets()).withoutErrors()) {
payloadData.push(rtp.getPayloadData());
}
audioGenerator.setPayloadData(payloadData);
audioGenerator.recordToFile();
...
用此有效负载填充一定大小的临时缓冲区后，我将处理此缓冲区，即提取整个有效负载并使用ffmpeg对其进行编码，以进一步保存为Matroska格式的音频文件。但是我有一个问题。由于RTP数据包的有效载荷为type 8，因此我必须将pcm_alaw格式的原始音频数据保存为mka音频格式。但是，当将原始数据pcm_alaw保存到音频文件时，我会从库中获得以下消息：
...
[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
...
当您在vlc中打开音频文件时，没有任何播放（音频轨道时间戳丢失）。
我的项目的任务是简单地提取 pcm_alaw 数据并将其以mka格式包装在容器中。确定编解码器的最佳方法是使用av_guess_codec()函数，该函数依次自动选择所需的编解码器ID。但是，我不知道如何将原始数据正确打包到容器中。
请注意，我可以获取由RTP数据包类型（ All类型的RTP数据包有效负载）。我所知道的是，在任何情况下，我都必须将音频数据打包在mka容器中。
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,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 @R_746_4045@ion
av_dump_format(m_FormatContext,audioFileName.toLocal8Bit().data(),1);
// Open file IO
if (!(m_OutputFormat->flags & AVFMT_NOFILE)) {
result = avio_open(&m_FormatContext->pb,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,_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,"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,_fileName.toStdString().c_str(),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],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;
}
这是重要，您可以帮助我找到在pcm_alaw音频文件中录制MKA或其他数据格式的最合适方法。
我问每个知道什么都可以帮助的人（实施该项目的时间太少了）
解决方法
这些有用的链接将为您提供帮助：
- libav中的数据处理顺序的很好概述：ffmpeg-libav-tutorial
- 来自ffmpeg开发人员本身的示例：avio_reading，resampling_audio，transcode_aac