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GetID3 - Bloc informations de fichiers
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Sur d’autres sites (7879)
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Audio/Video encoding with ffmpeg
16 mars 2017, par VrozAudio/Video encoding with ffmpeg :
I am trying to create an avi file with encoded video and audio, using ffmpeg.
First, I create the file :
//define BITRATE 10000000
//define GOP 300
//define FPS 60
//define VIDEOTYPE "avi"
if (!encoder_->createFile(QFileInfo(*(videoFile_.data())).absoluteFilePath(), targetRect.width(), targetRect.height(), BITRATE*(1000 / FPS), GOP, 1000))The buffers are initialized as :
audio_outbuf_size = 44100 * 0.005 * 16; //5ms of audio should be encoded, each time this function is called
audio_outbuf = new uint8_t[audio_outbuf_size];
outbuf_size = getWidth()*getHeight() * 3;
outbuf = new uint8_t[outbuf_size];Then add audio and video streams (audio : CODEC_ID_PCM_S16LE, 16000 kb/s and 44100 Hz, video : PIX_FMT_YUV420P)
void MediaMuxer::addAudioStream(QString fileName, ffmpeg::CodecID codec_id)
{
// Add the audio stream
ffmpeg::AVCodec *encoder = avcodec_find_encoder(codec_id);
pAudioStream_ = ffmpeg::av_new_stream(pOutputFormatCtx_, 0);
if (!pAudioStream_) {
printf("Could not allocate stream\n");
return;
}
pAudioCodecCtx_ = pAudioStream_->codec;
pAudioCodecCtx_->codec_id = codec_id;
pAudioCodecCtx_->codec_type = ffmpeg::AVMEDIA_TYPE_AUDIO;
pAudioCodecCtx_->sample_fmt = ffmpeg::AV_SAMPLE_FMT_S16;
pAudioCodecCtx_->sample_fmt = encoder->sample_fmts[0];
pAudioCodecCtx_->bit_rate = 16000;
//pAudioCodecCtx_->bit_rate = 64000;
pAudioCodecCtx_->sample_rate = N;
pAudioCodecCtx_->channels = 1;
pAudioCodecCtx_->time_base.den = FPS;
pAudioCodecCtx_->time_base.num = 1;
avcodec_thread_init(pAudioCodecCtx_, 10);
// some formats want stream headers to be separate
if (pOutputFormatCtx_->oformat->flags & AVFMT_GLOBALHEADER)
pAudioCodecCtx_->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (av_set_parameters(pOutputFormatCtx_, NULL) < 0)
{
printf("Invalid output format parameters\n");
return;
}
//ffmpeg::dump_format(pOutputFormatCtx_, 0, fileName.toStdString().c_str(), 1);
// open_video
// find the audio encoder
pAudioCodec_ = avcodec_find_encoder(pAudioCodecCtx_->codec_id);
if (!pAudioCodec_)
{
printf("codec not found\n");
return;
}
// open the codec
if (avcodec_open(pAudioCodecCtx_, pAudioCodec_) < 0)
{
printf("could not open codec\n");
return;
}
// Allocate memory for output
if (!initAudioOutputBuf())
{
printf("Can't allocate memory for audio output bitstream\n");
return;
}
// Allocate the audio frame
if (!initAudioFrame())
{
printf("Can't init audio frame\n");
return;
}
if (url_fopen(&pOutputFormatCtx_->pb, fileName.toStdString().c_str(), URL_WRONLY) < 0)
{
printf("Could not open '%s'\n", fileName.toStdString().c_str());
return;
}
av_write_header(pOutputFormatCtx_);
}
void MediaMuxer::addVideoStream(QString fileName)
{
// Add the video stream
pVideoStream_ = ffmpeg::av_new_stream(pOutputFormatCtx_, 0);
if (!pVideoStream_)
{
printf("Could not allocate stream\n");
return;
}
pVideoCodecCtx_ = pVideoStream_->codec;
pVideoCodecCtx_->codec_id = pOutputFormat_->video_codec;
pVideoCodecCtx_->codec_type = ffmpeg::AVMEDIA_TYPE_VIDEO;
pVideoCodecCtx_->bit_rate = Bitrate;
pVideoCodecCtx_->width = getWidth();
pVideoCodecCtx_->height = getHeight();
pVideoCodecCtx_->time_base.den = FPS;
pVideoCodecCtx_->time_base.num = 1;
pVideoCodecCtx_->gop_size = Gop;
pVideoCodecCtx_->pix_fmt = ffmpeg::PIX_FMT_YUV420P;
avcodec_thread_init(pVideoCodecCtx_, 10);
// some formats want stream headers to be separate
if (pOutputFormatCtx_->oformat->flags & AVFMT_GLOBALHEADER)
pVideoCodecCtx_->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (av_set_parameters(pOutputFormatCtx_, NULL) < 0)
{
printf("Invalid output format parameters\n");
return;
}
//ffmpeg::dump_format(pOutputFormatCtx_, 0, fileName.toStdString().c_str(), 1);
// open_video
// find the video encoder
pVideoCodec_ = avcodec_find_encoder(pVideoCodecCtx_->codec_id);
if (!pVideoCodec_)
{
printf("codec not found\n");
return;
}
// open the codec
if (avcodec_open(pVideoCodecCtx_, pVideoCodec_) < 0)
{
printf("could not open codec\n");
return;
}
// Allocate memory for output
if (!initOutputBuf())
{
printf("Can't allocate memory for output bitstream\n");
return;
}
// Allocate the YUV frame
if (!initFrame())
{
printf("Can't init frame\n");
return;
}
if (url_fopen(&pOutputFormatCtx_->pb, fileName.toStdString().c_str(), URL_WRONLY) < 0)
{
printf("Could not open '%s'\n", fileName.toStdString().c_str());
return;
}
av_write_header(pOutputFormatCtx_);
}Finally, I call alternatively encodeVideo/encodeAudio to encode video and PCM audio frames at specific recording times(pts) :
int MediaMuxer::encodeVideo(const QImage &img, unsigned pts)
{
convertImage_sws(img); // SWS conversion
pVideoCodecCtx_->coded_frame->pts = pts; // Set the time stamp
int out_size = ffmpeg::avcodec_encode_video(pVideoCodecCtx_, outbuf, outbuf_size, ppicture);
pVideoCodecCtx_->coded_frame->pts = pts; // Set the time stamp
if (out_size > 0)
{
ffmpeg::av_init_packet(&pkt);
if (pVideoCodecCtx_->coded_frame->pts != (0x8000000000000000LL))
pkt.pts = av_rescale_q(pVideoCodecCtx_->coded_frame->pts, pVideoCodecCtx_->time_base, pVideoStream_->time_base);
if (pVideoCodecCtx_->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = pVideoStream_->index;
pkt.data = outbuf;
pkt.size = out_size;
int ret = ffmpeg::av_interleaved_write_frame(pOutputFormatCtx_, &pkt);
if (ret<0)
return -1;
}
return out_size;
}
int MediaMuxer::encodeAudio(unsigned pts)
{
pAudioCodecCtx_->coded_frame->pts = pts; // Set the time stamp
// simple sound encoding
int16_t samples[220] = { 0 }; // buffer
int n; // buffer index
double Fs = 44100.0; // sampling frequency
// Generate audio data
for (n = 0; n < 220; ++n) //220 samples (44100*.005sec as the interval between 2 video frames is 10ms)
samples[n] = 16383.0 * sin(n*1000.0*2.0*M_PI / Fs); //sine wav
int out_size = ffmpeg::avcodec_encode_audio(pAudioCodecCtx_, audio_outbuf, audio_outbuf_size, (const short*)samples);
pAudioCodecCtx_->coded_frame->pts = pts; // Set the time stamp
if (out_size>0)
{
// Packet
ffmpeg::AVPacket pkt = { 0 };
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
if (pAudioCodecCtx_->coded_frame->pts != (0x8000000000000000LL))
pkt.pts = av_rescale_q(pAudioCodecCtx_->coded_frame->pts, pAudioCodecCtx_->time_base, pAudioStream_->time_base);
if (pAudioCodecCtx_->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = pAudioStream_->index;
pkt.data = audio_outbuf;
pkt.size = out_size;
int ret = av_interleaved_write_frame(pOutputFormatCtx_, &pkt);
if (ret<0)
return -1;
av_free_packet(&pkt);
}
//end simple sound encoding
return pkt.size;
}The result is a nice video with some audio behind (either a regular beeping sound at regular intervals but ending way earlier than the video or a continuous longer sound that also last shorter than the video).
I want to generate a beeping sound each time the function encodeAudio() is called - at non-regular intervals. I have tried to modify the sampling rate, the buffer size, the pkt size and the number of samples but without any success. I also tried to set the pts at different times but it did not get me where I want to be. Could someone please help ?
-
ffmpeg stream replication of multiple sources to multiple destinations
16 mars 2017, par dpxI’m looking to replicate multiple streams from a single source, to multiple destinations. So for instance 12 streams with a dest of 1.1.1.1:1000-1011 being reflected to two destinations @ 2.2.2.2:1000-1011 and 3.3.3.3:1000-1011
I don’t want any stream processing, no modification, just using -vcodec / -acodec. Receiving MPEG-TS and reflecting MPEG-TS at the same bitrate.
How would I run this with FFMPEG ? Trying to read the documentation but it’s not quite clear on how to handle this.
-
FFMPEG AAC encoding causes audio to be lower in pitch
14 février 2017, par Paul KnopfI built a sample application that encodes AAC (from PortAudio) into a MP4 container (no video stream).
The resulting audio is lower in pitch.
#include "stdafx.h"
#include "TestRecording.h"
#include "libffmpeg.h"
TestRecording::TestRecording()
{
}
TestRecording::~TestRecording()
{
}
struct RecordingContext
{
RecordingContext()
{
formatContext = NULL;
audioStream = NULL;
audioFrame = NULL;
audioFrameframeNumber = 0;
}
libffmpeg::AVFormatContext* formatContext;
libffmpeg::AVStream* audioStream;
libffmpeg::AVFrame* audioFrame;
int audioFrameframeNumber;
};
static int AudioRecordCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData)
{
RecordingContext* recordingContext = (RecordingContext*)userData;
libffmpeg::avcodec_fill_audio_frame(recordingContext->audioFrame,
recordingContext->audioFrame->channels,
recordingContext->audioStream->codec->sample_fmt,
static_cast<const unsigned="unsigned">(inputBuffer),
(framesPerBuffer * sizeof(float) * recordingContext->audioFrame->channels),
0);
libffmpeg::AVPacket pkt;
libffmpeg::av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
int gotpacket;
int result = avcodec_encode_audio2(recordingContext->audioStream->codec, &pkt, recordingContext->audioFrame, &gotpacket);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't encode the audio frame to acc");
return paContinue;
}
if (gotpacket)
{
pkt.stream_index = recordingContext->audioStream->index;
recordingContext->audioFrameframeNumber++;
// this codec requires no bitstream filter, just send it to the muxer!
result = libffmpeg::av_write_frame(recordingContext->formatContext, &pkt);
if (result < 0)
{
LOG(ERROR) << "Couldn't write the encoded audio frame";
libffmpeg::av_free_packet(&pkt);
return paContinue;
}
libffmpeg::av_free_packet(&pkt);
}
return paContinue;
}
static bool InitializeRecordingContext(RecordingContext* recordingContext)
{
int result = libffmpeg::avformat_alloc_output_context2(&recordingContext->formatContext, NULL, NULL, "C:\\Users\\Paul\\Desktop\\test.mp4");
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't create output format context");
return false;
}
libffmpeg::AVCodec *audioCodec;
audioCodec = libffmpeg::avcodec_find_encoder(libffmpeg::AV_CODEC_ID_AAC);
if (audioCodec == NULL)
{
LOG(ERROR) << "Couldn't find the encoder for AAC";
}
recordingContext->audioStream = libffmpeg::avformat_new_stream(recordingContext->formatContext, audioCodec);
if (!recordingContext->audioStream)
{
LOG(ERROR) << "Couldn't create the audio stream";
return false;
}
recordingContext->audioStream->codec->bit_rate = 64000;
recordingContext->audioStream->codec->sample_fmt = libffmpeg::AV_SAMPLE_FMT_FLTP;
recordingContext->audioStream->codec->sample_rate = 48000;
recordingContext->audioStream->codec->channel_layout = AV_CH_LAYOUT_STEREO;
recordingContext->audioStream->codec->channels = libffmpeg::av_get_channel_layout_nb_channels(recordingContext->audioStream->codec->channel_layout);
recordingContext->audioStream->codecpar->bit_rate = recordingContext->audioStream->codec->bit_rate;
recordingContext->audioStream->codecpar->format = recordingContext->audioStream->codec->sample_fmt;
recordingContext->audioStream->codecpar->sample_rate = recordingContext->audioStream->codec->sample_rate;
recordingContext->audioStream->codecpar->channel_layout = recordingContext->audioStream->codec->channel_layout;
recordingContext->audioStream->codecpar->channels = recordingContext->audioStream->codec->channels;
result = libffmpeg::avcodec_open2(recordingContext->audioStream->codec, audioCodec, NULL);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio codec");
return false;
}
// create a new frame to store the audio samples
recordingContext->audioFrame = libffmpeg::av_frame_alloc();
if (!recordingContext->audioFrame)
{
LOG(ERROR) << "Couldn't alloce the output audio frame";
return false;
}
recordingContext->audioFrame->nb_samples = recordingContext->audioStream->codec->frame_size;
recordingContext->audioFrame->channel_layout = recordingContext->audioStream->codec->channel_layout;
recordingContext->audioFrame->channels = recordingContext->audioStream->codec->channels;
recordingContext->audioFrame->format = recordingContext->audioStream->codec->sample_fmt;
recordingContext->audioFrame->sample_rate = recordingContext->audioStream->codec->sample_rate;
result = libffmpeg::av_frame_get_buffer(recordingContext->audioFrame, 0);
if (result < 0)
{
LOG(ERROR) << "Coudln't initialize the output audio frame buffer";
return false;
}
// some formats want video_stream headers to be separate
if (!strcmp(recordingContext->formatContext->oformat->name, "mp4") || !strcmp(recordingContext->formatContext->oformat->name, "mov") || !strcmp(recordingContext->formatContext->oformat->name, "3gp"))
{
recordingContext->audioStream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// open the ouput file
if (!(recordingContext->formatContext->oformat->flags & AVFMT_NOFILE))
{
result = libffmpeg::avio_open(&recordingContext->formatContext->pb, recordingContext->formatContext->filename, AVIO_FLAG_WRITE);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the output file");
return false;
}
}
// write the stream headers
result = libffmpeg::avformat_write_header(recordingContext->formatContext, NULL);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the headers to the file");
return false;
}
return true;
}
static bool FinalizeRecordingContext(RecordingContext* recordingContext)
{
int result = 0;
// write the trailing information
if (recordingContext->formatContext->pb)
{
result = libffmpeg::av_write_trailer(recordingContext->formatContext);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the trailer information");
return false;
}
}
// close all the codes
for (int i = 0; i < (int)recordingContext->formatContext->nb_streams; i++)
{
result = libffmpeg::avcodec_close(recordingContext->formatContext->streams[i]->codec);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the codec");
return false;
}
}
// close the output file
if (recordingContext->formatContext->pb)
{
if (!(recordingContext->formatContext->oformat->flags & AVFMT_NOFILE))
{
result = libffmpeg::avio_close(recordingContext->formatContext->pb);
if (result < 0)
{
LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the output file");
return false;
}
}
}
// free the format context and all of its data
libffmpeg::avformat_free_context(recordingContext->formatContext);
recordingContext->formatContext = NULL;
recordingContext->audioStream = NULL;
if (recordingContext->audioFrame)
{
libffmpeg::av_frame_free(&recordingContext->audioFrame);
recordingContext->audioFrame = NULL;
}
return true;
}
int TestRecording::Test()
{
PaError result = paNoError;
result = Pa_Initialize();
if (result != paNoError) LOGINT_WITH_MESSAGE(ERROR, result, "Error initializing audio device framework");
RecordingContext recordingContext;
if (!InitializeRecordingContext(&recordingContext))
{
LOG(ERROR) << "Couldn't start recording file";
return 0;
}
auto defaultDevice = Pa_GetDefaultInputDevice();
auto deviceInfo = Pa_GetDeviceInfo(defaultDevice);
PaStreamParameters inputParameters;
inputParameters.device = defaultDevice;
inputParameters.channelCount = 2;
inputParameters.sampleFormat = paFloat32;
inputParameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
inputParameters.hostApiSpecificStreamInfo = NULL;
PaStream* stream = NULL;
result = Pa_OpenStream(
&stream,
&inputParameters,
NULL,
48000,
1024,
paClipOff,
AudioRecordCallback,
&recordingContext);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio stream");
result = Pa_StartStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't start the audio stream");
Sleep(1000 * 5);
result = Pa_StopStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
if (!FinalizeRecordingContext(&recordingContext)) LOG(ERROR) << "Couldn't stop recording file";
result = Pa_CloseStream(stream);
if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
return 0;
}
</const>Here is the
stdout, in case it helps.https://gist.github.com/pauldotknopf/9f24a604ce1f8a081aa68da1bf169e98
Why is the audio lower in pitch ? I assume I am overlooking a parameter that needs to be configured between PortAudio and FFMPEG. Is there something super obvious that I am missing ?
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