Visualise audio waveform from video local file using Qmedia player - c++

I try without success to plot waveform using qMediaPlayer and QaudioProbe object to get the QAudioBuffer but it's always fails when I try:
player = new QMediaPlayer;
audio = new QAudioProbe ;
QAudioRecorder *recorder = new QAudioRecorder();
if (audio->setSource(player))
{
// Probing succeeded, audioProbe->isValid() should be true.
std::cout << "probing succed"<< std::endl;
connect(audio, SIGNAL(audioBufferProbed(QAudioBuffer)), this,
SLOT(processBuffer(QAudioBuffer)));
}
this line:
if (audio->setSource(player))
always return false!
when I replace QMediaPlayer by QAudioRecorder the setSource function works well.
do you have any idea to do that, or m'I in a wrong direction?
otherwise is there other way to split audio from video file.
thanks a lot

From the documentation on QMediaPlayer, I would gather that since the property audioAvailable can change, the default is that audioAvailable is false.
If there is no audio available, the documentation of setSource states that
"If the media object does not support monitoring audio, this function
will return false."
Try loading an actual piece of media, that has audio available (check that first) before trying to set the source

Related

Use Source Reader to get H264 samples from webcam source

When using the Source Reader I can use it to get decoded YUV samples using an mp4 file source (example code).
How can I do the opposite with a webcam source? Use the Source Reader to provide encoded H264 samples? My webcam supports RGB24 and I420 pixel formats and I can get H264 samples if I manually wire up the H264 MFT transform. But it seems as is the Source Reader should be able to take care of the transform for me. I get an error whenever I attempt to set MF_MT_SUBTYPE of MFVideoFormat_H264 on the Source Reader.
Sample snippet is shown below and the full example is here.
// Get the first available webcam.
CHECK_HR(MFCreateAttributes(&videoConfig, 1), "Error creating video configuration.");
// Request video capture devices.
CHECK_HR(videoConfig->SetGUID(
MF_DEVSOURCE_ATTRIBUTE_SOURCE_TYPE,
MF_DEVSOURCE_ATTRIBUTE_SOURCE_TYPE_VIDCAP_GUID), "Error initialising video configuration object.");
CHECK_HR(videoConfig->SetGUID(MF_MT_SUBTYPE, WMMEDIASUBTYPE_I420),
"Failed to set video sub type to I420.");
CHECK_HR(MFEnumDeviceSources(videoConfig, &videoDevices, &videoDeviceCount), "Error enumerating video devices.");
CHECK_HR(videoDevices[WEBCAM_DEVICE_INDEX]->GetAllocatedString(MF_DEVSOURCE_ATTRIBUTE_FRIENDLY_NAME, &webcamFriendlyName, &nameLength),
"Error retrieving video device friendly name.\n");
wprintf(L"First available webcam: %s\n", webcamFriendlyName);
CHECK_HR(videoDevices[WEBCAM_DEVICE_INDEX]->ActivateObject(IID_PPV_ARGS(&pVideoSource)),
"Error activating video device.");
CHECK_HR(MFCreateAttributes(&pAttributes, 1),
"Failed to create attributes.");
// Adding this attribute creates a video source reader that will handle
// colour conversion and avoid the need to manually convert between RGB24 and RGB32 etc.
CHECK_HR(pAttributes->SetUINT32(MF_SOURCE_READER_ENABLE_VIDEO_PROCESSING, 1),
"Failed to set enable video processing attribute.");
CHECK_HR(pAttributes->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Video), "Failed to set major video type.");
// Create a source reader.
CHECK_HR(MFCreateSourceReaderFromMediaSource(
pVideoSource,
pAttributes,
&pVideoReader), "Error creating video source reader.");
MFCreateMediaType(&pSrcOutMediaType);
CHECK_HR(pSrcOutMediaType->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Video), "Failed to set major video type.");
CHECK_HR(pSrcOutMediaType->SetGUID(MF_MT_SUBTYPE, MFVideoFormat_H264), "Error setting video sub type.");
CHECK_HR(pSrcOutMediaType->SetUINT32(MF_MT_AVG_BITRATE, 240000), "Error setting average bit rate.");
CHECK_HR(pSrcOutMediaType->SetUINT32(MF_MT_INTERLACE_MODE, 2), "Error setting interlace mode.");
CHECK_HR(pVideoReader->SetCurrentMediaType((DWORD)MF_SOURCE_READER_FIRST_VIDEO_STREAM, NULL, pSrcOutMediaType),
"Failed to set media type on source reader.");
CHECK_HR(pVideoReader->GetCurrentMediaType((DWORD)MF_SOURCE_READER_FIRST_VIDEO_STREAM, &pFirstOutputType),
"Error retrieving current media type from first video stream.");
std::cout << "Source reader output media type: " << GetMediaTypeDescription(pFirstOutputType) << std::endl << std::endl;
Output:
bind returned success
First available webcam: Logitech QuickCam Pro 9000
Failed to set media type on source reader. Error: C00D5212.
finished.
Source Reader does not look like suitable API here. It is API to implement "half of pipeline" which includes necessary decoding but not encoding. The other half is Sink Writer API which is capable to handle encoding, and which can encode H.264.
Or your another option, unless you are developing a UWP project, is Media Session API which implements a pipeline end to end.
Even though technically (in theory) you could have an encoding MFT as a part of Source Reader pipeline, Source Reader API itself is insufficiently flexible to add encoding style tansforms based on requested media types.
So, one solution could be to have Source Reader to read with necessary decoding (such as up to having RGB32 or NV12 video frames), then Sink Writer to manage encoding with respectively appropriate media sink on its end (or Sample Grabber as media sink). Another solution is to put Media Foundation primitives into Media Session pipeline which can manage both decoding and encoding parts, connected together.
Now, your use case is clearer.
For me, your MFWebCamRtp is the best optimized way of doing : WebCam Source Reader -> Encoding -> RTP Streaming.
But you are experiencing presentation clock issues, synchronization issues, or unsynchronized audio video issues. Am I right ?
So you tried Sample Grabber Sink, and now Source Reader, like I suggested to you. Of course, you can think that a Media Session will be able to do it better.
I think so, but extra work will be needed.
Here is what I would do in your case :
Code a custom RTP Sink
Create a topology with webcam source, h264 encoder, your custom RTP Sink
Add your topology to a MediaSession
Use the MediaSession to play the process
If you want a networkstream sink sample, see this : MFSkJpegHttpStreamer
This is old, but it's a good start. This program also uses winsock, like your.
You should be aware that RTP protocol uses UDP. A very good way to have synchronization issues... Definitely your main problem, as I guess.
What I think. You are trying to compensate for the weaknesses of the RTP protocol (UDP), with a management of the audio / video synchronization of MediaFoundation. I think you will just fail with this approach.
I think your main problem is RTP protocol.
EDIT
No I'm not having synchronisation issues. The Source Reader and Sample Grabber both provide correct timestamps which I can use in the RTP header. Likewise no problems with RTP/UDP etc. that's the bit I do know about. My questions are originating from a desire to understand the most efficient (least amount of plumbing code) and flexible solution. And yes it does look like a custom sink writer is the optimal solution.
Again things are clearer. If you need help with a custom RTP sink, I'll be there.

ParaView: Live point cloud visualization plugin

I am writing a ParaView version 5.1.2 plugin in C++ to visualize point cloud data produced by a LiDAR sensor. I noticed that Velodyne has an open source ParaView custom application to visualize their LiDAR data called Veloview. I tweaked some of their code to start but I am stuck now.
So far I wrote a reader that takes a pcap file and renders a point cloud that can be played back frame by frame. I also wrote a ParaView source that listens on a port and captures udp packets and after they are captured uses the reader to split them into frames and visualize the PC.
Now I would like to take live udp packets and render the point cloud in real time as each frame is completed.
I am having trouble accomplishing this because of the ParaView plugin structure. Currently, my reader displays a frame when the method RequestData is called. My method looks something like this.
int RequestData(vtkInformation *request, vtkInformationVector **inputVector, vtkInformationVector *outputVector){
vtkPolyData* output = vtkPolyData::GetData(outputVector);
vtkInformation* info = outputVector->GetInformationObject(0);
int timestep = 0;
if (info->Has(vtkStreamingDemandDrivenPipeline::UPDATE_TIME_STEP()))
{
double timeRequest = info->Get(vtkStreamingDemandDrivenPipeline::UPDATE_TIME_STEP());
int length = info->Length(vtkStreamingDemandDrivenPipeline::TIME_STEPS());
timestep = static_cast<int>(floor(timeRequest + 0.5));
}
this->Open();
// GetFrame returns a vtkSmartPointer<vtkPolyData> that is the frame
output->ShallowCopy(this->GetFrame(timestep));
this->Close();
return 1;
}
The RequestData method is called every time the timestep is updated in the ParaView gui. Then the frame from that timestep is copied into the outputVector.
I am not sure how to implement this with live data because in that circumstance the RequestData method is not called because no timesteps are requested. I saw there is a way to keep RequestData executing by using CONTINUE_EXECUTING() in this way.
request->Set(vtkStreamingDemandDrivenPipeline::CONTINUE_EXECUTING(), 1);
But I do not know if that is supposed to be used to visualize live data.
For now I am interested in simply reading live packets and throwing them away as soon as their frame is rendered. Does anyone know how I can achieve this?
In the code of VeloView (which basically is a bundled ParaView+LidarPlugin), the timesteps of ParaView is changed by the main code, not the Lidar Plugin.
We advice you to start from VeloView code, which is much closer to your goal.
If you really want to start from scratch within ParaView, you need to increment this requested timestep yourself.
Newest version of VeloView (unreleased) uses the same mechanism as ParaView “LiveSource” plugin (available in 5.6+), where the plugin tells ParaView to set a QtTimer that will automatically increment the available and requested timesteps.
request->Set(vtkStreamingDemandDrivenPipeline::CONTINUE_EXECUTING(), 1); relates to another mechanism that will run request Data multiple time, but won’t take care of updating the requested timestep.
Best,
Bastien Jacquet
VeloView project leader

Stream live audio live555

I was writing as I could not find the answer in previous topics. I am using live555 to stream live video (h264) and audio(g723), which are being recorded by a web camera. The video part is already done and it works perfectly, but I have no clue about the audio task.
As long as I have read I have to create a ServerMediaSession to which I should add two subsessions: one for the video and one for the audio. For the video part I created a subclass of OnDemandServerMediaSubsession, a subclass of FramedSource and the Encoder class, but for the audio aspect I do not know on which classes should I base the implementation.
The web camera records and delivers audio frames in g723 format separatedly from the video. I would say the audio is raw as when I try to play it in VLC it says that it could not find any startcode; so I suppose it is the raw audio stream what is recorded by the web cam.
I was wondering if someone could give me a hint.
For an audio stream ,your override of OnDemandServerMediaSubsession::createNewRTPSink should create a SimpleRTPSink.
Something like :
RTPSink* YourAudioMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* inputSource)
{
return SimpleRTPSink::createNew(envir(), rtpGroupsock,
4,
frequency,
"audio",
"G723",
channels );
}
The frequency and the number of channels should comes from the inputSource.

Recording and playing MP3 audio

I have followed the different threads on how to record and play MP3 but I still always get this exception trying to play MP3 files that I have recorded:
mp3filereader does not support sample rate changes
So here is my code to record :
waveInStream = new WaveIn();
waveInStream.WaveFormat = new WaveFormat(8000, 16, 1);
writer = new WaveFileWriter(outputfileName, waveInStream.WaveFormat);
waveInStream.DataAvailable += new EventHandler<WaveInEventArgs>(waveInStream_DataAvailable);
waveInStream.StartRecording();
The waveInStream_DataAvailable is :
void waveInStream_DataAvailable(object sender, WaveInEventArgs e)
{
writer.Write(e.Buffer, 0, e.BytesRecorded);
}
At this point the recorded file should be PCM uncompressed right?
Do I need to transcode it to MP3 before being able to play it?
My playing code:
WaveChannel32 inputStream;
WaveStream mp3Reader = new Mp3FileReader(fileName); var pStream = NAudio.Wave.WaveFormatConversionStream.CreatePcmStream(mp3Reader);
inputStream = new WaveChannel32(mp3Reader);
volumeStream = inputStream;
return volumeStream;
The exception occurs every time at the call of Mp3FileReader and says something like:
Got a frame at sample rate 44100, in a MP3 sample rate 32000
Mp3FileReader does not support sample rate change
Yes, you have saved a WAV file, not an MP3 file. Either convert to MP3 using something like LAME.exe, or just use the WaveFileReader instead of the Mp3FileReader. MP3 doesn't really support low sample rates like 8kHz in any case, which is typically only used for telephony.

How to play an audio file with continously updated QBuffer (for audio file) with Phonon?

I’m using Phonon player to play the audio files.
Scenarios:
Files played from local drive : Plays properly.
Files played from remote drive : As the audio files are on a USB device I have to keep updating the buffer (QBuffer) and simultaneously play the file. But for some reason the file is not playing in Phonon player. Can anyone please tell me the right way to play the audio file while the buffer is still getting updated?
//Code
Phonon::MediaObject* m_pMediaObject = new Phonon::MediaObject(this);
Phonon::AudioOutput* audioOutput = new Phonon::AudioOutput(Phonon::MusicCategory, this);
Phonon::Path path = Phonon::createPath(m_pMediaObject, audioOutput);
QBuffer m_pBufferLoop = new QBuffer(this);
m_pBufferLoop->open(QIODevice::Append || | QIODevice::ReadWrite);
functionToUpdateBuffer();//updates the buffer dynamically.
m_pMediaObject->setCurrentSource(m_pBufferLoop);
m_pMediaObject->play();
Nothing happens after I call play(). But if I give the complete buffer then the same code works fine.