How to play AAC encoded audio data in Memory. UWP samples showing playback using files. playAudio() callback will be called for every 100ms.
void AACPlay::playAudio(void *aacData) {
// To do - play aacData
}
Solved the issue by supplying AAC data to MediaStreamSource. Used MediaPlayer to play MediaStreamSample data.
Related
I am trying to build a web-application with the functionality of screen-recording with system audio + headphone-mic audio being captured in the saved video.
I have been thoroughly googling on a solution for this, however my findings show multiple browser solutions where the above works so long as headphones are NOT connected, meaning the microphone input is coming from the system rather than headset.
In the case that you connect headphones, all of these solutions capture the screen without video-audio, and the microphone audio from my headset. So to re-clarify on this, it should have recorded video-audio from the video being played whilst recording, and the headset-mic audio also.
This is thoroughly available in native applications, however I am searching for a way to do this on a browser.
If there are no solutions for this currently that anybody knows of, some insight on the limitations around developing this would also really help, thank you.
Your browser manages the media input being received in the selected tab/window
To receive media input, you need to ensure you have the checkbox Share Audio in the image below checked. However this will only record media-audio being played in your headphones, when it comes to receiving microphone audio, the opposite must be done i.e the checkbox should be unchecked, or merge the microphone audio separately on saving the recorded video
https://slack-files.com/T1JA07M6W-F0297CM7F32-89e7407216
create two const, one retrieving on-screen video, other retrieving audio media:
const DISPLAY_STREAM = await navigator.mediaDevices.getDisplayMedia({video: {cursor: "motion"}, audio: {'echoCancellation': true}}); // retrieving screen-media
const VOICE_STREAM = await navigator.mediaDevices.getUserMedia({ audio: {'echoCancellation': true}, video: false }); // retrieving microphone-media
Use AudioContext to retrieve audio sources from getUserMedia() and getDisplayMedia() separately:
const AUDIO_CONTEXT = new AudioContext();
const MEDIA_AUDIO = AUDIO_CONTEXT.createMediaStreamSource(DISPLAY_STREAM); // passing source of on-screen audio
const MIC_AUDIO = AUDIO_CONTEXT.createMediaStreamSource(VOICE_STREAM); // passing source of microphone audio
Use the method below to create a new audio source which will be used as as the merger or merged version of audio, then passing audios into the merger:
const AUDIO_MERGER = AUDIO_CONTEXT.createMediaStreamDestination(); // audio merger
MEDIA_AUDIO.connect(AUDIO_MERGER); // passing media-audio to merger
MIC_AUDIO.connect(AUDIO_MERGER); // passing microphone-audio to merger
Finally, connect the merged-audio and video together into one array to form a track, and pass it to the MediaStreamer:
const TRACKS = [...DISPLAY_STREAM.getVideoTracks(), ...AUDIO_MERGER.stream.getTracks()] // connecting on-screen video with merged-audio
stream = new MediaStream(TRACKS);
I have a video and a WebRTC audio stream and want to use Web Audio API to send the audio from the video to the Left channel, while the WebRTC to the right channel. So basically I'm doing:
video = document.getElementsByTagName("video")[0]
video.src = "http://link/to/my/video"
video.load()
audioContext = new AudioContext()
videoSourceL = audioContext.createMediaElementSource(video)
#create merger with 2 inputs, left (0) and right (1)
merger = audioContext.createChannelMerger(2)
merger.connect(audioContext.destination)
#now strange work around for WebRTC
audio = new Audio();
audio.muted = true
audio.srcObject = remoteStream
audioStreamR = audioContext.createMediaStreamSource(remoteStream)
# connect remote audio stream channel 0 to input 1 (right)
audioStreamR.connect(merger, 0, 1)
#connect video source channel 0 to input 0 (left)
videoSourceL.connect(merger, 0, 0)
The problem I have is that although the remote audio does go to the right channel (And is not audible in the Left), the audio from the video is also still slightly present in the right channel. So basically I have audio bleeding. The weird thing is that if I redirect both the remote stream and the video to the same channel, then the other channel has absolute silence.
Whereas if I had used an oscillator in place of the video audio, I would have a perfect separation. Any idea what I'm doing wrong?
EDIT: I also tried from the OS audio settings to turn off the left channel, and the audio bleeding to the right channel stopped (also tried this on a colleagues machine), so is this maybe a hardware/configuration
issue?
Was a hardware issue after all, effect is not there with good headphones.
I have C# project where stream from ip-camera recorded to the file, I use libvlc.
This is part of code with vlc parameters:
string VlcArguments = #":sout=#transcode{acodec=mpga,deinterlace}:standard{access=file,mux=mp4,dst="C:\Users\I\Desktop\Output.mp4"}";
var media = factory.CreateMedia<IMedia>(rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov, VlcArguments);
var player = factory.CreatePlayer<IPlayer>();
player.Open(media);
filename is the path of the result file.
It works fine, but I need to record sound from a microphone Microphone (High Definition Audio Device).
What I need to change to achieve that?
UPD
It should look something like this
var media = factory.CreateMedia<IMedia>("dshow:// dshow-vdev=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov dshow-adev=Microphone (High Definition Audio Device)", VlcArguments)
But it doesn't work (
UPD2
So, I think I found the answer
https://forum.videolan.org/viewtopic.php?f=14&t=124229&p=425550&hilit=camera+microphone+dshow#p425550
Unfortunately this will not work
I am trying to run the example code of Media Codec API with H264 Encoder on 4.3 explained in following link of bigflake
http://bigflake.com/mediacodec/CameraToMpegTest.java.txt
I have faced following problem.
-> In H264 encoder code the color format,height and width are not getting updated because there is problem in getpatameter implemetation. So i applied this patch (https://code.google.com/p/android/issues/detail?id=58834).
-> After applying the patch,also encoder does not encode
-> I have seen the observation like
D/CameraToMpegTest( 3421): encoder output format changed: {csd-1=java.nio.ByteArrayBuffer[position=0,limit=8,capacity=8], height=144, mime=video/avc, csd-0=java.nio.ByteArrayBuffer[position=0,limit=12,capacity=12], what=1869968451, width=176}
SO why this value is getting changed, No idea...
After that we always see encoder gives status of queueOutputBuffer as INFO_TRY_AGAIN_LATER.
So it creates the file but it does not encode anything and it stops as
I/MPEG4Writer( 3421): Received total/0-length (0/0) buffers and encoded 0 frames. - video
D/MPEG4Writer( 3421): Stopping Video track
D/MPEG4Writer( 3421): Stopping Video track source
D/MPEG4Writer( 3421): Video track stopped
D/MPEG4Writer( 3421): Stopping writer thread
D/MPEG4Writer( 3421): 0 chunks are written in the last batch
D/MPEG4Writer( 3421): Writer thread stopped
So in my understanding it should work but looks like still encoder is not getting configured properly...
Please guide on this...
Thanks
Nehal
The "encoder output format changed" message is normal in Android 4.3. That's how the encoder gives you a MediaFormat with csd-0/csd-1 keys, needed by MediaMuxer#addTrack().
Bug 58834 is for the VP8 software encoder; those patches shouldn't be needed for the hardware AVC codec.
The most common reason for INFO_TRY_AGAIN_LATER is lack of input. The encoder may queue up a fair number of input frames before producing any output, so you can't just submit one frame and then wait for output to appear. Turn on the VERBOSE flag and make sure that frames are being submitted.
I have tried running CameraToMpegTest sample on Android 4.3 emulator. As you'd have realized by now, it's not going to work as it is, and some fixes are required.
Implement getparameter properly in SoftAVCEncoder (in case of MIME type - "video/avc") for parameters like width, height, colour format. Otherwise your MediaFormat is not configured properly, and createInputSurface would fail. (I am not sure why this doesn't cause any problem when running H.264 encoding using Mediarecorder)
Fix the EGL attributes
Most importantly, if you're trying to execute this code in Activity context, make sure you don't block onFrameAvailable callback
(final void join()
Blocks the current Thread (Thread.currentThread()) until the receiver finishes its execution and dies.)
As the code snippet, you should remove th.join();
/** Entry point. */
public static void runTest(CameraToMpegTest obj) throws Throwable {
CameraToMpegWrapper wrapper = new CameraToMpegWrapper(obj);
Thread th = new Thread(wrapper, "codec test");
th.start();
// th.join();
if (wrapper.mThrowable != null) {
throw wrapper.mThrowable;
}
}
It works well for me.
Does anyone know of a good repository to get sample code for the BlackBerry? Specifically, samples that will help me learn the mechanics of recording audio, possibly even sampling it and doing some on the fly signal processing on it?
I'd like to read incoming audio, sample by sample if need be, then process it to produce a desired result, in this case a visualizer.
RIM API contains JSR 135 Java Mobile Media API for handling audio & video content.
You correct about mess on BB Knowledge Base. The only way is browse it, hoping they'll not going to change site map again.
It's Developers->Resources->Knowledge Base->Java API's&Samples->Audio&Video
Audio Recording
Basically it's simple to record audio:
create Player with correct audio encoding
get RecordControl
start recording
stop recording
Links:
RIM 4.6.0 API ref: Package javax.microedition.media
How To - Record Audio on a BlackBerry smartphone
How To - Play audio in an application
How To - Support streaming audio to the media application
How To - Specify Audio Path Routing
How To - Obtain the media playback time from a media application
What Is - Supported audio formats
What Is - Media application error codes
Audio Record Sample
Thread with Player, RecordControl and resources is declared:
final class VoiceNotesRecorderThread extends Thread{
private Player _player;
private RecordControl _rcontrol;
private ByteArrayOutputStream _output;
private byte _data[];
VoiceNotesRecorderThread() {}
private int getSize(){
return (_output != null ? _output.size() : 0);
}
private byte[] getVoiceNote(){
return _data;
}
}
On Thread.run() audio recording is started:
public void run() {
try {
// Create a Player that captures live audio.
_player = Manager.createPlayer("capture://audio");
_player.realize();
// Get the RecordControl, set the record stream,
_rcontrol = (RecordControl)_player.getControl("RecordControl");
//Create a ByteArrayOutputStream to capture the audio stream.
_output = new ByteArrayOutputStream();
_rcontrol.setRecordStream(_output);
_rcontrol.startRecord();
_player.start();
} catch (final Exception e) {
UiApplication.getUiApplication().invokeAndWait(new Runnable() {
public void run() {
Dialog.inform(e.toString());
}
});
}
}
And on thread.stop() recording is stopped:
public void stop() {
try {
//Stop recording, capture data from the OutputStream,
//close the OutputStream and player.
_rcontrol.commit();
_data = _output.toByteArray();
_output.close();
_player.close();
} catch (Exception e) {
synchronized (UiApplication.getEventLock()) {
Dialog.inform(e.toString());
}
}
}
Processing and sampling audio stream
In the end of recording you will have output stream filled with data in specific audio format. So to process or sample it you will have to decode this audio stream.
Talking about on the fly processing, that will be more complex. You will have to read output stream during recording without record commiting. So there will be several problems to solve:
synch access to output stream for Recorder and Sampler - threading issue
read the correct amount of audio data - go deep into audio format decode to find out markup rules
Also may be useful:
java.net: Experiments in Streaming Content in Java ME by Vikram Goyal
While not audio specific, this question does have some good "getting started" references.
Writing Blackberry Applications
I spent ages trying to figure this out too. Once you've installed the BlackBerry Component Packs (available from their website), you can find the sample code inside the component pack.
In my case, once I had installed the Component Packs into Eclipse, I found the extracted sample code in this location:
C:\Program
Files\Eclipse\eclipse3.4\plugins\net.rim.eide.componentpack4.5.0_4.5.0.16\components\samples
Unfortunately when I imported all that sample code I had a bunch of compile errors. To workaround that I just deleted the 20% of packages with compile errors.
My next problem was that launching the Simulator always launched the first sample code package (in my case activetextfieldsdemo), I couldn't get it to run just the package I am interested in. Workaround for that was to delete all the packages listed alphabetically before the one I wanted.
Other gotchas:
-Right click on the project in Eclipse and select Activate for BlackBerry
-Choose BlackBerry -> Build Configurations... -> Edit... and select your new project so it builds.
-Make sure you put your BlackBerry source code under a "src" folder in the Eclipse project, otherwise you might hit build issues.