Develop Reference

Android Multimedia Tunneling Sync Header

Context
I'm creating an Android application playing Media Source Extensions streams using Multimedia Tunneling. I'm using the API call flow as provided by the documentation. Audio part is handled with an AudioTrack. AudioSessionID is shared between the video MediaCodec and AudioTrack. Android SDK version is 26.
Problem
Video is being played correctly but no audio can be heard.
I do not have any error reported by the API.
Buffers are written in OutputBuffer using AudioTrack.write.
Non tunneling playback audio works well.
audio_hal does not produce any error in the logs.
Question
I've looked into the ExoPlayer implementation and I see the use of a sync header before writing the buffer to the AudioTrack in tunneling playback.
ByteBuffer avSyncHeader = ByteBuffer.allocate(16);
avSyncHeader.order(ByteOrder.BIG_ENDIAN);
avSyncHeader.putInt(0x55550001);
avSyncHeader.putInt(4, size);
avSyncHeader.putLong(8, presentationTimeUs * 1000);
avSyncHeader.position(0);
audioTrack.write(avSyncHeader, avSyncHeader.remaining(), WRITE_NON_BLOCKING);
I have tried adding that header too but audio was still not heard.
Is this sync header necessary?
Is there any other non documented requirement for Multimedia Tunneling?

Avsync header is for the level SDK, you can use another AudioTrack.write, to write the every buffer timestamp. It can auto generate the AV sync header.
Use another API, which can write timestamp.
Try:
int write(ByteBuffer audioData, int sizeInBytes, int writeMode, long timestamp)
Writes the audio data to the audio sink for playback in streaming mode on a HW_AV_SYNC track

Streaming Audio in FLAC or AMR_WB to the Google Speech API

I need to run the google speech api in somewhat low bandwidth environments.
Based on reading about best practices, it seems my best bet is to use the AMR_WB format.
However, the following code produces no exceptions, and I get no responses in the onError(t: Throwable) method, but the API is not returning any values at all in the onNext(value: StreamingRecognizeResponse) method.
If I change the format in .setEncoding() from FLAC or AMR_WB back to LINEAR16 everything works fine.
AudioEmitter.kt
fun start(
encoding: Int = AudioFormat.ENCODING_PCM_16BIT,
channel: Int = AudioFormat.CHANNEL_IN_MONO,
sampleRate: Int = 16000,
subscriber: (ByteString) -> Unit
)
MainActivity.kt
builder.streamingConfig = StreamingRecognitionConfig.newBuilder()
.setConfig(RecognitionConfig.newBuilder()
.setLanguageCode("en-US")
.setEncoding(RecognitionConfig.AudioEncoding.AMR_WB)
.setSampleRateHertz(16000)
.build())
.setInterimResults(true)
.setSingleUtterance(false)
.build()

Google won't recognize your data because you tell it the data is in FLAC or AMR_WB format, while you keep passing raw, uncompressed audio chunks that AudioRecord.read() produces.
Now, in order to make it work you have two choices. The first is to convert the data to the required format yourself, possibly using some third-party library. The second one is to use MediaRecorder from the Android library. Unfortunately, it supports only writing to a file-like destination, so you cannot simply replace AudioRecorder with it, but there's a workaround described in this answer.

How can we distinguish between audio and video under Android audioflinger

we can make distinguish between audio and video if we use android standard api to implement apk to play music/movie. no matter under libaudioflinger or decoder's lib.
when decode audio/video in awesomeplayer.cpp,we can judge the source data't type,audio? or video?
we can make distinguish the app's type under libaudioflinger
use getCallingPid()
Question:
how can we make distinguish 3rd's data source type(Audio?video?)under audioflinger?

yes audioflinger process the pcm data .
However if you want to set some parameters from Application then you can use AudioManager's setParametes API and then have a handling for that parameter in AudioFlinger .
AudioManager am = (AudioManager)context.getSystemService(context.AUDIO_SERVICE);
am.setParameters("key_value_pair");

Android - Playing mp3 directly from the input stream without writing into temp files

I'm creating an input stream to buffer and stream a mp3 from cloud .
URL url = new URL("http://xxxx.yyy.com/Demo.mp3");
InputStream inputStream = url.openStream();
Now how do i playback the mp3 from media player without using a temporary file to store it and read back from the same ? I'm developing for Android Lollipop

I'm pretty sure the MediaPlayer can handle remote URLs. Take a look at this example. Check the setDataSource method from the MediaPlayer class as well.
EDIT: Since you really really want to use an inputstream, I think you'll need to go low-level. Check the AudioTrack class. This SO answer might help. There are also a couple of issues here and here that might be relevant.

This problem persists even today !!! Check these link out https://code.google.com/p/android/issues/detail?id=29870 and
http://www.piterwilson.com/blog/2014/03/11/android-mediaplayer-not-quite-there-yet/ .
There is absolutely no way either to get access and control over the MediaPlayer buffer , neither to feed the buffered mp3 content stored in an byte array into MediaPplayer as an argument to play it . So People either convert the mp3 buffer to PCM and use AudioTrack to play it or write the byte array of the input stream into a local socket and make Mediaplayer read back using the socket file descriptor like mentioned this following link Audio stream buffering

The solution I'm using to feed binary data directly to MediaPlayer is to use ParcelFileDescriptor#createPipe() (API level 9) and MediaPlayer#setDataSource(java.io.FileDescriptor).
Here's sample code (untested):
ParcelFileDescriptor[] pipe = ParcelFileDescriptor.createPipe();
FileDescriptor fd = pipe[0].getFileDescriptor();
mediaPlayer.setDataSource(fd);
OutputStream out = new ParcelFileDescriptor.AutoCloseOutputStream(pipe[1]);
From this point on, whatever you write in the output stream will be received by the MediaPlayer. This is pretty fast since it uses a kernel FIFO to transfer data (no sockets, no TCP) and as far as I understand is fully in RAM (no actual files are used).

How to decode the H.264 video stream received from parcelfiledescriptor

I'm creating an Android application of live video streaming between two android phone. I've already established a socket connection between these devices. I'm capturing video in one device and send the stream to other device but currently I just want to save in the receiver side mobile device and save it. I'm recording using MediaRecorder in one device , so to stream to the receiver I,m using parcelfiledescriptor object by setting the data
Client side code
mediaRecorder.setAudioSource(MediaRecorder.AudioSource.CAMCORDER);
mediaRecorder.setVideoSource(MediaRecorder.VideoSource.CAMERA);
mediaRecorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
mediaRecorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
mediaRecorder.setVideoEncoder(MediaRecorder.VideoEncoder.H263);
mediaRecorder.setOutputFile(pfd.getFileDescriptor());
Receiver side code
pfd= ParcelFileDescriptor.fromSocket(s);
InputStream in = new FileInputStream(pfd.getFileDescriptor());
DataInputStream clientData = new DataInputStream(in);
OutputStream newDatabase = new FileOutputStream(file);
int available=in.available();
byte[] buffer = new byte[available];
int length;
while((length = in.read(buffer)) > 0)
{
newDatabase.write(buffer, 0, length);
}
newDatabase.close();
The video file is being created on the receiver side mobile, but it's not able to receive any bytes. So Do I've to decode the coming stream on the receiver side since the video stream sent is encoded while recording. So how can I decode the stream that is received ? I found some solution like MediaExtractor and MediaCodec...but will this work with live video capturing and moreover I'm testing on android version 2.3.6 GingerBread
Is it possible to decode the video stream from MediaCodec for version 2.3.6 or some other method is available ?

The video file is being created on the receiver side mobile, but it's not able to receive any bytes.
If I understand you right, you are getting no data from the socket. That is a separate problem, which has nothing to do with the video format, decoding or encoding.
To debug your sockets, it may be helpful to use a separate application which just dumps the recieved data. Once the data looks fine, you can go to the next step - decoding the video.
Second part of the problem is the video format. You are using mp4, which is not usable for streaming. Here is more info about the format structure. You can use mp4 to record a video into a local file and then transfer the whole file over socket somewhere, but true realtime streaming cannot be done because of the non-seekable nature of the socket (as described in the linked article). There is a block of metadata at the beginning of the file, which acts as a "table of contents" and without it, the previous data are just junk. The problem is, you can assemble a "table of contents" only after you got all the contents. But at that moment, the data was already sent through the socket and you cannot insert anything at its beginning.
There are few walkarounds, but that's just for your future research and I haven't used them yet.
The most intuitive way would be to switch from mp4 to mpeg-ts, a container designed for streaming. Take a look at a hidden constant in MediaRecorder.OutputFormat with value 8.
Another option is to pack the raw H.264 data into RTP/RTCP packets, which is again a protocol designed for streaming. Also your application would be able to stream to any device that support this protocol (for example a PC running VLC). To further reasearch, take a look at Spydroid IP camera, which does exactly the thing.