Develop Reference

Android MediaCodec realtime h264 encoding/decoding latency

I'm working with Android MediaCodec and use it for a realtime H264 encoding and decoding frames from camera. I use MediaCodec in synchronous manner and render the output to the Surface of decoder and everething works fine except that I have a long latency from a realtime, it takes 1.5-2 seconds and I'm very confused why is it so.
I measured a total time of encoding and decoding processes and it keeps around 50-65 milliseconds so I think the problem isn't in them.
I tried to change the configuration of the encoder but it didn't help and currently it configured like this:
val formatEncoder = MediaFormat.createVideoFormat("video/avc", 1920, 1080)
formatEncoder.setInteger(MediaFormat.KEY_FRAME_RATE, 30)
formatEncoder.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, 5)
formatEncoder.setInteger(MediaFormat.KEY_BIT_RATE, 1920 * 1080)
formatEncoder.setInteger(MediaFormat.KEY_COLOR_FORMAT, MediaCodecInfo.CodecCapabilities.COLOR_FormatSurface)
val encoder = MediaCodec.createEncoderByType("video/avc")
encoder.configure(formatEncoder, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE)
val inputSurface = encoder.createInputSurface() // I use it to send frames from camera to encoder
encoder.start()
Changing the configuration of the decoder also didn't help me at all and currently I configured it like this:
val formatDecoder = MediaFormat.createVideoFormat("video/avc", 1920, 1080)
val decoder = MediaCodec.createDecoderByType("video/avc")
decoder.configure(formatDecoder , outputSurface, null, 0) // I use outputSurface to render decoded frames into it
decoder.start()
I use the following timeouts for waiting for available encoder/decoder buffers I tried to reduce their values but it didn't help me and I left them like this:
var TIMEOUT_IN_BUFFER = 10000L // microseconds
var TIMEOUT_OUT_BUFFER = 10000L // microseconds
Also I measured the time of consuming the inputSurface a frame and this time takes 0.03-0.05 milliseconds so it isn't a bottleneck. Actually I measured all the places where a bottleneck could be, but I wasn't found anything and I think the problem is in the encoder or decoder itself or in their configurations, or maybe I should use some special routine for sending frames to encoding/decoding..
I also tried to use HW accelerated codec and it's the only thing that helped me, when I use it the latency reduces to ~ 500-800 milliseconds but it still doesn't fit me for a realtime streaming.
It seems to me that the encoder or decoder buffers several frames before start displaying them on the surface and eventually it leads to the latency and if it really so then how can I disable bufferization or reduce the time of it?
Please help me I'm stucking on this problem for about half a year and have no idea how to reduce the latency, I'm sure that it's possible because popular apps like Telegram, Viber, WhatsApp etc. work fine and without latency so what's the secret here?
UPD 07.07.2021:
I still haven't found a solution to get rid of the latency. I've tried to change h264 profiles, increase and decrease I-frame inteval, bitrate, framerate, but result the same, the only thing that hepls a little to reduce the latency - downgrade the resolution from 1920x1080 to e.g. 640x480, but this "solution" doesn't suit me because I want to encode/decode a realtime video with 1920x1080 resolution.
UPD 08.07.2021:
I found out that if I change the values of TIMEOUT_IN_BUFFER and TIMEOUT_OUT_BUFFER from 10_000L to 100_000L it decreases the latency a bit but increases the delay of showing the first frame quite a lot after start encoding/decoding process.

It's possible your encoder is producing B frames -- bilinear interpolation frames. They increase quality and latency, and are great for movies. But no good for low-latency applications.
Key frames = I (interframes)
Predicted frames = P (difference from previous frames)
Interpolated frames = B
A sequence of frames including B frames might look like this:
IBBBPBBBPBBBPBBBI
11111111
12345678901234567
The encoder must encode each P frame, and the decoder must decode it, before the preceding B frames make any sense. So in this example the frames get encoded out of order like this:
1 5 2 3 4 9 6 7 8 13 10 11 12 17 17 13 14 15
In this example the decoder can't handle frame 2 until the encoder has sent frame 5.
On the other hand, this sequence without B frames allows coding and decoding the frames in order.
IPPPPPPPPPPIPPPPPPPPP
Try using the Constrained Baseline Profile setting. It's designed for low latency and low power use. It suppresses B frames. I think this works.
mediaFormat.setInteger(
"profile",
CodecProfileLevel.AVCProfileConstrainedBaseline);

I believe android h264 decoder have latency (at-least in most cases i've tried). Probably that's why android developers added PARAMETER_KEY_LOW_LATENCY from API level 30.
However I could decrease the delay some frames by querying for the output some more times.
Reason: no idea. It's just result of boring trial and errors
int inputIndex = m_codec.dequeueInputBuffer(-1);// Pass in -1 here bc we don't have a playback time reference
if (inputIndex >= 0) {
ByteBuffer buffer;
if (android.os.Build.VERSION.SDK_INT >= android.os.Build.VERSION_CODES.LOLLIPOP) {
buffer = m_codec.getInputBuffer(inputIndex);
} else {
ByteBuffer[] bbuf = m_codec.getInputBuffers();
buffer = bbuf[inputIndex];
}
buffer.put(frame);
// tell the decoder to process the frame
m_codec.queueInputBuffer(inputIndex, 0, frame.length, 0, 0);
}
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
int outputIndex = m_codec.dequeueOutputBuffer(info, 0);
if (outputIndex >= 0) {
m_codec.releaseOutputBuffer(outputIndex, true);
}
outputIndex = m_codec.dequeueOutputBuffer(info, 0);
if (outputIndex >= 0) {
m_codec.releaseOutputBuffer(outputIndex, true);
}
outputIndex = m_codec.dequeueOutputBuffer(info, 0);
if (outputIndex >= 0) {
m_codec.releaseOutputBuffer(outputIndex, true);
}

You need to configure customized(or KEY_LOW_LATENCY if it is supported) low latency parameters for different cpu venders. It is a common problem for android phone.
Check this code https://github.com/moonlight-stream/moonlight-android/blob/master/app/src/main/java/com/limelight/binding/video/MediaCodecHelper.java

Android oboe c++ Some sounds distorted on playback

I'm using the Android oboe library for high performance audio in a music game.
In the assets folder I have 2 .raw files (both 48000Hz 16 bit PCM wavs and about 60kB)
std_kit_sn.raw
std_kit_ht.raw
These are loaded into memory as SoundRecordings and added to a Mixer. kSampleRateHz is 48000:
stdSN= SoundRecording::loadFromAssets(mAssetManager, "std_kit_sn.raw");
stdHT= SoundRecording::loadFromAssets(mAssetManager, "std_kit_ht.raw");
mMixer.addTrack(stdSN);
mMixer.addTrack(stdFT);
// Create a builder
AudioStreamBuilder builder;
builder.setFormat(AudioFormat::I16);
builder.setChannelCount(1);
builder.setSampleRate(kSampleRateHz);
builder.setCallback(this);
builder.setPerformanceMode(PerformanceMode::LowLatency);
builder.setSharingMode(SharingMode::Exclusive);
LOGD("After creating a builder");
// Open stream
Result result = builder.openStream(&mAudioStream);
if (result != Result::OK){
LOGE("Failed to open stream. Error: %s", convertToText(result));
}
LOGD("After openstream");
// Reduce stream latency by setting the buffer size to a multiple of the burst size
mAudioStream->setBufferSizeInFrames(mAudioStream->getFramesPerBurst() * 2);
// Start the stream
result = mAudioStream->requestStart();
if (result != Result::OK){
LOGE("Failed to start stream. Error: %s", convertToText(result));
}
LOGD("After starting stream");
They are called appropriately to play with standard code (as per Google tutorials) at required times:
stdSN->setPlaying(true);
stdHT->setPlaying(true); //Nasty Sound
The audio callback is standard (as per Google tutorials):
DataCallbackResult SoundFunctions::onAudioReady(AudioStream *mAudioStream, void *audioData, int32_t numFrames) {
// Play the stream
mMixer.renderAudio(static_cast<int16_t*>(audioData), numFrames);
return DataCallbackResult::Continue;
}
The std_kit_sn.raw plays fine. But std_kit_ht.raw has a nasty distortion. Both play with low latency. Why is one playing fine and the other has a nasty distortion?

I loaded your sample project and I believe the distortion you hear is caused by clipping/wraparound during mixing of sounds.
The Mixer object from the sample is a summing mixer. It just adds the values of each track together and outputs the sum.
You need to add some code to reduce the volume of each track to avoid exceeding the limits of an int16_t (although you're welcome to file a bug on the oboe project and I'll try to add this in an upcoming version). If you exceed this limit you'll get wraparound which is causing the distortion.
Additionally, your app is hardcoded to run at 22050 frames/sec. This will result in sub-optimal latency across most mobile devices because the stream is forced to upsample to the audio device's native frame rate. A better approach would be to leave the sample rate undefined when opening the stream - this will give you the optimal frame rate for the current audio device - then use a resampler on your source files to supply audio at this frame rate.

audio latency issues

In the application which I want to create, I face some technical obstacles. I have two music tracks in the application. For example, a user imports the music background as a first track. The second path is a voice recorded by the user to the rhythm of the first track played by the speaker device (or headphones). At this moment we face latency. After recording and playing back in the app, the user hears the loss of synchronisation between tracks, which occurs because of the microphone and speaker latencies.
Firstly, I try to detect the delay by filtering the input sound. I use android’s AudioRecord class, and the method read(). This method fills my short array with audio data.
I found that the initial values of this array are zeros so I decided to cut them out before I will start to write them into the output stream.
So I consider those zeros as a „warmup” latency of the microphone. Is this approach correct? This operation gives some results, but it doesn’t resolve the problem, and at this stage, I’m far away from that.
But the worse case is with the delay between starting the speakers and playing the music. This delay I cannot filter or detect. I tried to create some calibration feature which counts the delay. I play a „beep” sound through the speakers, and when I start to play it, I also begin to measure time. Then, I start recording and listen for this sound being detected by the microphone. When I recognise this sound in the app, I stop measuring time. I repeat this process several times, and the final value is the average from those results. That is how I try to measure the latency of the device. Now, when I have this value, I can simply shift the second track backwards to achieve synchronisation of both records (I will lose some initial milliseconds of the recording, but I skip this case, for now, there are some possibilities to fix it).
I thought that this approach would resolve the problem, but it turned out this is not as simple as I thought. I found two issues here:
1. Delay while playing two tracks simultaneously
2. Random in device audio latency.
The first: I play two tracks using AudioTrack class and I run method play() like this:
val firstTrack = //creating a track
val secondTrack = //creating a track
firstTrack.play()
secondTrack.play()
This code causes delays at the stage of playing tracks. Now, I don’t even have to think about latency while recording; I cannot play two tracks simultaneously without delays. I tested this with some external audio file (not recorded in my app) - I’m starting the same audio file using the code above, and I can see a delay. I also tried it with MediaPlayer class, and I have the same results. In this case, I even try to play tracks when callback OnPreparedListener invoke:
val firstTrack = //AudioPlayer
val secondTrack = //AudioPlayer
second.setOnPreparedListener {
first.start()
second.start()
}
And it doesn’t help.
I know that there is one more class provided by Android called SoundPool. According to the documentation, it can be better with playing tracks simultaneously, but I can’t use it because it supports only small audio files and that can't limit me.
How can I resolve this problem? How can I start playing two tracks precisely at the same time?
The second: Audio latency is not deterministic - sometimes it is smaller, and sometimes it’s huge, and it’s out of my hands. So measuring device latency can help but again - it cannot resolve the problem.
To sum up: is there any solution, which can give me exact latency per device (or app session?) or other triggers which detect actual delay, to provide the best synchronisation while playback two tracks at the same time?
Thank you in advance!

Synchronising audio for karaoke apps is tough. The main issue you seem to be facing is variable latency in the output stream.
This is almost certainly caused by "warm up" latency: the time it takes from hitting "play" on your backing track to the first frame of audio data being rendered by the audio device (e.g. headphones). This can have large variance and is difficult to measure.
The first (and easiest) thing to try is to use MODE_STREAM when constructing your AudioTrack and prime it with bufferSizeInBytes of data prior to calling play (more here). This should result in lower, more consistent "warm up" latency.
A better way is to use the Android NDK to have a continuously running audio stream which is just outputting silence until the moment you hit play, then start sending audio frames immediately. The only latency you have here is the continuous output latency.
If you decide to go down this route I recommend taking a look at the Oboe library (full disclosure: I am one of the authors).
To answer one of your specific questions...
Is there a way to calculate the latency of the audio output stream programatically?
Yes. The easiest way to explain this is with a code sample (this is C++ for the AAudio API but the principle is the same using Java AudioTrack):
// Get the index and time that a known audio frame was presented for playing
int64_t existingFrameIndex;
int64_t existingFramePresentationTime;
AAudioStream_getTimestamp(stream, CLOCK_MONOTONIC, &existingFrameIndex, &existingFramePresentationTime);
// Get the write index for the next audio frame
int64_t writeIndex = AAudioStream_getFramesWritten(stream);
// Calculate the number of frames between our known frame and the write index
int64_t frameIndexDelta = writeIndex - existingFrameIndex;
// Calculate the time which the next frame will be presented
int64_t frameTimeDelta = (frameIndexDelta * NANOS_PER_SECOND) / sampleRate_;
int64_t nextFramePresentationTime = existingFramePresentationTime + frameTimeDelta;
// Assume that the next frame will be written into the stream at the current time
int64_t nextFrameWriteTime = get_time_nanoseconds(CLOCK_MONOTONIC);
// Calculate the latency
*latencyMillis = (double) (nextFramePresentationTime - nextFrameWriteTime) / NANOS_PER_MILLISECOND;
A caveat: This method relies on accurate timestamps being reported by the audio hardware. I know this works on Google Pixel devices but have heard reports that it isn't so accurate on other devices so YMMV.

Following the answer of donturner, here's a Java version (that also uses other methods depending on the SDK version)
/** The audio latency has not been estimated yet */
private static long AUDIO_LATENCY_NOT_ESTIMATED = Long.MIN_VALUE+1;
/** The audio latency default value if we cannot estimate it */
private static long DEFAULT_AUDIO_LATENCY = 100L * 1000L * 1000L; // 100ms
/**
* Estimate the audio latency
*
* Not accurate at all, depends on SDK version, etc. But that's the best
* we can do.
*/
private static void estimateAudioLatency(AudioTrack track, long audioFramesWritten) {
long estimatedAudioLatency = AUDIO_LATENCY_NOT_ESTIMATED;
// First method. SDK >= 19.
if (Build.VERSION.SDK_INT >= 19 && track != null) {
AudioTimestamp audioTimestamp = new AudioTimestamp();
if (track.getTimestamp(audioTimestamp)) {
// Calculate the number of frames between our known frame and the write index
long frameIndexDelta = audioFramesWritten - audioTimestamp.framePosition;
// Calculate the time which the next frame will be presented
long frameTimeDelta = _framesToNanoSeconds(frameIndexDelta);
long nextFramePresentationTime = audioTimestamp.nanoTime + frameTimeDelta;
// Assume that the next frame will be written at the current time
long nextFrameWriteTime = System.nanoTime();
// Calculate the latency
estimatedAudioLatency = nextFramePresentationTime - nextFrameWriteTime;
}
}
// Second method. SDK >= 18.
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED && Build.VERSION.SDK_INT >= 18) {
Method getLatencyMethod;
try {
getLatencyMethod = AudioTrack.class.getMethod("getLatency", (Class<?>[]) null);
estimatedAudioLatency = (Integer) getLatencyMethod.invoke(track, (Object[]) null) * 1000000L;
} catch (Exception ignored) {}
}
// If no method has successfully gave us a value, let's try a third method
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED) {
AudioManager audioManager = (AudioManager) CRT.getInstance().getSystemService(Context.AUDIO_SERVICE);
try {
Method getOutputLatencyMethod = audioManager.getClass().getMethod("getOutputLatency", int.class);
estimatedAudioLatency = (Integer) getOutputLatencyMethod.invoke(audioManager, AudioManager.STREAM_MUSIC) * 1000000L;
} catch (Exception ignored) {}
}
// No method gave us a value. Let's use a default value. Better than nothing.
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED) {
estimatedAudioLatency = DEFAULT_AUDIO_LATENCY;
}
return estimatedAudioLatency
}
private static long _framesToNanoSeconds(long frames) {
return frames * 1000000000L / SAMPLE_RATE;
}

The android MediaPlayer class is notoriously slow to begin audio playback, I experienced an issue in an app I was creating where there was a greater than one second delay to begin playing an audio clip. I resolved it by switching to ExoPlayer which resulted in the playback starting within 100ms. I've also read that ffmpeg has even faster start audio startup time than ExoPlayer but I haven't used it so I can't make any promises.

Android camera2 jpeg framerate

I am trying to save image sequences with fixed framerates (preferably up to 30) on an android device with FULL capability for camera2 (Galaxy S7), but I am unable to a) get a steady framerate, b) reach even 20fps (with jpeg encoding). I already included the suggestions from Android camera2 capture burst is too slow.
The minimum frame duration for JPEG is 33.33 milliseconds (for resolutions below 1920x1080) according to
characteristics.get(CameraCharacteristics.SCALER_STREAM_CONFIGURATION_MAP).getOutputMinFrameDuration(ImageFormat.JPEG, size);
and the stallduration is 0ms for every size (similar for YUV_420_888).
My capture builder looks as follows:
captureBuilder.set(CaptureRequest.CONTROL_AE_MODE, CONTROL_AE_MODE_OFF);
captureBuilder.set(CaptureRequest.SENSOR_EXPOSURE_TIME, _exp_time);
captureBuilder.set(CaptureRequest.CONTROL_AE_LOCK, true);
captureBuilder.set(CaptureRequest.SENSOR_SENSITIVITY, _iso_value);
captureBuilder.set(CaptureRequest.LENS_FOCUS_DISTANCE, _foc_dist);
captureBuilder.set(CaptureRequest.CONTROL_AF_MODE, CONTROL_AF_MODE_OFF);
captureBuilder.set(CaptureRequest.CONTROL_AWB_MODE, _wb_value);
// https://stackoverflow.com/questions/29265126/android-camera2-capture-burst-is-too-slow
captureBuilder.set(CaptureRequest.EDGE_MODE,CaptureRequest.EDGE_MODE_OFF);
captureBuilder.set(CaptureRequest.COLOR_CORRECTION_ABERRATION_MODE, CaptureRequest.COLOR_CORRECTION_ABERRATION_MODE_OFF);
captureBuilder.set(CaptureRequest.NOISE_REDUCTION_MODE, CaptureRequest.NOISE_REDUCTION_MODE_OFF);
captureBuilder.set(CaptureRequest.CONTROL_AF_TRIGGER, CaptureRequest.CONTROL_AF_TRIGGER_CANCEL);
// Orientation
int rotation = getWindowManager().getDefaultDisplay().getRotation();
captureBuilder.set(CaptureRequest.JPEG_ORIENTATION,ORIENTATIONS.get(rotation));
Focus distance is set to 0.0 (inf), iso is set to 100, exposure-time 5ms. Whitebalance can be set to OFF/AUTO/ANY VALUE, it does not impact the times below.
I start the capture session with the following command:
session.setRepeatingRequest(_capReq.build(), captureListener, mBackgroundHandler);
Note: It does not make a difference if I request RepeatingRequest or RepeatingBurst..
In the preview (only texture surface attached), everything is at 30fps.
However, as soon as I attach an image reader (listener running on HandlerThread) which I instantiate like follows (without saving, only measuring time between frames):
reader = ImageReader.newInstance(_img_width, _img_height, ImageFormat.JPEG, 2);
reader.setOnImageAvailableListener(readerListener, mBackgroundHandler);
With time-measuring code:
ImageReader.OnImageAvailableListener readerListener = new ImageReader.OnImageAvailableListener() {
#Override
public void onImageAvailable(ImageReader myreader) {
Image image = null;
image = myreader.acquireNextImage();
if (image == null) {
return;
}
long curr = image.getTimestamp();
Log.d("curr- _last_ts", "" + ((curr - last_ts) / 1000000) + " ms");
last_ts = curr;
image.close();
}
}
I get periodically repeating time differences like this:
99 ms - 66 ms - 66 ms - 99 ms - 66 ms - 66 ms ...
I do not understand why these take double or triple the time that the stream configuration map advertised for jpeg? The exposure time is well below the frame duration of 33ms. Is there some other internal processing happening that I am not aware of?
I tried the same for the YUV_420_888 format, which resulted in constant time-differences of 33ms. The problem I have here is that the cellphone lacks the bandwidth to store the images fast enough (I tried the method described in How to save a YUV_420_888 image?). If you know of any method to compress or encode these images fast enough myself, please let me know.
Edit: From the documentation of getOutputStallDuration: "In other words, using a repeating YUV request would result in a steady frame rate (let's say it's 30 FPS). If a single JPEG request is submitted periodically, the frame rate will stay at 30 FPS (as long as we wait for the previous JPEG to return each time). If we try to submit a repeating YUV + JPEG request, then the frame rate will drop from 30 FPS." Does this imply that I need to periodically request a single capture()?
Edit2: From https://developer.android.com/reference/android/hardware/camera2/CaptureRequest.html: "The necessary information for the application, given the model above, is provided via the android.scaler.streamConfigurationMap field using getOutputMinFrameDuration(int, Size). These are used to determine the maximum frame rate / minimum frame duration that is possible for a given stream configuration.
Specifically, the application can use the following rules to determine the minimum frame duration it can request from the camera device:
Let the set of currently configured input/output streams be called S.
Find the minimum frame durations for each stream in S, by looking it up in android.scaler.streamConfigurationMap using getOutputMinFrameDuration(int, Size) (with its respective size/format). Let this set of frame durations be called F.
For any given request R, the minimum frame duration allowed for R is the maximum out of all values in F. Let the streams used in R be called S_r.
If none of the streams in S_r have a stall time (listed in getOutputStallDuration(int, Size) using its respective size/format), then the frame duration in F determines the steady state frame rate that the application will get if it uses R as a repeating request."

The JPEG output is by way not the fastest way to fetch frames. You can accomplish this a lot faster by drawing the frames directly onto a Quad using OpenGL.
For burst capture, a faster solution would be capturing the images to RAM without encoding them, then encoding and saving them asynchronously.
On this website you can find a lot of excellent code related to android multimedia in general.
This specific program uses OpenGL to fetch the pixel data from an MPEG video. It's not difficult to use the camera as input instead of a video. You can basically use the texture used in the CodecOutputSurface class from the mentioned program as output texture for your capture request.

A possible solution I found consists of using and dumping YUV without encoding it as JPEG in combination with a micro Sd-card that is able to save up to 95Mb per second. (I had the misconception that YUV images would be larger, so with a cellphone that has full support for the camera2-pipeline, the write speed should be the limiting factor.
With this setup, I was able to achieve the following stable rates:
1920x1080, 15fps (approx. 4Mb * 15 == 60Mb/sec)
960x720, 30fps. (approx. 1.5Mb * 30 == 45Mb/sec)
I then encode the images offline from YUV to PNG using a python script.

Android: ExoPlayer - Get current frame number from video

I'm developing an Android video app where I need to get the current frame number of the video being displayed while in pause mode.
I need to send my Server the frame number currently paused in video and get back a list of items regarding that frame/time, right now I'm sending the current paused time in milliseconds, but it doesn't work quite well, because the Server compare the time sent to a specific frame it calculated, based on the time, but sometimes the comparison is not exact.
I know you can get a bitmap from that frame if you use MediaMetaDataRetriever, and I did it but it returns bitmap image and what I need is an index.
I'm using ExoPlayer (I need that feature for MP4 and for HLS, too, if that matters).
Is there a way to get that info from the video?

I post a solution to my problem, In order to get the exact frame time I simply extended MediaCodecVideoTrackRenderer.java class from ExoPlayer library and used the value of lastOutputBufferTimestamp which is in function:
#Override
protected boolean processOutputBuffer(long positionUs, long elapsedRealtimeUs,
MediaCodec codec, ByteBuffer buffer, MediaCodec.BufferInfo bufferInfo, int bufferIndex,
boolean shouldSkip) {
boolean processed = super.processOutputBuffer(positionUs, elapsedRealtimeUs, codec, buffer,
bufferInfo, bufferIndex, shouldSkip);
if (!shouldSkip && processed) {
lastOutputBufferTimestamp = bufferInfo.presentationTimeUs;
}
return processed;
}
It does give me the exact time and not a rounded time from, last say, mPlayer.getDuration() or something like that.
If you have a constant FPS in your video you can calculate that by division and get the number of the frame.
It was simply enough for me to know the exact frame time.
I'm using ExoPlayer version r1.5.3 so I don't know if this solution will work for newer version since code has probably changed.