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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java | 585 |
1 files changed, 585 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java new file mode 100644 index 0000000000..2b34e34013 --- /dev/null +++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java @@ -0,0 +1,585 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.audio; + +import android.annotation.TargetApi; +import android.content.Context; +import android.media.AudioAttributes; +import android.media.AudioFormat; +import android.media.AudioManager; +import android.media.AudioTrack; +import android.os.Build; +import android.os.Process; +import androidx.annotation.Nullable; +import java.nio.ByteBuffer; +import org.webrtc.CalledByNative; +import org.webrtc.Logging; +import org.webrtc.ThreadUtils; +import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback; +import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStartErrorCode; +import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStateCallback; +import org.webrtc.audio.LowLatencyAudioBufferManager; + +class WebRtcAudioTrack { + private static final String TAG = "WebRtcAudioTrackExternal"; + + // Default audio data format is PCM 16 bit per sample. + // Guaranteed to be supported by all devices. + private static final int BITS_PER_SAMPLE = 16; + + // Requested size of each recorded buffer provided to the client. + private static final int CALLBACK_BUFFER_SIZE_MS = 10; + + // Average number of callbacks per second. + private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS; + + // The AudioTrackThread is allowed to wait for successful call to join() + // but the wait times out afther this amount of time. + private static final long AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS = 2000; + + // By default, WebRTC creates audio tracks with a usage attribute + // corresponding to voice communications, such as telephony or VoIP. + private static final int DEFAULT_USAGE = AudioAttributes.USAGE_VOICE_COMMUNICATION; + + // Indicates the AudioTrack has started playing audio. + private static final int AUDIO_TRACK_START = 0; + + // Indicates the AudioTrack has stopped playing audio. + private static final int AUDIO_TRACK_STOP = 1; + + private long nativeAudioTrack; + private final Context context; + private final AudioManager audioManager; + private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker(); + + private ByteBuffer byteBuffer; + + private @Nullable final AudioAttributes audioAttributes; + private @Nullable AudioTrack audioTrack; + private @Nullable AudioTrackThread audioThread; + private final VolumeLogger volumeLogger; + + // Samples to be played are replaced by zeros if `speakerMute` is set to true. + // Can be used to ensure that the speaker is fully muted. + private volatile boolean speakerMute; + private byte[] emptyBytes; + private boolean useLowLatency; + private int initialBufferSizeInFrames; + + private final @Nullable AudioTrackErrorCallback errorCallback; + private final @Nullable AudioTrackStateCallback stateCallback; + + /** + * Audio thread which keeps calling AudioTrack.write() to stream audio. + * Data is periodically acquired from the native WebRTC layer using the + * nativeGetPlayoutData callback function. + * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority. + */ + private class AudioTrackThread extends Thread { + private volatile boolean keepAlive = true; + private LowLatencyAudioBufferManager bufferManager; + + public AudioTrackThread(String name) { + super(name); + bufferManager = new LowLatencyAudioBufferManager(); + } + + @Override + public void run() { + Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO); + Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo()); + assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING); + + // Audio playout has started and the client is informed about it. + doAudioTrackStateCallback(AUDIO_TRACK_START); + + // Fixed size in bytes of each 10ms block of audio data that we ask for + // using callbacks to the native WebRTC client. + final int sizeInBytes = byteBuffer.capacity(); + + while (keepAlive) { + // Get 10ms of PCM data from the native WebRTC client. Audio data is + // written into the common ByteBuffer using the address that was + // cached at construction. + nativeGetPlayoutData(nativeAudioTrack, sizeInBytes); + // Write data until all data has been written to the audio sink. + // Upon return, the buffer position will have been advanced to reflect + // the amount of data that was successfully written to the AudioTrack. + assertTrue(sizeInBytes <= byteBuffer.remaining()); + if (speakerMute) { + byteBuffer.clear(); + byteBuffer.put(emptyBytes); + byteBuffer.position(0); + } + int bytesWritten = audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING); + if (bytesWritten != sizeInBytes) { + Logging.e(TAG, "AudioTrack.write played invalid number of bytes: " + bytesWritten); + // If a write() returns a negative value, an error has occurred. + // Stop playing and report an error in this case. + if (bytesWritten < 0) { + keepAlive = false; + reportWebRtcAudioTrackError("AudioTrack.write failed: " + bytesWritten); + } + } + if (useLowLatency) { + bufferManager.maybeAdjustBufferSize(audioTrack); + } + // The byte buffer must be rewinded since byteBuffer.position() is + // increased at each call to AudioTrack.write(). If we don't do this, + // next call to AudioTrack.write() will fail. + byteBuffer.rewind(); + + // TODO(henrika): it is possible to create a delay estimate here by + // counting number of written frames and subtracting the result from + // audioTrack.getPlaybackHeadPosition(). + } + } + + // Stops the inner thread loop which results in calling AudioTrack.stop(). + // Does not block the calling thread. + public void stopThread() { + Logging.d(TAG, "stopThread"); + keepAlive = false; + } + } + + @CalledByNative + WebRtcAudioTrack(Context context, AudioManager audioManager) { + this(context, audioManager, null /* audioAttributes */, null /* errorCallback */, + null /* stateCallback */, false /* useLowLatency */, true /* enableVolumeLogger */); + } + + WebRtcAudioTrack(Context context, AudioManager audioManager, + @Nullable AudioAttributes audioAttributes, @Nullable AudioTrackErrorCallback errorCallback, + @Nullable AudioTrackStateCallback stateCallback, boolean useLowLatency, + boolean enableVolumeLogger) { + threadChecker.detachThread(); + this.context = context; + this.audioManager = audioManager; + this.audioAttributes = audioAttributes; + this.errorCallback = errorCallback; + this.stateCallback = stateCallback; + this.volumeLogger = enableVolumeLogger ? new VolumeLogger(audioManager) : null; + this.useLowLatency = useLowLatency; + Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo()); + } + + @CalledByNative + public void setNativeAudioTrack(long nativeAudioTrack) { + this.nativeAudioTrack = nativeAudioTrack; + } + + @CalledByNative + private int initPlayout(int sampleRate, int channels, double bufferSizeFactor) { + threadChecker.checkIsOnValidThread(); + Logging.d(TAG, + "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels + + ", bufferSizeFactor=" + bufferSizeFactor + ")"); + final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8); + byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND)); + Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity()); + emptyBytes = new byte[byteBuffer.capacity()]; + // Rather than passing the ByteBuffer with every callback (requiring + // the potentially expensive GetDirectBufferAddress) we simply have the + // the native class cache the address to the memory once. + nativeCacheDirectBufferAddress(nativeAudioTrack, byteBuffer); + + // Get the minimum buffer size required for the successful creation of an + // AudioTrack object to be created in the MODE_STREAM mode. + // Note that this size doesn't guarantee a smooth playback under load. + final int channelConfig = channelCountToConfiguration(channels); + final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig, + AudioFormat.ENCODING_PCM_16BIT) + * bufferSizeFactor); + Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes); + // For the streaming mode, data must be written to the audio sink in + // chunks of size (given by byteBuffer.capacity()) less than or equal + // to the total buffer size `minBufferSizeInBytes`. But, we have seen + // reports of "getMinBufferSize(): error querying hardware". Hence, it + // can happen that `minBufferSizeInBytes` contains an invalid value. + if (minBufferSizeInBytes < byteBuffer.capacity()) { + reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value."); + return -1; + } + + // Don't use low-latency mode when a bufferSizeFactor > 1 is used. When bufferSizeFactor > 1 + // we want to use a larger buffer to prevent underruns. However, low-latency mode would + // decrease the buffer size, which makes the bufferSizeFactor have no effect. + if (bufferSizeFactor > 1.0) { + useLowLatency = false; + } + + // Ensure that prevision audio session was stopped correctly before trying + // to create a new AudioTrack. + if (audioTrack != null) { + reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack."); + return -1; + } + try { + // Create an AudioTrack object and initialize its associated audio buffer. + // The size of this buffer determines how long an AudioTrack can play + // before running out of data. + if (useLowLatency && Build.VERSION.SDK_INT >= Build.VERSION_CODES.O) { + // On API level 26 or higher, we can use a low latency mode. + audioTrack = createAudioTrackOnOreoOrHigher( + sampleRate, channelConfig, minBufferSizeInBytes, audioAttributes); + } else { + // As we are on API level 21 or higher, it is possible to use a special AudioTrack + // constructor that uses AudioAttributes and AudioFormat as input. It allows us to + // supersede the notion of stream types for defining the behavior of audio playback, + // and to allow certain platforms or routing policies to use this information for more + // refined volume or routing decisions. + audioTrack = createAudioTrackBeforeOreo( + sampleRate, channelConfig, minBufferSizeInBytes, audioAttributes); + } + } catch (IllegalArgumentException e) { + reportWebRtcAudioTrackInitError(e.getMessage()); + releaseAudioResources(); + return -1; + } + + // It can happen that an AudioTrack is created but it was not successfully + // initialized upon creation. Seems to be the case e.g. when the maximum + // number of globally available audio tracks is exceeded. + if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) { + reportWebRtcAudioTrackInitError("Initialization of audio track failed."); + releaseAudioResources(); + return -1; + } + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) { + initialBufferSizeInFrames = audioTrack.getBufferSizeInFrames(); + } else { + initialBufferSizeInFrames = -1; + } + logMainParameters(); + logMainParametersExtended(); + return minBufferSizeInBytes; + } + + @CalledByNative + private boolean startPlayout() { + threadChecker.checkIsOnValidThread(); + if (volumeLogger != null) { + volumeLogger.start(); + } + Logging.d(TAG, "startPlayout"); + assertTrue(audioTrack != null); + assertTrue(audioThread == null); + + // Starts playing an audio track. + try { + audioTrack.play(); + } catch (IllegalStateException e) { + reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION, + "AudioTrack.play failed: " + e.getMessage()); + releaseAudioResources(); + return false; + } + if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) { + reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_STATE_MISMATCH, + "AudioTrack.play failed - incorrect state :" + audioTrack.getPlayState()); + releaseAudioResources(); + return false; + } + + // Create and start new high-priority thread which calls AudioTrack.write() + // and where we also call the native nativeGetPlayoutData() callback to + // request decoded audio from WebRTC. + audioThread = new AudioTrackThread("AudioTrackJavaThread"); + audioThread.start(); + return true; + } + + @CalledByNative + private boolean stopPlayout() { + threadChecker.checkIsOnValidThread(); + if (volumeLogger != null) { + volumeLogger.stop(); + } + Logging.d(TAG, "stopPlayout"); + assertTrue(audioThread != null); + logUnderrunCount(); + audioThread.stopThread(); + + Logging.d(TAG, "Stopping the AudioTrackThread..."); + audioThread.interrupt(); + if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) { + Logging.e(TAG, "Join of AudioTrackThread timed out."); + WebRtcAudioUtils.logAudioState(TAG, context, audioManager); + } + Logging.d(TAG, "AudioTrackThread has now been stopped."); + audioThread = null; + if (audioTrack != null) { + Logging.d(TAG, "Calling AudioTrack.stop..."); + try { + audioTrack.stop(); + Logging.d(TAG, "AudioTrack.stop is done."); + doAudioTrackStateCallback(AUDIO_TRACK_STOP); + } catch (IllegalStateException e) { + Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage()); + } + } + releaseAudioResources(); + return true; + } + + // Get max possible volume index for a phone call audio stream. + @CalledByNative + private int getStreamMaxVolume() { + threadChecker.checkIsOnValidThread(); + Logging.d(TAG, "getStreamMaxVolume"); + return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL); + } + + // Set current volume level for a phone call audio stream. + @CalledByNative + private boolean setStreamVolume(int volume) { + threadChecker.checkIsOnValidThread(); + Logging.d(TAG, "setStreamVolume(" + volume + ")"); + if (audioManager.isVolumeFixed()) { + Logging.e(TAG, "The device implements a fixed volume policy."); + return false; + } + audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0); + return true; + } + + /** Get current volume level for a phone call audio stream. */ + @CalledByNative + private int getStreamVolume() { + threadChecker.checkIsOnValidThread(); + Logging.d(TAG, "getStreamVolume"); + return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL); + } + + @CalledByNative + private int GetPlayoutUnderrunCount() { + if (Build.VERSION.SDK_INT >= 24) { + if (audioTrack != null) { + return audioTrack.getUnderrunCount(); + } else { + return -1; + } + } else { + return -2; + } + } + + private void logMainParameters() { + Logging.d(TAG, + "AudioTrack: " + + "session ID: " + audioTrack.getAudioSessionId() + ", " + + "channels: " + audioTrack.getChannelCount() + ", " + + "sample rate: " + audioTrack.getSampleRate() + + ", " + // Gain (>=1.0) expressed as linear multiplier on sample values. + + "max gain: " + AudioTrack.getMaxVolume()); + } + + private static void logNativeOutputSampleRate(int requestedSampleRateInHz) { + final int nativeOutputSampleRate = + AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL); + Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate); + if (requestedSampleRateInHz != nativeOutputSampleRate) { + Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native"); + } + } + + private static AudioAttributes getAudioAttributes(@Nullable AudioAttributes overrideAttributes) { + AudioAttributes.Builder attributesBuilder = + new AudioAttributes.Builder() + .setUsage(DEFAULT_USAGE) + .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH); + + if (overrideAttributes != null) { + if (overrideAttributes.getUsage() != AudioAttributes.USAGE_UNKNOWN) { + attributesBuilder.setUsage(overrideAttributes.getUsage()); + } + if (overrideAttributes.getContentType() != AudioAttributes.CONTENT_TYPE_UNKNOWN) { + attributesBuilder.setContentType(overrideAttributes.getContentType()); + } + + attributesBuilder.setFlags(overrideAttributes.getFlags()); + + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) { + attributesBuilder = applyAttributesOnQOrHigher(attributesBuilder, overrideAttributes); + } + } + return attributesBuilder.build(); + } + + // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input. + // It allows certain platforms or routing policies to use this information for more + // refined volume or routing decisions. + private static AudioTrack createAudioTrackBeforeOreo(int sampleRateInHz, int channelConfig, + int bufferSizeInBytes, @Nullable AudioAttributes overrideAttributes) { + Logging.d(TAG, "createAudioTrackBeforeOreo"); + logNativeOutputSampleRate(sampleRateInHz); + + // Create an audio track where the audio usage is for VoIP and the content type is speech. + return new AudioTrack(getAudioAttributes(overrideAttributes), + new AudioFormat.Builder() + .setEncoding(AudioFormat.ENCODING_PCM_16BIT) + .setSampleRate(sampleRateInHz) + .setChannelMask(channelConfig) + .build(), + bufferSizeInBytes, AudioTrack.MODE_STREAM, AudioManager.AUDIO_SESSION_ID_GENERATE); + } + + // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input. + // Use the low-latency mode to improve audio latency. Note that the low-latency mode may + // prevent effects (such as AEC) from working. Assuming AEC is working, the delay changes + // that happen in low-latency mode during the call will cause the AEC to perform worse. + // The behavior of the low-latency mode may be device dependent, use at your own risk. + @TargetApi(Build.VERSION_CODES.O) + private static AudioTrack createAudioTrackOnOreoOrHigher(int sampleRateInHz, int channelConfig, + int bufferSizeInBytes, @Nullable AudioAttributes overrideAttributes) { + Logging.d(TAG, "createAudioTrackOnOreoOrHigher"); + logNativeOutputSampleRate(sampleRateInHz); + + // Create an audio track where the audio usage is for VoIP and the content type is speech. + return new AudioTrack.Builder() + .setAudioAttributes(getAudioAttributes(overrideAttributes)) + .setAudioFormat(new AudioFormat.Builder() + .setEncoding(AudioFormat.ENCODING_PCM_16BIT) + .setSampleRate(sampleRateInHz) + .setChannelMask(channelConfig) + .build()) + .setBufferSizeInBytes(bufferSizeInBytes) + .setPerformanceMode(AudioTrack.PERFORMANCE_MODE_LOW_LATENCY) + .setTransferMode(AudioTrack.MODE_STREAM) + .setSessionId(AudioManager.AUDIO_SESSION_ID_GENERATE) + .build(); + } + + @TargetApi(Build.VERSION_CODES.Q) + private static AudioAttributes.Builder applyAttributesOnQOrHigher( + AudioAttributes.Builder builder, AudioAttributes overrideAttributes) { + return builder.setAllowedCapturePolicy(overrideAttributes.getAllowedCapturePolicy()); + } + + private void logBufferSizeInFrames() { + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) { + Logging.d(TAG, + "AudioTrack: " + // The effective size of the AudioTrack buffer that the app writes to. + + "buffer size in frames: " + audioTrack.getBufferSizeInFrames()); + } + } + + @CalledByNative + private int getBufferSizeInFrames() { + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) { + return audioTrack.getBufferSizeInFrames(); + } + return -1; + } + + @CalledByNative + private int getInitialBufferSizeInFrames() { + return initialBufferSizeInFrames; + } + + private void logBufferCapacityInFrames() { + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) { + Logging.d(TAG, + "AudioTrack: " + // Maximum size of the AudioTrack buffer in frames. + + "buffer capacity in frames: " + audioTrack.getBufferCapacityInFrames()); + } + } + + private void logMainParametersExtended() { + logBufferSizeInFrames(); + logBufferCapacityInFrames(); + } + + // Prints the number of underrun occurrences in the application-level write + // buffer since the AudioTrack was created. An underrun occurs if the app does + // not write audio data quickly enough, causing the buffer to underflow and a + // potential audio glitch. + // TODO(henrika): keep track of this value in the field and possibly add new + // UMA stat if needed. + private void logUnderrunCount() { + if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) { + Logging.d(TAG, "underrun count: " + audioTrack.getUnderrunCount()); + } + } + + // Helper method which throws an exception when an assertion has failed. + private static void assertTrue(boolean condition) { + if (!condition) { + throw new AssertionError("Expected condition to be true"); + } + } + + private int channelCountToConfiguration(int channels) { + return (channels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO); + } + + private static native void nativeCacheDirectBufferAddress( + long nativeAudioTrackJni, ByteBuffer byteBuffer); + private static native void nativeGetPlayoutData(long nativeAudioTrackJni, int bytes); + + // Sets all samples to be played out to zero if `mute` is true, i.e., + // ensures that the speaker is muted. + public void setSpeakerMute(boolean mute) { + Logging.w(TAG, "setSpeakerMute(" + mute + ")"); + speakerMute = mute; + } + + // Releases the native AudioTrack resources. + private void releaseAudioResources() { + Logging.d(TAG, "releaseAudioResources"); + if (audioTrack != null) { + audioTrack.release(); + audioTrack = null; + } + } + + private void reportWebRtcAudioTrackInitError(String errorMessage) { + Logging.e(TAG, "Init playout error: " + errorMessage); + WebRtcAudioUtils.logAudioState(TAG, context, audioManager); + if (errorCallback != null) { + errorCallback.onWebRtcAudioTrackInitError(errorMessage); + } + } + + private void reportWebRtcAudioTrackStartError( + AudioTrackStartErrorCode errorCode, String errorMessage) { + Logging.e(TAG, "Start playout error: " + errorCode + ". " + errorMessage); + WebRtcAudioUtils.logAudioState(TAG, context, audioManager); + if (errorCallback != null) { + errorCallback.onWebRtcAudioTrackStartError(errorCode, errorMessage); + } + } + + private void reportWebRtcAudioTrackError(String errorMessage) { + Logging.e(TAG, "Run-time playback error: " + errorMessage); + WebRtcAudioUtils.logAudioState(TAG, context, audioManager); + if (errorCallback != null) { + errorCallback.onWebRtcAudioTrackError(errorMessage); + } + } + + private void doAudioTrackStateCallback(int audioState) { + Logging.d(TAG, "doAudioTrackStateCallback: " + audioState); + if (stateCallback != null) { + if (audioState == WebRtcAudioTrack.AUDIO_TRACK_START) { + stateCallback.onWebRtcAudioTrackStart(); + } else if (audioState == WebRtcAudioTrack.AUDIO_TRACK_STOP) { + stateCallback.onWebRtcAudioTrackStop(); + } else { + Logging.e(TAG, "Invalid audio state"); + } + } + } +} |