| /* |
| * Copyright 2018 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.content.Context; |
| import android.media.AudioAttributes; |
| import android.media.AudioDeviceInfo; |
| import android.media.AudioManager; |
| import android.os.Build; |
| import androidx.annotation.RequiresApi; |
| import java.util.concurrent.ScheduledExecutorService; |
| import org.webrtc.JniCommon; |
| import org.webrtc.Logging; |
| |
| /** |
| * AudioDeviceModule implemented using android.media.AudioRecord as input and |
| * android.media.AudioTrack as output. |
| */ |
| public class JavaAudioDeviceModule implements AudioDeviceModule { |
| private static final String TAG = "JavaAudioDeviceModule"; |
| |
| public static Builder builder(Context context) { |
| return new Builder(context); |
| } |
| |
| public static class Builder { |
| private final Context context; |
| private ScheduledExecutorService scheduler; |
| private final AudioManager audioManager; |
| private int inputSampleRate; |
| private int outputSampleRate; |
| private int audioSource = WebRtcAudioRecord.DEFAULT_AUDIO_SOURCE; |
| private int audioFormat = WebRtcAudioRecord.DEFAULT_AUDIO_FORMAT; |
| private AudioTrackErrorCallback audioTrackErrorCallback; |
| private AudioRecordErrorCallback audioRecordErrorCallback; |
| private SamplesReadyCallback samplesReadyCallback; |
| private AudioTrackStateCallback audioTrackStateCallback; |
| private AudioRecordStateCallback audioRecordStateCallback; |
| private boolean useHardwareAcousticEchoCanceler = isBuiltInAcousticEchoCancelerSupported(); |
| private boolean useHardwareNoiseSuppressor = isBuiltInNoiseSuppressorSupported(); |
| private boolean useStereoInput; |
| private boolean useStereoOutput; |
| private AudioAttributes audioAttributes; |
| private boolean useLowLatency; |
| private boolean enableVolumeLogger; |
| |
| private Builder(Context context) { |
| this.context = context; |
| this.audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE); |
| this.inputSampleRate = WebRtcAudioManager.getSampleRate(audioManager); |
| this.outputSampleRate = WebRtcAudioManager.getSampleRate(audioManager); |
| this.useLowLatency = false; |
| this.enableVolumeLogger = true; |
| } |
| |
| public Builder setScheduler(ScheduledExecutorService scheduler) { |
| this.scheduler = scheduler; |
| return this; |
| } |
| |
| /** |
| * Call this method if the default handling of querying the native sample rate shall be |
| * overridden. Can be useful on some devices where the available Android APIs are known to |
| * return invalid results. |
| */ |
| public Builder setSampleRate(int sampleRate) { |
| Logging.d(TAG, "Input/Output sample rate overridden to: " + sampleRate); |
| this.inputSampleRate = sampleRate; |
| this.outputSampleRate = sampleRate; |
| return this; |
| } |
| |
| /** |
| * Call this method to specifically override input sample rate. |
| */ |
| public Builder setInputSampleRate(int inputSampleRate) { |
| Logging.d(TAG, "Input sample rate overridden to: " + inputSampleRate); |
| this.inputSampleRate = inputSampleRate; |
| return this; |
| } |
| |
| /** |
| * Call this method to specifically override output sample rate. |
| */ |
| public Builder setOutputSampleRate(int outputSampleRate) { |
| Logging.d(TAG, "Output sample rate overridden to: " + outputSampleRate); |
| this.outputSampleRate = outputSampleRate; |
| return this; |
| } |
| |
| /** |
| * Call this to change the audio source. The argument should be one of the values from |
| * android.media.MediaRecorder.AudioSource. The default is AudioSource.VOICE_COMMUNICATION. |
| */ |
| public Builder setAudioSource(int audioSource) { |
| this.audioSource = audioSource; |
| return this; |
| } |
| |
| /** |
| * Call this to change the audio format. The argument should be one of the values from |
| * android.media.AudioFormat ENCODING_PCM_8BIT, ENCODING_PCM_16BIT or ENCODING_PCM_FLOAT. |
| * Default audio data format is PCM 16 bit per sample. |
| * Guaranteed to be supported by all devices. |
| */ |
| public Builder setAudioFormat(int audioFormat) { |
| this.audioFormat = audioFormat; |
| return this; |
| } |
| |
| /** |
| * Set a callback to retrieve errors from the AudioTrack. |
| */ |
| public Builder setAudioTrackErrorCallback(AudioTrackErrorCallback audioTrackErrorCallback) { |
| this.audioTrackErrorCallback = audioTrackErrorCallback; |
| return this; |
| } |
| |
| /** |
| * Set a callback to retrieve errors from the AudioRecord. |
| */ |
| public Builder setAudioRecordErrorCallback(AudioRecordErrorCallback audioRecordErrorCallback) { |
| this.audioRecordErrorCallback = audioRecordErrorCallback; |
| return this; |
| } |
| |
| /** |
| * Set a callback to listen to the raw audio input from the AudioRecord. |
| */ |
| public Builder setSamplesReadyCallback(SamplesReadyCallback samplesReadyCallback) { |
| this.samplesReadyCallback = samplesReadyCallback; |
| return this; |
| } |
| |
| /** |
| * Set a callback to retrieve information from the AudioTrack on when audio starts and stop. |
| */ |
| public Builder setAudioTrackStateCallback(AudioTrackStateCallback audioTrackStateCallback) { |
| this.audioTrackStateCallback = audioTrackStateCallback; |
| return this; |
| } |
| |
| /** |
| * Set a callback to retrieve information from the AudioRecord on when audio starts and stops. |
| */ |
| public Builder setAudioRecordStateCallback(AudioRecordStateCallback audioRecordStateCallback) { |
| this.audioRecordStateCallback = audioRecordStateCallback; |
| return this; |
| } |
| |
| /** |
| * Control if the built-in HW noise suppressor should be used or not. The default is on if it is |
| * supported. It is possible to query support by calling isBuiltInNoiseSuppressorSupported(). |
| */ |
| public Builder setUseHardwareNoiseSuppressor(boolean useHardwareNoiseSuppressor) { |
| if (useHardwareNoiseSuppressor && !isBuiltInNoiseSuppressorSupported()) { |
| Logging.e(TAG, "HW NS not supported"); |
| useHardwareNoiseSuppressor = false; |
| } |
| this.useHardwareNoiseSuppressor = useHardwareNoiseSuppressor; |
| return this; |
| } |
| |
| /** |
| * Control if the built-in HW acoustic echo canceler should be used or not. The default is on if |
| * it is supported. It is possible to query support by calling |
| * isBuiltInAcousticEchoCancelerSupported(). |
| */ |
| public Builder setUseHardwareAcousticEchoCanceler(boolean useHardwareAcousticEchoCanceler) { |
| if (useHardwareAcousticEchoCanceler && !isBuiltInAcousticEchoCancelerSupported()) { |
| Logging.e(TAG, "HW AEC not supported"); |
| useHardwareAcousticEchoCanceler = false; |
| } |
| this.useHardwareAcousticEchoCanceler = useHardwareAcousticEchoCanceler; |
| return this; |
| } |
| |
| /** |
| * Control if stereo input should be used or not. The default is mono. |
| */ |
| public Builder setUseStereoInput(boolean useStereoInput) { |
| this.useStereoInput = useStereoInput; |
| return this; |
| } |
| |
| /** |
| * Control if stereo output should be used or not. The default is mono. |
| */ |
| public Builder setUseStereoOutput(boolean useStereoOutput) { |
| this.useStereoOutput = useStereoOutput; |
| return this; |
| } |
| |
| /** |
| * Control if the low-latency mode should be used. The default is disabled. |
| */ |
| public Builder setUseLowLatency(boolean useLowLatency) { |
| this.useLowLatency = useLowLatency; |
| return this; |
| } |
| |
| /** |
| * Set custom {@link AudioAttributes} to use. |
| */ |
| public Builder setAudioAttributes(AudioAttributes audioAttributes) { |
| this.audioAttributes = audioAttributes; |
| return this; |
| } |
| |
| /** Disables the volume logger on the audio output track. */ |
| public Builder setEnableVolumeLogger(boolean enableVolumeLogger) { |
| this.enableVolumeLogger = enableVolumeLogger; |
| return this; |
| } |
| |
| /** |
| * Construct an AudioDeviceModule based on the supplied arguments. The caller takes ownership |
| * and is responsible for calling release(). |
| */ |
| public JavaAudioDeviceModule createAudioDeviceModule() { |
| Logging.d(TAG, "createAudioDeviceModule"); |
| if (useHardwareNoiseSuppressor) { |
| Logging.d(TAG, "HW NS will be used."); |
| } else { |
| if (isBuiltInNoiseSuppressorSupported()) { |
| Logging.d(TAG, "Overriding default behavior; now using WebRTC NS!"); |
| } |
| Logging.d(TAG, "HW NS will not be used."); |
| } |
| if (useHardwareAcousticEchoCanceler) { |
| Logging.d(TAG, "HW AEC will be used."); |
| } else { |
| if (isBuiltInAcousticEchoCancelerSupported()) { |
| Logging.d(TAG, "Overriding default behavior; now using WebRTC AEC!"); |
| } |
| Logging.d(TAG, "HW AEC will not be used."); |
| } |
| // Low-latency mode was introduced in API version 26, see |
| // https://developer.android.com/reference/android/media/AudioTrack#PERFORMANCE_MODE_LOW_LATENCY |
| final int MIN_LOW_LATENCY_SDK_VERSION = 26; |
| if (useLowLatency && Build.VERSION.SDK_INT >= MIN_LOW_LATENCY_SDK_VERSION) { |
| Logging.d(TAG, "Low latency mode will be used."); |
| } |
| ScheduledExecutorService executor = this.scheduler; |
| if (executor == null) { |
| executor = WebRtcAudioRecord.newDefaultScheduler(); |
| } |
| final WebRtcAudioRecord audioInput = new WebRtcAudioRecord(context, executor, audioManager, |
| audioSource, audioFormat, audioRecordErrorCallback, audioRecordStateCallback, |
| samplesReadyCallback, useHardwareAcousticEchoCanceler, useHardwareNoiseSuppressor); |
| final WebRtcAudioTrack audioOutput = |
| new WebRtcAudioTrack(context, audioManager, audioAttributes, audioTrackErrorCallback, |
| audioTrackStateCallback, useLowLatency, enableVolumeLogger); |
| return new JavaAudioDeviceModule(context, audioManager, audioInput, audioOutput, |
| inputSampleRate, outputSampleRate, useStereoInput, useStereoOutput); |
| } |
| } |
| |
| /* AudioRecord */ |
| // Audio recording error handler functions. |
| public enum AudioRecordStartErrorCode { |
| AUDIO_RECORD_START_EXCEPTION, |
| AUDIO_RECORD_START_STATE_MISMATCH, |
| } |
| |
| public static interface AudioRecordErrorCallback { |
| void onWebRtcAudioRecordInitError(String errorMessage); |
| void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage); |
| void onWebRtcAudioRecordError(String errorMessage); |
| } |
| |
| /** Called when audio recording starts and stops. */ |
| public static interface AudioRecordStateCallback { |
| void onWebRtcAudioRecordStart(); |
| void onWebRtcAudioRecordStop(); |
| } |
| |
| /** |
| * Contains audio sample information. |
| */ |
| public static class AudioSamples { |
| /** See {@link AudioRecord#getAudioFormat()} */ |
| private final int audioFormat; |
| /** See {@link AudioRecord#getChannelCount()} */ |
| private final int channelCount; |
| /** See {@link AudioRecord#getSampleRate()} */ |
| private final int sampleRate; |
| |
| private final byte[] data; |
| |
| public AudioSamples(int audioFormat, int channelCount, int sampleRate, byte[] data) { |
| this.audioFormat = audioFormat; |
| this.channelCount = channelCount; |
| this.sampleRate = sampleRate; |
| this.data = data; |
| } |
| |
| public int getAudioFormat() { |
| return audioFormat; |
| } |
| |
| public int getChannelCount() { |
| return channelCount; |
| } |
| |
| public int getSampleRate() { |
| return sampleRate; |
| } |
| |
| public byte[] getData() { |
| return data; |
| } |
| } |
| |
| /** Called when new audio samples are ready. This should only be set for debug purposes */ |
| public static interface SamplesReadyCallback { |
| void onWebRtcAudioRecordSamplesReady(AudioSamples samples); |
| } |
| |
| /* AudioTrack */ |
| // Audio playout/track error handler functions. |
| public enum AudioTrackStartErrorCode { |
| AUDIO_TRACK_START_EXCEPTION, |
| AUDIO_TRACK_START_STATE_MISMATCH, |
| } |
| |
| public static interface AudioTrackErrorCallback { |
| void onWebRtcAudioTrackInitError(String errorMessage); |
| void onWebRtcAudioTrackStartError(AudioTrackStartErrorCode errorCode, String errorMessage); |
| void onWebRtcAudioTrackError(String errorMessage); |
| } |
| |
| /** Called when audio playout starts and stops. */ |
| public static interface AudioTrackStateCallback { |
| void onWebRtcAudioTrackStart(); |
| void onWebRtcAudioTrackStop(); |
| } |
| |
| /** |
| * Returns true if the device supports built-in HW AEC, and the UUID is approved (some UUIDs can |
| * be excluded). |
| */ |
| public static boolean isBuiltInAcousticEchoCancelerSupported() { |
| return WebRtcAudioEffects.isAcousticEchoCancelerSupported(); |
| } |
| |
| /** |
| * Returns true if the device supports built-in HW NS, and the UUID is approved (some UUIDs can be |
| * excluded). |
| */ |
| public static boolean isBuiltInNoiseSuppressorSupported() { |
| return WebRtcAudioEffects.isNoiseSuppressorSupported(); |
| } |
| |
| private final Context context; |
| private final AudioManager audioManager; |
| private final WebRtcAudioRecord audioInput; |
| private final WebRtcAudioTrack audioOutput; |
| private final int inputSampleRate; |
| private final int outputSampleRate; |
| private final boolean useStereoInput; |
| private final boolean useStereoOutput; |
| |
| private final Object nativeLock = new Object(); |
| private long nativeAudioDeviceModule; |
| |
| private JavaAudioDeviceModule(Context context, AudioManager audioManager, |
| WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput, int inputSampleRate, |
| int outputSampleRate, boolean useStereoInput, boolean useStereoOutput) { |
| this.context = context; |
| this.audioManager = audioManager; |
| this.audioInput = audioInput; |
| this.audioOutput = audioOutput; |
| this.inputSampleRate = inputSampleRate; |
| this.outputSampleRate = outputSampleRate; |
| this.useStereoInput = useStereoInput; |
| this.useStereoOutput = useStereoOutput; |
| } |
| |
| @Override |
| public long getNativeAudioDeviceModulePointer() { |
| synchronized (nativeLock) { |
| if (nativeAudioDeviceModule == 0) { |
| nativeAudioDeviceModule = nativeCreateAudioDeviceModule(context, audioManager, audioInput, |
| audioOutput, inputSampleRate, outputSampleRate, useStereoInput, useStereoOutput); |
| } |
| return nativeAudioDeviceModule; |
| } |
| } |
| |
| @Override |
| public void release() { |
| synchronized (nativeLock) { |
| if (nativeAudioDeviceModule != 0) { |
| JniCommon.nativeReleaseRef(nativeAudioDeviceModule); |
| nativeAudioDeviceModule = 0; |
| } |
| } |
| } |
| |
| @Override |
| public void setSpeakerMute(boolean mute) { |
| Logging.d(TAG, "setSpeakerMute: " + mute); |
| audioOutput.setSpeakerMute(mute); |
| } |
| |
| @Override |
| public void setMicrophoneMute(boolean mute) { |
| Logging.d(TAG, "setMicrophoneMute: " + mute); |
| audioInput.setMicrophoneMute(mute); |
| } |
| |
| /** |
| * Start to prefer a specific {@link AudioDeviceInfo} device for recording. Typically this should |
| * only be used if a client gives an explicit option for choosing a physical device to record |
| * from. Otherwise the best-matching device for other parameters will be used. Calling after |
| * recording is started may cause a temporary interruption if the audio routing changes. |
| */ |
| @RequiresApi(Build.VERSION_CODES.M) |
| public void setPreferredInputDevice(AudioDeviceInfo preferredInputDevice) { |
| Logging.d(TAG, "setPreferredInputDevice: " + preferredInputDevice); |
| audioInput.setPreferredDevice(preferredInputDevice); |
| } |
| |
| private static native long nativeCreateAudioDeviceModule(Context context, |
| AudioManager audioManager, WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput, |
| int inputSampleRate, int outputSampleRate, boolean useStereoInput, boolean useStereoOutput); |
| } |