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Android: Simplify moved audio device module

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This CL performs some simplifications and cleanups of the moved audio code.
 * All JNI interaction now goes from the C++ audio manager calling into
   the Java audio manager. The calls back from the Java code to the C++
   audio manager are removed (this was related to caching audio parameters).
   It's simpler this way because the Java code is now unaware of the C++
   layer and it will be easier to make this into a Java interface.
 * A bunch of state was removed that was related to caching the audio parameters.
 * Some unused functions from audio manager was removed.
 * The Java audio manager no longer depends on ContextUtils, and the context has
   to be passed in externally instead. This is done because we want to get rid of
   ContextUtils eventually.
 * The selection of what AudioDeviceModule to create (AAudio, OpenSLES
   input/output is now exposed in the interface. The reason is that client should
   decide and create what they need explicitly instead of setting blacklists
   in static global WebRTC classes. This will be more modular long term.
 * Selection of what audio device module to create (OpenSLES combinations) no
   longer requires instantiating a C++ AudioManager and is done with static
   enumeration methods instead.

Bug: webrtc:7452
Change-Id: Iba29cf7447a1f6063abd9544d7315e10095167c8
Reviewed-on: https://webrtc-review.googlesource.com/63760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22542}
This commit is contained in:
Magnus Jedvert
2018-03-21 17:26:09 +01:00
committed by Commit Bot
parent 2955d82eca
commit 32362a6729
10 changed files with 249 additions and 427 deletions
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153
sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
View File

@ -20,10 +20,8 @@ import android.media.AudioTrack;
import android.os.Build;
import java.util.Timer;
import java.util.TimerTask;
import org.webrtc.ContextUtils;
import org.webrtc.Logging;
import org.webrtc.CalledByNative;
import org.webrtc.NativeClassQualifiedName;
// WebRtcAudioManager handles tasks that uses android.media.AudioManager.
// At construction, storeAudioParameters() is called and it retrieves
@ -39,10 +37,6 @@ class WebRtcAudioManager {
private static final String TAG = "WebRtcAudioManager";
// TODO(bugs.webrtc.org/8914): disabled by default until AAudio support has
// been completed. Goal is to always return false on Android O MR1 and higher.
private static final boolean blacklistDeviceForAAudioUsage = true;
// Use mono as default for both audio directions.
private static boolean useStereoOutput = false;
private static boolean useStereoInput = false;
@ -79,13 +73,15 @@ class WebRtcAudioManager {
// TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
@SuppressWarnings("NoSynchronizedMethodCheck")
public static synchronized boolean getStereoOutput() {
@CalledByNative
public synchronized boolean getStereoOutput() {
return useStereoOutput;
}
// TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
@SuppressWarnings("NoSynchronizedMethodCheck")
public static synchronized boolean getStereoInput() {
@CalledByNative
public synchronized boolean getStereoInput() {
return useStereoInput;
}
@ -150,42 +146,34 @@ class WebRtcAudioManager {
}
}
private final long nativeAudioManager;
private final AudioManager audioManager;
private boolean initialized = false;
private int nativeSampleRate;
private int nativeChannels;
private boolean hardwareAEC;
private boolean hardwareAGC;
private boolean hardwareNS;
private boolean lowLatencyOutput;
private boolean lowLatencyInput;
private boolean proAudio;
private boolean aAudio;
private int sampleRate;
private int outputChannels;
private int inputChannels;
private int outputBufferSize;
private int inputBufferSize;
private final int sampleRate;
private final int outputBufferSize;
private final int inputBufferSize;
private final VolumeLogger volumeLogger;
private boolean initialized = false;
@CalledByNative
WebRtcAudioManager(long nativeAudioManager) {
WebRtcAudioManager(Context context) {
Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
this.nativeAudioManager = nativeAudioManager;
audioManager =
(AudioManager) ContextUtils.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
this.audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
if (DEBUG) {
WebRtcAudioUtils.logDeviceInfo(TAG);
}
volumeLogger = new VolumeLogger(audioManager);
storeAudioParameters();
nativeCacheAudioParameters(nativeAudioManager, sampleRate, outputChannels, inputChannels,
hardwareAEC, hardwareAGC, hardwareNS, lowLatencyOutput, lowLatencyInput, proAudio, aAudio,
outputBufferSize, inputBufferSize);
this.volumeLogger = new VolumeLogger(audioManager);
final int outputChannels = getStereoOutput() ? 2 : 1;
final int inputChannels = getStereoInput() ? 2 : 1;
this.sampleRate = getNativeOutputSampleRate();
this.outputBufferSize = isLowLatencyOutputSupported(context)
? getLowLatencyOutputFramesPerBuffer()
: getMinOutputFrameSize(sampleRate, outputChannels);
this.inputBufferSize = isLowLatencyInputSupported(context)
? getLowLatencyInputFramesPerBuffer()
: getMinInputFrameSize(sampleRate, inputChannels);
WebRtcAudioUtils.logAudioState(TAG);
}
@ -216,7 +204,7 @@ class WebRtcAudioManager {
}
@CalledByNative
private boolean isDeviceBlacklistedForOpenSLESUsage() {
private static boolean isDeviceBlacklistedForOpenSLESUsage() {
boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
? blacklistDeviceForOpenSLESUsage
: WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
@ -226,64 +214,22 @@ class WebRtcAudioManager {
return blacklisted;
}
private void storeAudioParameters() {
outputChannels = getStereoOutput() ? 2 : 1;
inputChannels = getStereoInput() ? 2 : 1;
sampleRate = getNativeOutputSampleRate();
hardwareAEC = isAcousticEchoCancelerSupported();
// TODO(henrika): use of hardware AGC is no longer supported. Currently
// hardcoded to false. To be removed.
hardwareAGC = false;
hardwareNS = isNoiseSuppressorSupported();
lowLatencyOutput = isLowLatencyOutputSupported();
lowLatencyInput = isLowLatencyInputSupported();
proAudio = isProAudioSupported();
aAudio = isAAudioSupported();
outputBufferSize = lowLatencyOutput ? getLowLatencyOutputFramesPerBuffer()
: getMinOutputFrameSize(sampleRate, outputChannels);
inputBufferSize = lowLatencyInput ? getLowLatencyInputFramesPerBuffer()
: getMinInputFrameSize(sampleRate, inputChannels);
}
// Gets the current earpiece state.
private boolean hasEarpiece() {
return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
PackageManager.FEATURE_TELEPHONY);
}
// Returns true if low-latency audio output is supported.
private boolean isLowLatencyOutputSupported() {
return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
PackageManager.FEATURE_AUDIO_LOW_LATENCY);
@CalledByNative
public static boolean isLowLatencyOutputSupported(Context context) {
return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
}
// Returns true if low-latency audio input is supported.
// TODO(henrika): remove the hardcoded false return value when OpenSL ES
// input performance has been evaluated and tested more.
public boolean isLowLatencyInputSupported() {
@CalledByNative
public static boolean isLowLatencyInputSupported(Context context) {
// TODO(henrika): investigate if some sort of device list is needed here
// as well. The NDK doc states that: "As of API level 21, lower latency
// audio input is supported on select devices. To take advantage of this
// feature, first confirm that lower latency output is available".
return WebRtcAudioUtils.runningOnLollipopOrHigher() && isLowLatencyOutputSupported();
}
// Returns true if the device has professional audio level of functionality
// and therefore supports the lowest possible round-trip latency.
@TargetApi(23)
private boolean isProAudioSupported() {
return WebRtcAudioUtils.runningOnMarshmallowOrHigher()
&& ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
PackageManager.FEATURE_AUDIO_PRO);
}
// AAudio is supported on Androio Oreo MR1 (API 27) and higher.
// TODO(bugs.webrtc.org/8914): currently disabled by default.
private boolean isAAudioSupported() {
if (blacklistDeviceForAAudioUsage) {
Logging.w(TAG, "AAudio support is currently disabled on all devices!");
}
return !blacklistDeviceForAAudioUsage && WebRtcAudioUtils.runningOnOreoMR1OrHigher();
return WebRtcAudioUtils.runningOnLollipopOrHigher() && isLowLatencyOutputSupported(context);
}
// Returns the native output sample rate for this device's output stream.
@ -314,6 +260,11 @@ class WebRtcAudioManager {
return sampleRateHz;
}
@CalledByNative
int getSampleRate() {
return sampleRate;
}
@TargetApi(17)
private int getSampleRateOnJellyBeanMR10OrHigher() {
String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
@ -324,7 +275,6 @@ class WebRtcAudioManager {
// Returns the native output buffer size for low-latency output streams.
@TargetApi(17)
private int getLowLatencyOutputFramesPerBuffer() {
assertTrue(isLowLatencyOutputSupported());
if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
return DEFAULT_FRAME_PER_BUFFER;
}
@ -339,13 +289,26 @@ class WebRtcAudioManager {
// 2) explicit use (override) of a WebRTC based version must not be set,
// 3) the device must not be blacklisted for use of the effect, and
// 4) the UUID of the effect must be approved (some UUIDs can be excluded).
private static boolean isAcousticEchoCancelerSupported() {
@CalledByNative
boolean isAcousticEchoCancelerSupported() {
return WebRtcAudioEffects.canUseAcousticEchoCanceler();
}
private static boolean isNoiseSuppressorSupported() {
@CalledByNative
boolean isNoiseSuppressorSupported() {
return WebRtcAudioEffects.canUseNoiseSuppressor();
}
@CalledByNative
int getOutputBufferSize() {
return outputBufferSize;
}
@CalledByNative
int getInputBufferSize() {
return inputBufferSize;
}
// Returns the minimum output buffer size for Java based audio (AudioTrack).
// This size can also be used for OpenSL ES implementations on devices that
// lacks support of low-latency output.
@ -360,7 +323,6 @@ class WebRtcAudioManager {
// Returns the native input buffer size for input streams.
private int getLowLatencyInputFramesPerBuffer() {
assertTrue(isLowLatencyInputSupported());
return getLowLatencyOutputFramesPerBuffer();
}
@ -375,17 +337,4 @@ class WebRtcAudioManager {
sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
/ bytesPerFrame;
}
// 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");
}
}
@NativeClassQualifiedName("webrtc::android_adm::AudioManager")
private native void nativeCacheAudioParameters(long nativeAudioManager, int sampleRate,
int outputChannels, int inputChannels, boolean hardwareAEC, boolean hardwareAGC,
boolean hardwareNS, boolean lowLatencyOutput, boolean lowLatencyInput, boolean proAudio,
boolean aAudio, int outputBufferSize, int inputBufferSize);
}

5
sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
View File

@ -130,11 +130,6 @@ final class WebRtcAudioUtils {
public static boolean isNoiseSuppressorSupported() {
return WebRtcAudioEffects.canUseNoiseSuppressor();
}
// TODO(henrika): deprecated; remove when no longer used by any client.
public static boolean isAutomaticGainControlSupported() {
// Always return false here to avoid trying to use any built-in AGC.
return false;
}
// Call this method if the default handling of querying the native sample
// rate shall be overridden. Can be useful on some devices where the