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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
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/WebRtcAudioManager.java | 124 |
1 files changed, 124 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java new file mode 100644 index 0000000000..506e33ffe4 --- /dev/null +++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java @@ -0,0 +1,124 @@ +/* + * 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.content.Context; +import android.content.pm.PackageManager; +import android.media.AudioFormat; +import android.media.AudioManager; +import android.media.AudioRecord; +import android.media.AudioTrack; +import android.os.Build; +import org.webrtc.Logging; +import org.webrtc.CalledByNative; + +/** + * This class contains static functions to query sample rate and input/output audio buffer sizes. + */ +class WebRtcAudioManager { + private static final String TAG = "WebRtcAudioManagerExternal"; + + private static final int DEFAULT_SAMPLE_RATE_HZ = 16000; + + // 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; + + private static final int DEFAULT_FRAME_PER_BUFFER = 256; + + @CalledByNative + static AudioManager getAudioManager(Context context) { + return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE); + } + + @CalledByNative + static int getOutputBufferSize( + Context context, AudioManager audioManager, int sampleRate, int numberOfOutputChannels) { + return isLowLatencyOutputSupported(context) + ? getLowLatencyFramesPerBuffer(audioManager) + : getMinOutputFrameSize(sampleRate, numberOfOutputChannels); + } + + @CalledByNative + static int getInputBufferSize( + Context context, AudioManager audioManager, int sampleRate, int numberOfInputChannels) { + return isLowLatencyInputSupported(context) + ? getLowLatencyFramesPerBuffer(audioManager) + : getMinInputFrameSize(sampleRate, numberOfInputChannels); + } + + @CalledByNative + static boolean isLowLatencyOutputSupported(Context context) { + return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY); + } + + @CalledByNative + 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 isLowLatencyOutputSupported(context); + } + + /** + * Returns the native input/output sample rate for this device's output stream. + */ + @CalledByNative + static int getSampleRate(AudioManager audioManager) { + // Override this if we're running on an old emulator image which only + // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE. + if (WebRtcAudioUtils.runningOnEmulator()) { + Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz."); + return 8000; + } + // Deliver best possible estimate based on default Android AudioManager APIs. + final int sampleRateHz = getSampleRateForApiLevel(audioManager); + Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz"); + return sampleRateHz; + } + + private static int getSampleRateForApiLevel(AudioManager audioManager) { + String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE); + return (sampleRateString == null) ? DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString); + } + + // Returns the native output buffer size for low-latency output streams. + private static int getLowLatencyFramesPerBuffer(AudioManager audioManager) { + String framesPerBuffer = + audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER); + return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer); + } + + // 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. + private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) { + final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8); + final int channelConfig = + (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO); + return AudioTrack.getMinBufferSize( + sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT) + / bytesPerFrame; + } + + // Returns the minimum input buffer size for Java based audio (AudioRecord). + // This size can calso be used for OpenSL ES implementations on devices that + // lacks support of low-latency input. + private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) { + final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8); + final int channelConfig = + (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO); + return AudioRecord.getMinBufferSize( + sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT) + / bytesPerFrame; + } +} |