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/*
* 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);
}
private static boolean isLowLatencyOutputSupported(Context context) {
return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
}
private 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;
}
}
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