webrtc 音频设备操作之opensl与jni
本节主要分享视频通话中android和ios上操作音频设备的方式,如调解音量大小,启用扬声器
处理音频设备代码目录:src/modules/audio_device/
ls modules/audio_device/
BUILD.gn audio_device_buffer.h audio_device_impl.cc dummy linux
DEPS audio_device_config.h audio_device_impl.h fine_audio_buffer.cc mac
OWNERS audio_device_data_observer.cc audio_device_name.cc fine_audio_buffer.h mock_audio_device_buffer.h
android audio_device_generic.cc audio_device_name.h fine_audio_buffer_unittest.cc win
audio_device_buffer.cc audio_device_generic.h audio_device_unittest.cc include第一种方式就是直接调用Android或ObjectC的API,代码如下(WebRtcAudioTrack.java):
// Get max possible volume index for a phone call audio stream.
private int getStreamMaxVolume() {
threadChecker.checkIsOnValidThread();
Logging.d(TAG, "getStreamMaxVolume");
assertTrue(audioManager != null);
return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
}
// Set current volume level for a phone call audio stream.
private boolean setStreamVolume(int volume) {
threadChecker.checkIsOnValidThread();
Logging.d(TAG, "setStreamVolume(" + volume + ")");
assertTrue(audioManager != null);
if (isVolumeFixed()) {
Logging.e(TAG, "The device implements a fixed volume policy.");
return false;
}
audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
return true;
}通过调用Android提供的AudioManager类来操作音频设备(webrtc是用C++写的,通过JNI来调用这些JAVA方法).
下面是IOS中操作音频设备的方式(audio_device_impl.cc):
int32_t AudioDeviceModuleImpl::StartPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (Playing()) {
return 0;
}
audio_device_buffer_.StartPlayout();
int32_t result = audio_device_->StartPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::StopPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopPlayout();
audio_device_buffer_.StopPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}既然WebRTC是用C++来实现的,可以用JNI来调用Android的API,那能不能不通过JNI调用JAVA,而是直接调用他底层的实现类库呢?带着这个疑问我们直接看音频设备初始化方法(audio_device_impl.cc):
int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
RTC_LOG(INFO) << __FUNCTION__;
// Dummy ADM implementations if build flags are set.
#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
audio_device_.reset(new AudioDeviceDummy());
RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
if (audio_device_) {
RTC_LOG(INFO) << "Will use file-playing dummy device.";
} else {
// Create a dummy device instead.
audio_device_.reset(new AudioDeviceDummy());
RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
}
// Real (non-dummy) ADM implementations.
#else
AudioLayer audio_layer(PlatformAudioLayer());
// Windows ADM implementation.
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
if ((audio_layer == kWindowsCoreAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
RTC_LOG(INFO) << "Attempting to use the Windows Core Audio APIs...";
if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
audio_device_.reset(new AudioDeviceWindowsCore());
RTC_LOG(INFO) << "Windows Core Audio APIs will be utilized";
}
}
#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#if defined(WEBRTC_ANDROID)
// Create an Android audio manager.
audio_manager_android_.reset(new AudioManager());
// Select best possible combination of audio layers.
if (audio_layer == kPlatformDefaultAudio) {
if (audio_manager_android_->IsAAudioSupported()) {
// Use of AAudio for both playout and recording has highest priority.
audio_layer = kAndroidAAudioAudio;
} else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
audio_manager_android_->IsLowLatencyRecordSupported()) {
// Use OpenSL ES for both playout and recording.
audio_layer = kAndroidOpenSLESAudio;
} else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
!audio_manager_android_->IsLowLatencyRecordSupported()) {
// Use OpenSL ES for output on devices that only supports the
// low-latency output audio path.
audio_layer = kAndroidJavaInputAndOpenSLESOutputAudio;
} else {
// Use Java-based audio in both directions when low-latency output is
// not supported.
audio_layer = kAndroidJavaAudio;
}
}
AudioManager* audio_manager = audio_manager_android_.get();
if (audio_layer == kAndroidJavaAudio) {
// Java audio for both input and output audio.
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidOpenSLESAudio) {
// OpenSL ES based audio for both input and output audio.
audio_device_.reset(
new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidJavaInputAndOpenSLESOutputAudio) {
// Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
// This combination provides low-latency output audio and at the same
// time support for HW AEC using the AudioRecord Java API.
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidAAudioAudio) {
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
// AAudio based audio for both input and output.
audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
audio_layer, audio_manager));
#endif
} else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
// Java audio for input and AAudio for output audio (i.e. mixed APIs).
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
audio_layer, audio_manager));
#endif
} else {
RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
audio_device_.reset(nullptr);
}
// END #if defined(WEBRTC_ANDROID)
// Linux ADM implementation.
// Note that, LINUX_ALSA is always defined by default when WEBRTC_LINUX is
// defined. LINUX_PULSE depends on the 'rtc_include_pulse_audio' build flag.
// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
// PulseAudio is the default selection.
#elif defined(WEBRTC_LINUX)
#if !defined(LINUX_PULSE)
// Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
// - kPlatformDefaultAudio => ALSA, and
// - kLinuxAlsaAudio => ALSA, and
// - kLinuxPulseAudio => Invalid selection.
RTC_LOG(WARNING) << "PulseAudio is disabled using build flag.";
if ((audio_layer == kLinuxAlsaAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
audio_device_.reset(new AudioDeviceLinuxALSA());
RTC_LOG(INFO) << "Linux ALSA APIs will be utilized.";
}
#else
// Build flag 'rtc_include_pulse_audio' is set to true (default). In this
// mode:
// - kPlatformDefaultAudio => PulseAudio, and
// - kLinuxPulseAudio => PulseAudio, and
// - kLinuxAlsaAudio => ALSA (supported but not default).
RTC_LOG(INFO) << "PulseAudio support is enabled.";
if ((audio_layer == kLinuxPulseAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
// Linux PulseAudio implementation is default.
audio_device_.reset(new AudioDeviceLinuxPulse());
RTC_LOG(INFO) << "Linux PulseAudio APIs will be utilized";
} else if (audio_layer == kLinuxAlsaAudio) {
audio_device_.reset(new AudioDeviceLinuxALSA());
RTC_LOG(WARNING) << "Linux ALSA APIs will be utilized.";
}
#endif // #if !defined(LINUX_PULSE)
#endif // #if defined(WEBRTC_LINUX)
// iOS ADM implementation.
#if defined(WEBRTC_IOS)
if (audio_layer == kPlatformDefaultAudio) {
audio_device_.reset(new ios_adm::AudioDeviceIOS());
RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
}
// END #if defined(WEBRTC_IOS)
// Mac OS X ADM implementation.
#elif defined(WEBRTC_MAC)
if (audio_layer == kPlatformDefaultAudio) {
audio_device_.reset(new AudioDeviceMac());
RTC_LOG(INFO) << "Mac OS X Audio APIs will be utilized.";
}
#endif // WEBRTC_MAC
// Dummy ADM implementation.
if (audio_layer == kDummyAudio) {
audio_device_.reset(new AudioDeviceDummy());
RTC_LOG(INFO) << "Dummy Audio APIs will be utilized.";
}
#endif // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
if (!audio_device_) {
RTC_LOG(LS_ERROR)
<< "Failed to create the platform specific ADM implementation.";
return -1;
}
return 0;
}ls modules/audio_device/android/
aaudio_player.cc audio_common.h audio_record_jni.cc ensure_initialized.cc opensles_player.h
aaudio_player.h audio_device_template.h audio_record_jni.h ensure_initialized.h opensles_recorder.cc
aaudio_recorder.cc audio_device_unittest.cc audio_track_jni.cc java opensles_recorder.h
aaudio_recorder.h audio_manager.cc audio_track_jni.h opensles_common.cc
aaudio_wrapper.cc audio_manager.h build_info.cc opensles_common.h
aaudio_wrapper.h audio_manager_unittest.cc build_info.h opensles_player.cc通过audio_manager_android_->IsLowLatencyPlayoutSupported() && audio_manager_android_->IsLowLatencyRecordSupported()来控制操作音频设备的方式,如果启用则直接调用OpenSles,OpenSles说明如下:
OpenSL ES 提供了可通过C++调用的C语言接口,这些接口功能基本类似于Android中通过JAVA调用的接口:
android.media.MediaPlayer
android.media.MediaRecorder
对于Android NDK开发来说,通过JNI调用OpenSL ES 提供的API,您即可实现大部分音频操作需要而不用在通过Java来操作.
**粗体** _斜体_ [链接](http://example.com) `代码` - 列表 > 引用。你还可以使用@来通知其他用户。