Reland "Reland "Delete old Android ADM.""
This reverts commit 38a28603fd7b2eec46a362105b225dd6f08b4137.
Reason for revert: Attempt to reland, now that WebRTC dependency cycle has been broken.
Original change's description:
> Revert "Reland "Delete old Android ADM.""
>
> This reverts commit 6e4d7e606c4327eaa9298193e22794fcb9b30218.
>
> Reason for revert: Still breaks downstream build (though in a different way this time)
>
> Original change's description:
> > Reland "Delete old Android ADM."
> >
> > This is a reland of commit 4ec3e9c98873520b3171d40ab0426b2f05edbbd2
> >
> > Original change's description:
> > > Delete old Android ADM.
> > >
> > > The schedule move Android ADM code to sdk directory have been around
> > > for several years, but the old code still not delete.
> > >
> > > Bug: webrtc:7452
> > > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#37174}
> >
> > Bug: webrtc:7452
> > Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37236}
>
> Bug: webrtc:7452
> Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37242}
Bug: webrtc:7452
Change-Id: I6946d0fc28cf4c08387e451e6a07765f7410ce7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37356}
diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn
index 680a16d..f538149 100644
--- a/examples/androidnativeapi/BUILD.gn
+++ b/examples/androidnativeapi/BUILD.gn
@@ -15,7 +15,6 @@
deps = [
":resources",
- "//modules/audio_device:audio_device_java",
"//rtc_base:base_java",
"//sdk/android:camera_java",
"//sdk/android:surfaceviewrenderer_java",
diff --git a/examples/androidvoip/BUILD.gn b/examples/androidvoip/BUILD.gn
index 3120e06..b0ace2e 100644
--- a/examples/androidvoip/BUILD.gn
+++ b/examples/androidvoip/BUILD.gn
@@ -24,7 +24,6 @@
deps = [
":resources",
- "//modules/audio_device:audio_device_java",
"//rtc_base:base_java",
"//sdk/android:base_java",
"//sdk/android:java_audio_device_module_java",
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index b376955b..0624d62 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -249,39 +249,7 @@
"include/audio_device_data_observer.h",
]
if (is_android) {
- sources += [
- "android/audio_common.h",
- "android/audio_device_template.h",
- "android/audio_manager.cc",
- "android/audio_manager.h",
- "android/audio_record_jni.cc",
- "android/audio_record_jni.h",
- "android/audio_track_jni.cc",
- "android/audio_track_jni.h",
- "android/build_info.cc",
- "android/build_info.h",
- "android/opensles_common.cc",
- "android/opensles_common.h",
- "android/opensles_player.cc",
- "android/opensles_player.h",
- "android/opensles_recorder.cc",
- "android/opensles_recorder.h",
- ]
- libs = [
- "log",
- "OpenSLES",
- ]
- if (rtc_enable_android_aaudio) {
- sources += [
- "android/aaudio_player.cc",
- "android/aaudio_player.h",
- "android/aaudio_recorder.cc",
- "android/aaudio_recorder.h",
- "android/aaudio_wrapper.cc",
- "android/aaudio_wrapper.h",
- ]
- libs += [ "aaudio" ]
- }
+ deps += [ "../../sdk/android:native_api_audio_device_module" ]
if (build_with_mozilla) {
include_dirs += [
@@ -449,12 +417,6 @@
]
}
if (is_android) {
- sources += [
- "android/audio_device_unittest.cc",
- "android/audio_manager_unittest.cc",
- "android/ensure_initialized.cc",
- "android/ensure_initialized.h",
- ]
deps += [
"../../sdk/android:internal_jni",
"../../sdk/android:libjingle_peerconnection_java",
@@ -467,20 +429,3 @@
}
}
}
-
-if (!build_with_chromium && is_android) {
- rtc_android_library("audio_device_java") {
- sources = [
- "android/java/src/org/webrtc/voiceengine/BuildInfo.java",
- "android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
- "android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
- "android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
- "android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
- "android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
- ]
- deps = [
- "../../rtc_base:base_java",
- "//third_party/androidx:androidx_annotation_annotation_java",
- ]
- }
-}
diff --git a/modules/audio_device/DEPS b/modules/audio_device/DEPS
index 9cc627d..b0571de 100644
--- a/modules/audio_device/DEPS
+++ b/modules/audio_device/DEPS
@@ -9,5 +9,6 @@
],
"audio_device_impl\.cc": [
"+sdk/objc",
+ "+sdk/android",
],
}
diff --git a/modules/audio_device/android/aaudio_player.cc b/modules/audio_device/android/aaudio_player.cc
deleted file mode 100644
index 5257b2b..0000000
--- a/modules/audio_device/android/aaudio_player.cc
+++ /dev/null
@@ -1,228 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#include "modules/audio_device/android/aaudio_player.h"
-
-#include <memory>
-
-#include "api/array_view.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/fine_audio_buffer.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-
-namespace webrtc {
-
-enum AudioDeviceMessageType : uint32_t {
- kMessageOutputStreamDisconnected,
-};
-
-AAudioPlayer::AAudioPlayer(AudioManager* audio_manager)
- : main_thread_(rtc::Thread::Current()),
- aaudio_(audio_manager, AAUDIO_DIRECTION_OUTPUT, this) {
- RTC_LOG(LS_INFO) << "ctor";
- thread_checker_aaudio_.Detach();
-}
-
-AAudioPlayer::~AAudioPlayer() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- Terminate();
- RTC_LOG(LS_INFO) << "#detected underruns: " << underrun_count_;
-}
-
-int AAudioPlayer::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- if (aaudio_.audio_parameters().channels() == 2) {
- RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
- }
- return 0;
-}
-
-int AAudioPlayer::Terminate() {
- RTC_LOG(LS_INFO) << "Terminate";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- StopPlayout();
- return 0;
-}
-
-int AAudioPlayer::InitPlayout() {
- RTC_LOG(LS_INFO) << "InitPlayout";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!playing_);
- if (!aaudio_.Init()) {
- return -1;
- }
- initialized_ = true;
- return 0;
-}
-
-bool AAudioPlayer::PlayoutIsInitialized() const {
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- return initialized_;
-}
-
-int AAudioPlayer::StartPlayout() {
- RTC_LOG(LS_INFO) << "StartPlayout";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DCHECK(!playing_);
- if (!initialized_) {
- RTC_DLOG(LS_WARNING)
- << "Playout can not start since InitPlayout must succeed first";
- return 0;
- }
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetPlayout();
- }
- if (!aaudio_.Start()) {
- return -1;
- }
- underrun_count_ = aaudio_.xrun_count();
- first_data_callback_ = true;
- playing_ = true;
- return 0;
-}
-
-int AAudioPlayer::StopPlayout() {
- RTC_LOG(LS_INFO) << "StopPlayout";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- if (!initialized_ || !playing_) {
- return 0;
- }
- if (!aaudio_.Stop()) {
- RTC_LOG(LS_ERROR) << "StopPlayout failed";
- return -1;
- }
- thread_checker_aaudio_.Detach();
- initialized_ = false;
- playing_ = false;
- return 0;
-}
-
-bool AAudioPlayer::Playing() const {
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- return playing_;
-}
-
-void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_DLOG(LS_INFO) << "AttachAudioBuffer";
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- audio_device_buffer_ = audioBuffer;
- const AudioParameters audio_parameters = aaudio_.audio_parameters();
- audio_device_buffer_->SetPlayoutSampleRate(audio_parameters.sample_rate());
- audio_device_buffer_->SetPlayoutChannels(audio_parameters.channels());
- RTC_CHECK(audio_device_buffer_);
- // Create a modified audio buffer class which allows us to ask for any number
- // of samples (and not only multiple of 10ms) to match the optimal buffer
- // size per callback used by AAudio.
- fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
-}
-
-int AAudioPlayer::SpeakerVolumeIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
- RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
- // TODO(henrika): investigate if we can use a thread checker here. Initial
- // tests shows that this callback can sometimes be called on a unique thread
- // but according to the documentation it should be on the same thread as the
- // data callback.
- // RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
- if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
- // The stream is disconnected and any attempt to use it will return
- // AAUDIO_ERROR_DISCONNECTED.
- RTC_LOG(LS_WARNING) << "Output stream disconnected";
- // AAudio documentation states: "You should not close or reopen the stream
- // from the callback, use another thread instead". A message is therefore
- // sent to the main thread to do the restart operation.
- RTC_DCHECK(main_thread_);
- main_thread_->Post(RTC_FROM_HERE, this, kMessageOutputStreamDisconnected);
- }
-}
-
-aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
- int32_t num_frames) {
- RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
- // Log device id in first data callback to ensure that a valid device is
- // utilized.
- if (first_data_callback_) {
- RTC_LOG(LS_INFO) << "--- First output data callback: "
- "device id="
- << aaudio_.device_id();
- first_data_callback_ = false;
- }
-
- // Check if the underrun count has increased. If it has, increase the buffer
- // size by adding the size of a burst. It will reduce the risk of underruns
- // at the expense of an increased latency.
- // TODO(henrika): enable possibility to disable and/or tune the algorithm.
- const int32_t underrun_count = aaudio_.xrun_count();
- if (underrun_count > underrun_count_) {
- RTC_LOG(LS_ERROR) << "Underrun detected: " << underrun_count;
- underrun_count_ = underrun_count;
- aaudio_.IncreaseOutputBufferSize();
- }
-
- // Estimate latency between writing an audio frame to the output stream and
- // the time that same frame is played out on the output audio device.
- latency_millis_ = aaudio_.EstimateLatencyMillis();
- // TODO(henrika): use for development only.
- if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(LS_INFO) << "output latency: " << latency_millis_
- << ", num_frames: " << num_frames;
- }
-
- // Read audio data from the WebRTC source using the FineAudioBuffer object
- // and write that data into `audio_data` to be played out by AAudio.
- // Prime output with zeros during a short initial phase to avoid distortion.
- // TODO(henrika): do more work to figure out of if the initial forced silence
- // period is really needed.
- if (aaudio_.frames_written() < 50 * aaudio_.frames_per_burst()) {
- const size_t num_bytes =
- sizeof(int16_t) * aaudio_.samples_per_frame() * num_frames;
- memset(audio_data, 0, num_bytes);
- } else {
- fine_audio_buffer_->GetPlayoutData(
- rtc::MakeArrayView(static_cast<int16_t*>(audio_data),
- aaudio_.samples_per_frame() * num_frames),
- static_cast<int>(latency_millis_ + 0.5));
- }
-
- // TODO(henrika): possibly add trace here to be included in systrace.
- // See https://developer.android.com/studio/profile/systrace-commandline.html.
- return AAUDIO_CALLBACK_RESULT_CONTINUE;
-}
-
-void AAudioPlayer::OnMessage(rtc::Message* msg) {
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- switch (msg->message_id) {
- case kMessageOutputStreamDisconnected:
- HandleStreamDisconnected();
- break;
- }
-}
-
-void AAudioPlayer::HandleStreamDisconnected() {
- RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DLOG(LS_INFO) << "HandleStreamDisconnected";
- if (!initialized_ || !playing_) {
- return;
- }
- // Perform a restart by first closing the disconnected stream and then start
- // a new stream; this time using the new (preferred) audio output device.
- StopPlayout();
- InitPlayout();
- StartPlayout();
-}
-} // namespace webrtc
diff --git a/modules/audio_device/android/aaudio_player.h b/modules/audio_device/android/aaudio_player.h
deleted file mode 100644
index 4bf3ee3..0000000
--- a/modules/audio_device/android/aaudio_player.h
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
-
-#include <aaudio/AAudio.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/aaudio_wrapper.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "rtc_base/message_handler.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-class FineAudioBuffer;
-class AudioManager;
-
-// Implements low-latency 16-bit mono PCM audio output support for Android
-// using the C based AAudio API.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will DCHECK if any method is called on an invalid thread. Audio buffers
-// are requested on a dedicated high-priority thread owned by AAudio.
-//
-// The existing design forces the user to call InitPlayout() after StopPlayout()
-// to be able to call StartPlayout() again. This is in line with how the Java-
-// based implementation works.
-//
-// An audio stream can be disconnected, e.g. when an audio device is removed.
-// This implementation will restart the audio stream using the new preferred
-// device if such an event happens.
-//
-// Also supports automatic buffer-size adjustment based on underrun detections
-// where the internal AAudio buffer can be increased when needed. It will
-// reduce the risk of underruns (~glitches) at the expense of an increased
-// latency.
-class AAudioPlayer final : public AAudioObserverInterface,
- public rtc::MessageHandler {
- public:
- explicit AAudioPlayer(AudioManager* audio_manager);
- ~AAudioPlayer();
-
- int Init();
- int Terminate();
-
- int InitPlayout();
- bool PlayoutIsInitialized() const;
-
- int StartPlayout();
- int StopPlayout();
- bool Playing() const;
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- // Not implemented in AAudio.
- int SpeakerVolumeIsAvailable(bool& available); // NOLINT
- int SetSpeakerVolume(uint32_t volume) { return -1; }
- int SpeakerVolume(uint32_t& volume) const { return -1; } // NOLINT
- int MaxSpeakerVolume(uint32_t& maxVolume) const { return -1; } // NOLINT
- int MinSpeakerVolume(uint32_t& minVolume) const { return -1; } // NOLINT
-
- protected:
- // AAudioObserverInterface implementation.
-
- // For an output stream, this function should render and write `num_frames`
- // of data in the streams current data format to the `audio_data` buffer.
- // Called on a real-time thread owned by AAudio.
- aaudio_data_callback_result_t OnDataCallback(void* audio_data,
- int32_t num_frames) override;
- // AAudio calls this functions if any error occurs on a callback thread.
- // Called on a real-time thread owned by AAudio.
- void OnErrorCallback(aaudio_result_t error) override;
-
- // rtc::MessageHandler used for restart messages from the error-callback
- // thread to the main (creating) thread.
- void OnMessage(rtc::Message* msg) override;
-
- private:
- // Closes the existing stream and starts a new stream.
- void HandleStreamDisconnected();
-
- // Ensures that methods are called from the same thread as this object is
- // created on.
- SequenceChecker main_thread_checker_;
-
- // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
- // real-time thread owned by AAudio. Detached during construction of this
- // object.
- SequenceChecker thread_checker_aaudio_;
-
- // The thread on which this object is created on.
- rtc::Thread* main_thread_;
-
- // Wraps all AAudio resources. Contains an output stream using the default
- // output audio device. Can be accessed on both the main thread and the
- // real-time thread owned by AAudio. See separate AAudio documentation about
- // thread safety.
- AAudioWrapper aaudio_;
-
- // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
- // in chunks of 10ms. It then allows for this data to be pulled in
- // a finer or coarser granularity. I.e. interacting with this class instead
- // of directly with the AudioDeviceBuffer one can ask for any number of
- // audio data samples.
- // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
- // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192
- // in each callback (once every 4th ms). This class can then ask for 192 and
- // the FineAudioBuffer will ask WebRTC for new data approximately only every
- // second callback and also cache non-utilized audio.
- std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
-
- // Counts number of detected underrun events reported by AAudio.
- int32_t underrun_count_ = 0;
-
- // True only for the first data callback in each audio session.
- bool first_data_callback_ = true;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create().
- AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) =
- nullptr;
-
- bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false;
- bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false;
-
- // Estimated latency between writing an audio frame to the output stream and
- // the time that same frame is played out on the output audio device.
- double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
diff --git a/modules/audio_device/android/aaudio_recorder.cc b/modules/audio_device/android/aaudio_recorder.cc
deleted file mode 100644
index 4757cf8..0000000
--- a/modules/audio_device/android/aaudio_recorder.cc
+++ /dev/null
@@ -1,220 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#include "modules/audio_device/android/aaudio_recorder.h"
-
-#include <memory>
-
-#include "api/array_view.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/fine_audio_buffer.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/time_utils.h"
-
-namespace webrtc {
-
-enum AudioDeviceMessageType : uint32_t {
- kMessageInputStreamDisconnected,
-};
-
-AAudioRecorder::AAudioRecorder(AudioManager* audio_manager)
- : main_thread_(rtc::Thread::Current()),
- aaudio_(audio_manager, AAUDIO_DIRECTION_INPUT, this) {
- RTC_LOG(LS_INFO) << "ctor";
- thread_checker_aaudio_.Detach();
-}
-
-AAudioRecorder::~AAudioRecorder() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
- RTC_LOG(LS_INFO) << "detected owerflows: " << overflow_count_;
-}
-
-int AAudioRecorder::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (aaudio_.audio_parameters().channels() == 2) {
- RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
- }
- return 0;
-}
-
-int AAudioRecorder::Terminate() {
- RTC_LOG(LS_INFO) << "Terminate";
- RTC_DCHECK(thread_checker_.IsCurrent());
- StopRecording();
- return 0;
-}
-
-int AAudioRecorder::InitRecording() {
- RTC_LOG(LS_INFO) << "InitRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!recording_);
- if (!aaudio_.Init()) {
- return -1;
- }
- initialized_ = true;
- return 0;
-}
-
-int AAudioRecorder::StartRecording() {
- RTC_LOG(LS_INFO) << "StartRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(initialized_);
- RTC_DCHECK(!recording_);
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetPlayout();
- }
- if (!aaudio_.Start()) {
- return -1;
- }
- overflow_count_ = aaudio_.xrun_count();
- first_data_callback_ = true;
- recording_ = true;
- return 0;
-}
-
-int AAudioRecorder::StopRecording() {
- RTC_LOG(LS_INFO) << "StopRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_ || !recording_) {
- return 0;
- }
- if (!aaudio_.Stop()) {
- return -1;
- }
- thread_checker_aaudio_.Detach();
- initialized_ = false;
- recording_ = false;
- return 0;
-}
-
-void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(LS_INFO) << "AttachAudioBuffer";
- RTC_DCHECK(thread_checker_.IsCurrent());
- audio_device_buffer_ = audioBuffer;
- const AudioParameters audio_parameters = aaudio_.audio_parameters();
- audio_device_buffer_->SetRecordingSampleRate(audio_parameters.sample_rate());
- audio_device_buffer_->SetRecordingChannels(audio_parameters.channels());
- RTC_CHECK(audio_device_buffer_);
- // Create a modified audio buffer class which allows us to deliver any number
- // of samples (and not only multiples of 10ms which WebRTC uses) to match the
- // native AAudio buffer size.
- fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
-}
-
-int AAudioRecorder::EnableBuiltInAEC(bool enable) {
- RTC_LOG(LS_INFO) << "EnableBuiltInAEC: " << enable;
- RTC_LOG(LS_ERROR) << "Not implemented";
- return -1;
-}
-
-int AAudioRecorder::EnableBuiltInAGC(bool enable) {
- RTC_LOG(LS_INFO) << "EnableBuiltInAGC: " << enable;
- RTC_LOG(LS_ERROR) << "Not implemented";
- return -1;
-}
-
-int AAudioRecorder::EnableBuiltInNS(bool enable) {
- RTC_LOG(LS_INFO) << "EnableBuiltInNS: " << enable;
- RTC_LOG(LS_ERROR) << "Not implemented";
- return -1;
-}
-
-void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
- RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
- // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
- if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
- // The stream is disconnected and any attempt to use it will return
- // AAUDIO_ERROR_DISCONNECTED..
- RTC_LOG(LS_WARNING) << "Input stream disconnected => restart is required";
- // AAudio documentation states: "You should not close or reopen the stream
- // from the callback, use another thread instead". A message is therefore
- // sent to the main thread to do the restart operation.
- RTC_DCHECK(main_thread_);
- main_thread_->Post(RTC_FROM_HERE, this, kMessageInputStreamDisconnected);
- }
-}
-
-// Read and process `num_frames` of data from the `audio_data` buffer.
-// TODO(henrika): possibly add trace here to be included in systrace.
-// See https://developer.android.com/studio/profile/systrace-commandline.html.
-aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
- void* audio_data,
- int32_t num_frames) {
- // TODO(henrika): figure out why we sometimes hit this one.
- // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
- // RTC_LOG(LS_INFO) << "OnDataCallback: " << num_frames;
- // Drain the input buffer at first callback to ensure that it does not
- // contain any old data. Will also ensure that the lowest possible latency
- // is obtained.
- if (first_data_callback_) {
- RTC_LOG(LS_INFO) << "--- First input data callback: "
- "device id="
- << aaudio_.device_id();
- aaudio_.ClearInputStream(audio_data, num_frames);
- first_data_callback_ = false;
- }
- // Check if the overflow counter has increased and if so log a warning.
- // TODO(henrika): possible add UMA stat or capacity extension.
- const int32_t overflow_count = aaudio_.xrun_count();
- if (overflow_count > overflow_count_) {
- RTC_LOG(LS_ERROR) << "Overflow detected: " << overflow_count;
- overflow_count_ = overflow_count;
- }
- // Estimated time between an audio frame was recorded by the input device and
- // it can read on the input stream.
- latency_millis_ = aaudio_.EstimateLatencyMillis();
- // TODO(henrika): use for development only.
- if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(LS_INFO) << "input latency: " << latency_millis_
- << ", num_frames: " << num_frames;
- }
- // Copy recorded audio in `audio_data` to the WebRTC sink using the
- // FineAudioBuffer object.
- fine_audio_buffer_->DeliverRecordedData(
- rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
- aaudio_.samples_per_frame() * num_frames),
- static_cast<int>(latency_millis_ + 0.5));
-
- return AAUDIO_CALLBACK_RESULT_CONTINUE;
-}
-
-void AAudioRecorder::OnMessage(rtc::Message* msg) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- switch (msg->message_id) {
- case kMessageInputStreamDisconnected:
- HandleStreamDisconnected();
- break;
- default:
- RTC_LOG(LS_ERROR) << "Invalid message id: " << msg->message_id;
- break;
- }
-}
-
-void AAudioRecorder::HandleStreamDisconnected() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_LOG(LS_INFO) << "HandleStreamDisconnected";
- if (!initialized_ || !recording_) {
- return;
- }
- // Perform a restart by first closing the disconnected stream and then start
- // a new stream; this time using the new (preferred) audio input device.
- // TODO(henrika): resolve issue where a one restart attempt leads to a long
- // sequence of new calls to OnErrorCallback().
- // See b/73148976 for details.
- StopRecording();
- InitRecording();
- StartRecording();
-}
-} // namespace webrtc
diff --git a/modules/audio_device/android/aaudio_recorder.h b/modules/audio_device/android/aaudio_recorder.h
deleted file mode 100644
index d0ad6be..0000000
--- a/modules/audio_device/android/aaudio_recorder.h
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
-
-#include <aaudio/AAudio.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/aaudio_wrapper.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "rtc_base/message_handler.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-class FineAudioBuffer;
-class AudioManager;
-
-// Implements low-latency 16-bit mono PCM audio input support for Android
-// using the C based AAudio API.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
-// are delivered on a dedicated high-priority thread owned by AAudio.
-//
-// The existing design forces the user to call InitRecording() after
-// StopRecording() to be able to call StartRecording() again. This is in line
-// with how the Java- based implementation works.
-//
-// TODO(henrika): add comments about device changes and adaptive buffer
-// management.
-class AAudioRecorder : public AAudioObserverInterface,
- public rtc::MessageHandler {
- public:
- explicit AAudioRecorder(AudioManager* audio_manager);
- ~AAudioRecorder();
-
- int Init();
- int Terminate();
-
- int InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
-
- int StartRecording();
- int StopRecording();
- bool Recording() const { return recording_; }
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- double latency_millis() const { return latency_millis_; }
-
- // TODO(henrika): add support using AAudio APIs when available.
- int EnableBuiltInAEC(bool enable);
- int EnableBuiltInAGC(bool enable);
- int EnableBuiltInNS(bool enable);
-
- protected:
- // AAudioObserverInterface implementation.
-
- // For an input stream, this function should read `num_frames` of recorded
- // data, in the stream's current data format, from the `audio_data` buffer.
- // Called on a real-time thread owned by AAudio.
- aaudio_data_callback_result_t OnDataCallback(void* audio_data,
- int32_t num_frames) override;
-
- // AAudio calls this function if any error occurs on a callback thread.
- // Called on a real-time thread owned by AAudio.
- void OnErrorCallback(aaudio_result_t error) override;
-
- // rtc::MessageHandler used for restart messages.
- void OnMessage(rtc::Message* msg) override;
-
- private:
- // Closes the existing stream and starts a new stream.
- void HandleStreamDisconnected();
-
- // Ensures that methods are called from the same thread as this object is
- // created on.
- SequenceChecker thread_checker_;
-
- // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
- // real-time thread owned by AAudio. Detached during construction of this
- // object.
- SequenceChecker thread_checker_aaudio_;
-
- // The thread on which this object is created on.
- rtc::Thread* main_thread_;
-
- // Wraps all AAudio resources. Contains an input stream using the default
- // input audio device.
- AAudioWrapper aaudio_;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
- AudioDeviceBuffer* audio_device_buffer_ = nullptr;
-
- bool initialized_ = false;
- bool recording_ = false;
-
- // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
- // chunks of audio.
- std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
-
- // Counts number of detected overflow events reported by AAudio.
- int32_t overflow_count_ = 0;
-
- // Estimated time between an audio frame was recorded by the input device and
- // it can read on the input stream.
- double latency_millis_ = 0;
-
- // True only for the first data callback in each audio session.
- bool first_data_callback_ = true;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
diff --git a/modules/audio_device/android/aaudio_wrapper.cc b/modules/audio_device/android/aaudio_wrapper.cc
deleted file mode 100644
index 3d824b5..0000000
--- a/modules/audio_device/android/aaudio_wrapper.cc
+++ /dev/null
@@ -1,499 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#include "modules/audio_device/android/aaudio_wrapper.h"
-
-#include "modules/audio_device/android/audio_manager.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/strings/string_builder.h"
-#include "rtc_base/time_utils.h"
-
-#define LOG_ON_ERROR(op) \
- do { \
- aaudio_result_t result = (op); \
- if (result != AAUDIO_OK) { \
- RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
- } \
- } while (0)
-
-#define RETURN_ON_ERROR(op, ...) \
- do { \
- aaudio_result_t result = (op); \
- if (result != AAUDIO_OK) { \
- RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
- return __VA_ARGS__; \
- } \
- } while (0)
-
-namespace webrtc {
-
-namespace {
-
-const char* DirectionToString(aaudio_direction_t direction) {
- switch (direction) {
- case AAUDIO_DIRECTION_OUTPUT:
- return "OUTPUT";
- case AAUDIO_DIRECTION_INPUT:
- return "INPUT";
- default:
- return "UNKNOWN";
- }
-}
-
-const char* SharingModeToString(aaudio_sharing_mode_t mode) {
- switch (mode) {
- case AAUDIO_SHARING_MODE_EXCLUSIVE:
- return "EXCLUSIVE";
- case AAUDIO_SHARING_MODE_SHARED:
- return "SHARED";
- default:
- return "UNKNOWN";
- }
-}
-
-const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
- switch (mode) {
- case AAUDIO_PERFORMANCE_MODE_NONE:
- return "NONE";
- case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
- return "POWER_SAVING";
- case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
- return "LOW_LATENCY";
- default:
- return "UNKNOWN";
- }
-}
-
-const char* FormatToString(int32_t id) {
- switch (id) {
- case AAUDIO_FORMAT_INVALID:
- return "INVALID";
- case AAUDIO_FORMAT_UNSPECIFIED:
- return "UNSPECIFIED";
- case AAUDIO_FORMAT_PCM_I16:
- return "PCM_I16";
- case AAUDIO_FORMAT_PCM_FLOAT:
- return "FLOAT";
- default:
- return "UNKNOWN";
- }
-}
-
-void ErrorCallback(AAudioStream* stream,
- void* user_data,
- aaudio_result_t error) {
- RTC_DCHECK(user_data);
- AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
- RTC_LOG(LS_WARNING) << "ErrorCallback: "
- << DirectionToString(aaudio_wrapper->direction());
- RTC_DCHECK(aaudio_wrapper->observer());
- aaudio_wrapper->observer()->OnErrorCallback(error);
-}
-
-aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
- void* user_data,
- void* audio_data,
- int32_t num_frames) {
- RTC_DCHECK(user_data);
- RTC_DCHECK(audio_data);
- AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
- RTC_DCHECK(aaudio_wrapper->observer());
- return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
-}
-
-// Wraps the stream builder object to ensure that it is released properly when
-// the stream builder goes out of scope.
-class ScopedStreamBuilder {
- public:
- ScopedStreamBuilder() {
- LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
- RTC_DCHECK(builder_);
- }
- ~ScopedStreamBuilder() {
- if (builder_) {
- LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
- }
- }
-
- AAudioStreamBuilder* get() const { return builder_; }
-
- private:
- AAudioStreamBuilder* builder_ = nullptr;
-};
-
-} // namespace
-
-AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
- aaudio_direction_t direction,
- AAudioObserverInterface* observer)
- : direction_(direction), observer_(observer) {
- RTC_LOG(LS_INFO) << "ctor";
- RTC_DCHECK(observer_);
- direction_ == AAUDIO_DIRECTION_OUTPUT
- ? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
- : audio_parameters_ = audio_manager->GetRecordAudioParameters();
- aaudio_thread_checker_.Detach();
- RTC_LOG(LS_INFO) << audio_parameters_.ToString();
-}
-
-AAudioWrapper::~AAudioWrapper() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!stream_);
-}
-
-bool AAudioWrapper::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Creates a stream builder which can be used to open an audio stream.
- ScopedStreamBuilder builder;
- // Configures the stream builder using audio parameters given at construction.
- SetStreamConfiguration(builder.get());
- // Opens a stream based on options in the stream builder.
- if (!OpenStream(builder.get())) {
- return false;
- }
- // Ensures that the opened stream could activate the requested settings.
- if (!VerifyStreamConfiguration()) {
- return false;
- }
- // Optimizes the buffer scheme for lowest possible latency and creates
- // additional buffer logic to match the 10ms buffer size used in WebRTC.
- if (!OptimizeBuffers()) {
- return false;
- }
- LogStreamState();
- return true;
-}
-
-bool AAudioWrapper::Start() {
- RTC_LOG(LS_INFO) << "Start";
- RTC_DCHECK(thread_checker_.IsCurrent());
- // TODO(henrika): this state check might not be needed.
- aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
- if (current_state != AAUDIO_STREAM_STATE_OPEN) {
- RTC_LOG(LS_ERROR) << "Invalid state: "
- << AAudio_convertStreamStateToText(current_state);
- return false;
- }
- // Asynchronous request for the stream to start.
- RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
- LogStreamState();
- return true;
-}
-
-bool AAudioWrapper::Stop() {
- RTC_LOG(LS_INFO) << "Stop: " << DirectionToString(direction());
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Asynchronous request for the stream to stop.
- RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
- CloseStream();
- aaudio_thread_checker_.Detach();
- return true;
-}
-
-double AAudioWrapper::EstimateLatencyMillis() const {
- RTC_DCHECK(stream_);
- double latency_millis = 0.0;
- if (direction() == AAUDIO_DIRECTION_INPUT) {
- // For input streams. Best guess we can do is to use the current burst size
- // as delay estimate.
- latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
- rtc::kNumMillisecsPerSec;
- } else {
- int64_t existing_frame_index;
- int64_t existing_frame_presentation_time;
- // Get the time at which a particular frame was presented to audio hardware.
- aaudio_result_t result = AAudioStream_getTimestamp(
- stream_, CLOCK_MONOTONIC, &existing_frame_index,
- &existing_frame_presentation_time);
- // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
- if (result == AAUDIO_OK) {
- // Get write index for next audio frame.
- int64_t next_frame_index = frames_written();
- // Number of frames between next frame and the existing frame.
- int64_t frame_index_delta = next_frame_index - existing_frame_index;
- // Assume the next frame will be written now.
- int64_t next_frame_write_time = rtc::TimeNanos();
- // Calculate time when next frame will be presented to the hardware taking
- // sample rate into account.
- int64_t frame_time_delta =
- (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
- int64_t next_frame_presentation_time =
- existing_frame_presentation_time + frame_time_delta;
- // Derive a latency estimate given results above.
- latency_millis = static_cast<double>(next_frame_presentation_time -
- next_frame_write_time) /
- rtc::kNumNanosecsPerMillisec;
- }
- }
- return latency_millis;
-}
-
-// Returns new buffer size or a negative error value if buffer size could not
-// be increased.
-bool AAudioWrapper::IncreaseOutputBufferSize() {
- RTC_LOG(LS_INFO) << "IncreaseBufferSize";
- RTC_DCHECK(stream_);
- RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
- RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
- aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
- // Try to increase size of buffer with one burst to reduce risk of underrun.
- buffer_size += frames_per_burst();
- // Verify that the new buffer size is not larger than max capacity.
- // TODO(henrika): keep track of case when we reach the capacity limit.
- const int32_t max_buffer_size = buffer_capacity_in_frames();
- if (buffer_size > max_buffer_size) {
- RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
- << ") is higher than max: " << max_buffer_size;
- return false;
- }
- RTC_LOG(LS_INFO) << "Updating buffer size to: " << buffer_size
- << " (max=" << max_buffer_size << ")";
- buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
- if (buffer_size < 0) {
- RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
- << AAudio_convertResultToText(buffer_size);
- return false;
- }
- RTC_LOG(LS_INFO) << "Buffer size changed to: " << buffer_size;
- return true;
-}
-
-void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
- RTC_LOG(LS_INFO) << "ClearInputStream";
- RTC_DCHECK(stream_);
- RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
- RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
- aaudio_result_t cleared_frames = 0;
- do {
- cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
- } while (cleared_frames > 0);
-}
-
-AAudioObserverInterface* AAudioWrapper::observer() const {
- return observer_;
-}
-
-AudioParameters AAudioWrapper::audio_parameters() const {
- return audio_parameters_;
-}
-
-int32_t AAudioWrapper::samples_per_frame() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getSamplesPerFrame(stream_);
-}
-
-int32_t AAudioWrapper::buffer_size_in_frames() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getBufferSizeInFrames(stream_);
-}
-
-int32_t AAudioWrapper::buffer_capacity_in_frames() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getBufferCapacityInFrames(stream_);
-}
-
-int32_t AAudioWrapper::device_id() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getDeviceId(stream_);
-}
-
-int32_t AAudioWrapper::xrun_count() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getXRunCount(stream_);
-}
-
-int32_t AAudioWrapper::format() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getFormat(stream_);
-}
-
-int32_t AAudioWrapper::sample_rate() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getSampleRate(stream_);
-}
-
-int32_t AAudioWrapper::channel_count() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getChannelCount(stream_);
-}
-
-int32_t AAudioWrapper::frames_per_callback() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getFramesPerDataCallback(stream_);
-}
-
-aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getSharingMode(stream_);
-}
-
-aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getPerformanceMode(stream_);
-}
-
-aaudio_stream_state_t AAudioWrapper::stream_state() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getState(stream_);
-}
-
-int64_t AAudioWrapper::frames_written() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getFramesWritten(stream_);
-}
-
-int64_t AAudioWrapper::frames_read() const {
- RTC_DCHECK(stream_);
- return AAudioStream_getFramesRead(stream_);
-}
-
-void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
- RTC_LOG(LS_INFO) << "SetStreamConfiguration";
- RTC_DCHECK(builder);
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Request usage of default primary output/input device.
- // TODO(henrika): verify that default device follows Java APIs.
- // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
- AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
- // Use preferred sample rate given by the audio parameters.
- AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
- // Use preferred channel configuration given by the audio parameters.
- AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
- // Always use 16-bit PCM audio sample format.
- AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
- // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
- // Ask for exclusive mode since this will give us the lowest possible latency.
- // If exclusive mode isn't available, shared mode will be used instead.
- AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
- // Use the direction that was given at construction.
- AAudioStreamBuilder_setDirection(builder, direction_);
- // TODO(henrika): investigate performance using different performance modes.
- AAudioStreamBuilder_setPerformanceMode(builder,
- AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
- // Given that WebRTC applications require low latency, our audio stream uses
- // an asynchronous callback function to transfer data to and from the
- // application. AAudio executes the callback in a higher-priority thread that
- // has better performance.
- AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
- // Request that AAudio calls this functions if any error occurs on a callback
- // thread.
- AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
-}
-
-bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
- RTC_LOG(LS_INFO) << "OpenStream";
- RTC_DCHECK(builder);
- AAudioStream* stream = nullptr;
- RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
- stream_ = stream;
- LogStreamConfiguration();
- return true;
-}
-
-void AAudioWrapper::CloseStream() {
- RTC_LOG(LS_INFO) << "CloseStream";
- RTC_DCHECK(stream_);
- LOG_ON_ERROR(AAudioStream_close(stream_));
- stream_ = nullptr;
-}
-
-void AAudioWrapper::LogStreamConfiguration() {
- RTC_DCHECK(stream_);
- char ss_buf[1024];
- rtc::SimpleStringBuilder ss(ss_buf);
- ss << "Stream Configuration: ";
- ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
- ss << ", samples per frame=" << samples_per_frame();
- ss << ", format=" << FormatToString(format());
- ss << ", sharing mode=" << SharingModeToString(sharing_mode());
- ss << ", performance mode=" << PerformanceModeToString(performance_mode());
- ss << ", direction=" << DirectionToString(direction());
- ss << ", device id=" << AAudioStream_getDeviceId(stream_);
- ss << ", frames per callback=" << frames_per_callback();
- RTC_LOG(LS_INFO) << ss.str();
-}
-
-void AAudioWrapper::LogStreamState() {
- RTC_LOG(LS_INFO) << "AAudio stream state: "
- << AAudio_convertStreamStateToText(stream_state());
-}
-
-bool AAudioWrapper::VerifyStreamConfiguration() {
- RTC_LOG(LS_INFO) << "VerifyStreamConfiguration";
- RTC_DCHECK(stream_);
- // TODO(henrika): should we verify device ID as well?
- if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
- RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
- return false;
- }
- if (AAudioStream_getChannelCount(stream_) !=
- static_cast<int32_t>(audio_parameters().channels())) {
- RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
- return false;
- }
- if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
- RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
- return false;
- }
- if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
- RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
- return false;
- }
- if (AAudioStream_getPerformanceMode(stream_) !=
- AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
- RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
- return false;
- }
- if (AAudioStream_getDirection(stream_) != direction()) {
- RTC_LOG(LS_ERROR) << "Stream direction could not be set";
- return false;
- }
- if (AAudioStream_getSamplesPerFrame(stream_) !=
- static_cast<int32_t>(audio_parameters().channels())) {
- RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
- return false;
- }
- return true;
-}
-
-bool AAudioWrapper::OptimizeBuffers() {
- RTC_LOG(LS_INFO) << "OptimizeBuffers";
- RTC_DCHECK(stream_);
- // Maximum number of frames that can be filled without blocking.
- RTC_LOG(LS_INFO) << "max buffer capacity in frames: "
- << buffer_capacity_in_frames();
- // Query the number of frames that the application should read or write at
- // one time for optimal performance.
- int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
- RTC_LOG(LS_INFO) << "frames per burst for optimal performance: "
- << frames_per_burst;
- frames_per_burst_ = frames_per_burst;
- if (direction() == AAUDIO_DIRECTION_INPUT) {
- // There is no point in calling setBufferSizeInFrames() for input streams
- // since it has no effect on the performance (latency in this case).
- return true;
- }
- // Set buffer size to same as burst size to guarantee lowest possible latency.
- // This size might change for output streams if underruns are detected and
- // automatic buffer adjustment is enabled.
- AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
- int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
- if (buffer_size != frames_per_burst) {
- RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
- return false;
- }
- // Maximum number of frames that can be filled without blocking.
- RTC_LOG(LS_INFO) << "buffer burst size in frames: " << buffer_size;
- return true;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/aaudio_wrapper.h b/modules/audio_device/android/aaudio_wrapper.h
deleted file mode 100644
index 1f925b9..0000000
--- a/modules/audio_device/android/aaudio_wrapper.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
-
-#include <aaudio/AAudio.h>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-
-namespace webrtc {
-
-class AudioManager;
-
-// AAudio callback interface for audio transport to/from the AAudio stream.
-// The interface also contains an error callback method for notifications of
-// e.g. device changes.
-class AAudioObserverInterface {
- public:
- // Audio data will be passed in our out of this function dependning on the
- // direction of the audio stream. This callback function will be called on a
- // real-time thread owned by AAudio.
- virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data,
- int32_t num_frames) = 0;
- // AAudio will call this functions if any error occurs on a callback thread.
- // In response, this function could signal or launch another thread to reopen
- // a stream on another device. Do not reopen the stream in this callback.
- virtual void OnErrorCallback(aaudio_result_t error) = 0;
-
- protected:
- virtual ~AAudioObserverInterface() {}
-};
-
-// Utility class which wraps the C-based AAudio API into a more handy C++ class
-// where the underlying resources (AAudioStreamBuilder and AAudioStream) are
-// encapsulated. User must set the direction (in or out) at construction since
-// it defines the stream type and the direction of the data flow in the
-// AAudioObserverInterface.
-//
-// AAudio is a new Android C API introduced in the Android O (26) release.
-// It is designed for high-performance audio applications that require low
-// latency. Applications communicate with AAudio by reading and writing data
-// to streams.
-//
-// Each stream is attached to a single audio device, where each audio device
-// has a unique ID. The ID can be used to bind an audio stream to a specific
-// audio device but this implementation lets AAudio choose the default primary
-// device instead (device selection takes place in Java). A stream can only
-// move data in one direction. When a stream is opened, Android checks to
-// ensure that the audio device and stream direction agree.
-class AAudioWrapper {
- public:
- AAudioWrapper(AudioManager* audio_manager,
- aaudio_direction_t direction,
- AAudioObserverInterface* observer);
- ~AAudioWrapper();
-
- bool Init();
- bool Start();
- bool Stop();
-
- // For output streams: estimates latency between writing an audio frame to
- // the output stream and the time that same frame is played out on the output
- // audio device.
- // For input streams: estimates latency between reading an audio frame from
- // the input stream and the time that same frame was recorded on the input
- // audio device.
- double EstimateLatencyMillis() const;
-
- // Increases the internal buffer size for output streams by one burst size to
- // reduce the risk of underruns. Can be used while a stream is active.
- bool IncreaseOutputBufferSize();
-
- // Drains the recording stream of any existing data by reading from it until
- // it's empty. Can be used to clear out old data before starting a new audio
- // session.
- void ClearInputStream(void* audio_data, int32_t num_frames);
-
- AAudioObserverInterface* observer() const;
- AudioParameters audio_parameters() const;
- int32_t samples_per_frame() const;
- int32_t buffer_size_in_frames() const;
- int32_t buffer_capacity_in_frames() const;
- int32_t device_id() const;
- int32_t xrun_count() const;
- int32_t format() const;
- int32_t sample_rate() const;
- int32_t channel_count() const;
- int32_t frames_per_callback() const;
- aaudio_sharing_mode_t sharing_mode() const;
- aaudio_performance_mode_t performance_mode() const;
- aaudio_stream_state_t stream_state() const;
- int64_t frames_written() const;
- int64_t frames_read() const;
- aaudio_direction_t direction() const { return direction_; }
- AAudioStream* stream() const { return stream_; }
- int32_t frames_per_burst() const { return frames_per_burst_; }
-
- private:
- void SetStreamConfiguration(AAudioStreamBuilder* builder);
- bool OpenStream(AAudioStreamBuilder* builder);
- void CloseStream();
- void LogStreamConfiguration();
- void LogStreamState();
- bool VerifyStreamConfiguration();
- bool OptimizeBuffers();
-
- SequenceChecker thread_checker_;
- SequenceChecker aaudio_thread_checker_;
- AudioParameters audio_parameters_;
- const aaudio_direction_t direction_;
- AAudioObserverInterface* observer_ = nullptr;
- AAudioStream* stream_ = nullptr;
- int32_t frames_per_burst_ = 0;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
diff --git a/modules/audio_device/android/audio_common.h b/modules/audio_device/android/audio_common.h
deleted file mode 100644
index 81ea733..0000000
--- a/modules/audio_device/android/audio_common.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
-
-namespace webrtc {
-
-const int kDefaultSampleRate = 44100;
-// Delay estimates for the two different supported modes. These values are based
-// on real-time round-trip delay estimates on a large set of devices and they
-// are lower bounds since the filter length is 128 ms, so the AEC works for
-// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
-// cases, the lowest delay estimate will not be utilized since devices that
-// support low-latency output audio often supports HW AEC as well.
-const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
-const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
diff --git a/modules/audio_device/android/audio_device_template.h b/modules/audio_device/android/audio_device_template.h
deleted file mode 100644
index 999c587..0000000
--- a/modules/audio_device/android/audio_device_template.h
+++ /dev/null
@@ -1,435 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-
-namespace webrtc {
-
-// InputType/OutputType can be any class that implements the capturing/rendering
-// part of the AudioDeviceGeneric API.
-// Construction and destruction must be done on one and the same thread. Each
-// internal implementation of InputType and OutputType will RTC_DCHECK if that
-// is not the case. All implemented methods must also be called on the same
-// thread. See comments in each InputType/OutputType class for more info.
-// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
-// and ClearAndroidAudioDeviceObjects) from a different thread but both will
-// RTC_CHECK that the calling thread is attached to a Java VM.
-
-template <class InputType, class OutputType>
-class AudioDeviceTemplate : public AudioDeviceGeneric {
- public:
- AudioDeviceTemplate(AudioDeviceModule::AudioLayer audio_layer,
- AudioManager* audio_manager)
- : audio_layer_(audio_layer),
- audio_manager_(audio_manager),
- output_(audio_manager_),
- input_(audio_manager_),
- initialized_(false) {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- RTC_CHECK(audio_manager);
- audio_manager_->SetActiveAudioLayer(audio_layer);
- }
-
- virtual ~AudioDeviceTemplate() { RTC_LOG(LS_INFO) << __FUNCTION__; }
-
- int32_t ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- audioLayer = audio_layer_;
- return 0;
- }
-
- InitStatus Init() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- if (!audio_manager_->Init()) {
- return InitStatus::OTHER_ERROR;
- }
- if (output_.Init() != 0) {
- audio_manager_->Close();
- return InitStatus::PLAYOUT_ERROR;
- }
- if (input_.Init() != 0) {
- output_.Terminate();
- audio_manager_->Close();
- return InitStatus::RECORDING_ERROR;
- }
- initialized_ = true;
- return InitStatus::OK;
- }
-
- int32_t Terminate() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- RTC_DCHECK(thread_checker_.IsCurrent());
- int32_t err = input_.Terminate();
- err |= output_.Terminate();
- err |= !audio_manager_->Close();
- initialized_ = false;
- RTC_DCHECK_EQ(err, 0);
- return err;
- }
-
- bool Initialized() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- RTC_DCHECK(thread_checker_.IsCurrent());
- return initialized_;
- }
-
- int16_t PlayoutDevices() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 1;
- }
-
- int16_t RecordingDevices() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 1;
- }
-
- int32_t PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t SetPlayoutDevice(uint16_t index) override {
- // OK to use but it has no effect currently since device selection is
- // done using Andoid APIs instead.
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 0;
- }
-
- int32_t SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t SetRecordingDevice(uint16_t index) override {
- // OK to use but it has no effect currently since device selection is
- // done using Andoid APIs instead.
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 0;
- }
-
- int32_t SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t PlayoutIsAvailable(bool& available) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- available = true;
- return 0;
- }
-
- int32_t InitPlayout() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.InitPlayout();
- }
-
- bool PlayoutIsInitialized() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.PlayoutIsInitialized();
- }
-
- int32_t RecordingIsAvailable(bool& available) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- available = true;
- return 0;
- }
-
- int32_t InitRecording() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return input_.InitRecording();
- }
-
- bool RecordingIsInitialized() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return input_.RecordingIsInitialized();
- }
-
- int32_t StartPlayout() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- if (!audio_manager_->IsCommunicationModeEnabled()) {
- RTC_LOG(LS_WARNING)
- << "The application should use MODE_IN_COMMUNICATION audio mode!";
- }
- return output_.StartPlayout();
- }
-
- int32_t StopPlayout() override {
- // Avoid using audio manger (JNI/Java cost) if playout was inactive.
- if (!Playing())
- return 0;
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- int32_t err = output_.StopPlayout();
- return err;
- }
-
- bool Playing() const override {
- RTC_LOG(LS_INFO) << __FUNCTION__;
- return output_.Playing();
- }
-
- int32_t StartRecording() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- if (!audio_manager_->IsCommunicationModeEnabled()) {
- RTC_LOG(LS_WARNING)
- << "The application should use MODE_IN_COMMUNICATION audio mode!";
- }
- return input_.StartRecording();
- }
-
- int32_t StopRecording() override {
- // Avoid using audio manger (JNI/Java cost) if recording was inactive.
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- if (!Recording())
- return 0;
- int32_t err = input_.StopRecording();
- return err;
- }
-
- bool Recording() const override { return input_.Recording(); }
-
- int32_t InitSpeaker() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 0;
- }
-
- bool SpeakerIsInitialized() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return true;
- }
-
- int32_t InitMicrophone() override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return 0;
- }
-
- bool MicrophoneIsInitialized() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return true;
- }
-
- int32_t SpeakerVolumeIsAvailable(bool& available) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.SpeakerVolumeIsAvailable(available);
- }
-
- int32_t SetSpeakerVolume(uint32_t volume) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.SetSpeakerVolume(volume);
- }
-
- int32_t SpeakerVolume(uint32_t& volume) const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.SpeakerVolume(volume);
- }
-
- int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.MaxSpeakerVolume(maxVolume);
- }
-
- int32_t MinSpeakerVolume(uint32_t& minVolume) const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return output_.MinSpeakerVolume(minVolume);
- }
-
- int32_t MicrophoneVolumeIsAvailable(bool& available) override {
- available = false;
- return -1;
- }
-
- int32_t SetMicrophoneVolume(uint32_t volume) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t MicrophoneVolume(uint32_t& volume) const override {
- RTC_CHECK_NOTREACHED();
- return -1;
- }
-
- int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t SpeakerMuteIsAvailable(bool& available) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t SetSpeakerMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
-
- int32_t SpeakerMute(bool& enabled) const override { RTC_CHECK_NOTREACHED(); }
-
- int32_t MicrophoneMuteIsAvailable(bool& available) override {
- RTC_CHECK_NOTREACHED();
- }
-
- int32_t SetMicrophoneMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
-
- int32_t MicrophoneMute(bool& enabled) const override {
- RTC_CHECK_NOTREACHED();
- }
-
- // Returns true if the audio manager has been configured to support stereo
- // and false otherwised. Default is mono.
- int32_t StereoPlayoutIsAvailable(bool& available) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- available = audio_manager_->IsStereoPlayoutSupported();
- return 0;
- }
-
- int32_t SetStereoPlayout(bool enable) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- bool available = audio_manager_->IsStereoPlayoutSupported();
- // Android does not support changes between mono and stero on the fly.
- // Instead, the native audio layer is configured via the audio manager
- // to either support mono or stereo. It is allowed to call this method
- // if that same state is not modified.
- return (enable == available) ? 0 : -1;
- }
-
- int32_t StereoPlayout(bool& enabled) const override {
- enabled = audio_manager_->IsStereoPlayoutSupported();
- return 0;
- }
-
- int32_t StereoRecordingIsAvailable(bool& available) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- available = audio_manager_->IsStereoRecordSupported();
- return 0;
- }
-
- int32_t SetStereoRecording(bool enable) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- bool available = audio_manager_->IsStereoRecordSupported();
- // Android does not support changes between mono and stero on the fly.
- // Instead, the native audio layer is configured via the audio manager
- // to either support mono or stereo. It is allowed to call this method
- // if that same state is not modified.
- return (enable == available) ? 0 : -1;
- }
-
- int32_t StereoRecording(bool& enabled) const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- enabled = audio_manager_->IsStereoRecordSupported();
- return 0;
- }
-
- int32_t PlayoutDelay(uint16_t& delay_ms) const override {
- // Best guess we can do is to use half of the estimated total delay.
- delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
- RTC_DCHECK_GT(delay_ms, 0);
- return 0;
- }
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- output_.AttachAudioBuffer(audioBuffer);
- input_.AttachAudioBuffer(audioBuffer);
- }
-
- // Returns true if the device both supports built in AEC and the device
- // is not blacklisted.
- // Currently, if OpenSL ES is used in both directions, this method will still
- // report the correct value and it has the correct effect. As an example:
- // a device supports built in AEC and this method returns true. Libjingle
- // will then disable the WebRTC based AEC and that will work for all devices
- // (mainly Nexus) even when OpenSL ES is used for input since our current
- // implementation will enable built-in AEC by default also for OpenSL ES.
- // The only "bad" thing that happens today is that when Libjingle calls
- // OpenSLESRecorder::EnableBuiltInAEC() it will not have any real effect and
- // a "Not Implemented" log will be filed. This non-perfect state will remain
- // until I have added full support for audio effects based on OpenSL ES APIs.
- bool BuiltInAECIsAvailable() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return audio_manager_->IsAcousticEchoCancelerSupported();
- }
-
- // TODO(henrika): add implementation for OpenSL ES based audio as well.
- int32_t EnableBuiltInAEC(bool enable) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
- RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
- return input_.EnableBuiltInAEC(enable);
- }
-
- // Returns true if the device both supports built in AGC and the device
- // is not blacklisted.
- // TODO(henrika): add implementation for OpenSL ES based audio as well.
- // In addition, see comments for BuiltInAECIsAvailable().
- bool BuiltInAGCIsAvailable() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return audio_manager_->IsAutomaticGainControlSupported();
- }
-
- // TODO(henrika): add implementation for OpenSL ES based audio as well.
- int32_t EnableBuiltInAGC(bool enable) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
- RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
- return input_.EnableBuiltInAGC(enable);
- }
-
- // Returns true if the device both supports built in NS and the device
- // is not blacklisted.
- // TODO(henrika): add implementation for OpenSL ES based audio as well.
- // In addition, see comments for BuiltInAECIsAvailable().
- bool BuiltInNSIsAvailable() const override {
- RTC_DLOG(LS_INFO) << __FUNCTION__;
- return audio_manager_->IsNoiseSuppressorSupported();
- }
-
- // TODO(henrika): add implementation for OpenSL ES based audio as well.
- int32_t EnableBuiltInNS(bool enable) override {
- RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
- RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
- return input_.EnableBuiltInNS(enable);
- }
-
- private:
- SequenceChecker thread_checker_;
-
- // Local copy of the audio layer set during construction of the
- // AudioDeviceModuleImpl instance. Read only value.
- const AudioDeviceModule::AudioLayer audio_layer_;
-
- // Non-owning raw pointer to AudioManager instance given to use at
- // construction. The real object is owned by AudioDeviceModuleImpl and the
- // life time is the same as that of the AudioDeviceModuleImpl, hence there
- // is no risk of reading a NULL pointer at any time in this class.
- AudioManager* const audio_manager_;
-
- OutputType output_;
-
- InputType input_;
-
- bool initialized_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
diff --git a/modules/audio_device/android/audio_device_unittest.cc b/modules/audio_device/android/audio_device_unittest.cc
deleted file mode 100644
index 4e607bc..0000000
--- a/modules/audio_device/android/audio_device_unittest.cc
+++ /dev/null
@@ -1,1019 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/include/audio_device.h"
-
-#include <algorithm>
-#include <limits>
-#include <list>
-#include <memory>
-#include <numeric>
-#include <string>
-#include <vector>
-
-#include "api/scoped_refptr.h"
-#include "api/task_queue/default_task_queue_factory.h"
-#include "api/task_queue/task_queue_factory.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/android/build_info.h"
-#include "modules/audio_device/android/ensure_initialized.h"
-#include "modules/audio_device/audio_device_impl.h"
-#include "modules/audio_device/include/mock_audio_transport.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/event.h"
-#include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/time_utils.h"
-#include "test/gmock.h"
-#include "test/gtest.h"
-#include "test/testsupport/file_utils.h"
-
-using std::cout;
-using std::endl;
-using ::testing::_;
-using ::testing::AtLeast;
-using ::testing::Gt;
-using ::testing::Invoke;
-using ::testing::NiceMock;
-using ::testing::NotNull;
-using ::testing::Return;
-
-// #define ENABLE_DEBUG_PRINTF
-#ifdef ENABLE_DEBUG_PRINTF
-#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
-#else
-#define PRINTD(...) ((void)0)
-#endif
-#define PRINT(...) fprintf(stderr, __VA_ARGS__);
-
-namespace webrtc {
-
-// Number of callbacks (input or output) the tests waits for before we set
-// an event indicating that the test was OK.
-static const size_t kNumCallbacks = 10;
-// Max amount of time we wait for an event to be set while counting callbacks.
-static const int kTestTimeOutInMilliseconds = 10 * 1000;
-// Average number of audio callbacks per second assuming 10ms packet size.
-static const size_t kNumCallbacksPerSecond = 100;
-// Play out a test file during this time (unit is in seconds).
-static const int kFilePlayTimeInSec = 5;
-static const size_t kBitsPerSample = 16;
-static const size_t kBytesPerSample = kBitsPerSample / 8;
-// Run the full-duplex test during this time (unit is in seconds).
-// Note that first `kNumIgnoreFirstCallbacks` are ignored.
-static const int kFullDuplexTimeInSec = 5;
-// Wait for the callback sequence to stabilize by ignoring this amount of the
-// initial callbacks (avoids initial FIFO access).
-// Only used in the RunPlayoutAndRecordingInFullDuplex test.
-static const size_t kNumIgnoreFirstCallbacks = 50;
-// Sets the number of impulses per second in the latency test.
-static const int kImpulseFrequencyInHz = 1;
-// Length of round-trip latency measurements. Number of transmitted impulses
-// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
-static const int kMeasureLatencyTimeInSec = 11;
-// Utilized in round-trip latency measurements to avoid capturing noise samples.
-static const int kImpulseThreshold = 1000;
-static const char kTag[] = "[..........] ";
-
-enum TransportType {
- kPlayout = 0x1,
- kRecording = 0x2,
-};
-
-// Interface for processing the audio stream. Real implementations can e.g.
-// run audio in loopback, read audio from a file or perform latency
-// measurements.
-class AudioStreamInterface {
- public:
- virtual void Write(const void* source, size_t num_frames) = 0;
- virtual void Read(void* destination, size_t num_frames) = 0;
-
- protected:
- virtual ~AudioStreamInterface() {}
-};
-
-// Reads audio samples from a PCM file where the file is stored in memory at
-// construction.
-class FileAudioStream : public AudioStreamInterface {
- public:
- FileAudioStream(size_t num_callbacks,
- const std::string& file_name,
- int sample_rate)
- : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) {
- file_size_in_bytes_ = test::GetFileSize(file_name);
- sample_rate_ = sample_rate;
- EXPECT_GE(file_size_in_callbacks(), num_callbacks)
- << "Size of test file is not large enough to last during the test.";
- const size_t num_16bit_samples =
- test::GetFileSize(file_name) / kBytesPerSample;
- file_.reset(new int16_t[num_16bit_samples]);
- FILE* audio_file = fopen(file_name.c_str(), "rb");
- EXPECT_NE(audio_file, nullptr);
- size_t num_samples_read =
- fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
- EXPECT_EQ(num_samples_read, num_16bit_samples);
- fclose(audio_file);
- }
-
- // AudioStreamInterface::Write() is not implemented.
- void Write(const void* source, size_t num_frames) override {}
-
- // Read samples from file stored in memory (at construction) and copy
- // `num_frames` (<=> 10ms) to the `destination` byte buffer.
- void Read(void* destination, size_t num_frames) override {
- memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
- num_frames * sizeof(int16_t));
- file_pos_ += num_frames;
- }
-
- int file_size_in_seconds() const {
- return static_cast<int>(file_size_in_bytes_ /
- (kBytesPerSample * sample_rate_));
- }
- size_t file_size_in_callbacks() const {
- return file_size_in_seconds() * kNumCallbacksPerSecond;
- }
-
- private:
- size_t file_size_in_bytes_;
- int sample_rate_;
- std::unique_ptr<int16_t[]> file_;
- size_t file_pos_;
-};
-
-// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
-// buffers of fixed size and allows Write and Read operations. The idea is to
-// store recorded audio buffers (using Write) and then read (using Read) these
-// stored buffers with as short delay as possible when the audio layer needs
-// data to play out. The number of buffers in the FIFO will stabilize under
-// normal conditions since there will be a balance between Write and Read calls.
-// The container is a std::list container and access is protected with a lock
-// since both sides (playout and recording) are driven by its own thread.
-class FifoAudioStream : public AudioStreamInterface {
- public:
- explicit FifoAudioStream(size_t frames_per_buffer)
- : frames_per_buffer_(frames_per_buffer),
- bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
- fifo_(new AudioBufferList),
- largest_size_(0),
- total_written_elements_(0),
- write_count_(0) {
- EXPECT_NE(fifo_.get(), nullptr);
- }
-
- ~FifoAudioStream() { Flush(); }
-
- // Allocate new memory, copy `num_frames` samples from `source` into memory
- // and add pointer to the memory location to end of the list.
- // Increases the size of the FIFO by one element.
- void Write(const void* source, size_t num_frames) override {
- ASSERT_EQ(num_frames, frames_per_buffer_);
- PRINTD("+");
- if (write_count_++ < kNumIgnoreFirstCallbacks) {
- return;
- }
- int16_t* memory = new int16_t[frames_per_buffer_];
- memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
- MutexLock lock(&lock_);
- fifo_->push_back(memory);
- const size_t size = fifo_->size();
- if (size > largest_size_) {
- largest_size_ = size;
- PRINTD("(%zu)", largest_size_);
- }
- total_written_elements_ += size;
- }
-
- // Read pointer to data buffer from front of list, copy `num_frames` of stored
- // data into `destination` and delete the utilized memory allocation.
- // Decreases the size of the FIFO by one element.
- void Read(void* destination, size_t num_frames) override {
- ASSERT_EQ(num_frames, frames_per_buffer_);
- PRINTD("-");
- MutexLock lock(&lock_);
- if (fifo_->empty()) {
- memset(destination, 0, bytes_per_buffer_);
- } else {
- int16_t* memory = fifo_->front();
- fifo_->pop_front();
- memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
- delete memory;
- }
- }
-
- size_t size() const { return fifo_->size(); }
-
- size_t largest_size() const { return largest_size_; }
-
- size_t average_size() const {
- return (total_written_elements_ == 0)
- ? 0.0
- : 0.5 + static_cast<float>(total_written_elements_) /
- (write_count_ - kNumIgnoreFirstCallbacks);
- }
-
- private:
- void Flush() {
- for (auto it = fifo_->begin(); it != fifo_->end(); ++it) {
- delete *it;
- }
- fifo_->clear();
- }
-
- using AudioBufferList = std::list<int16_t*>;
- Mutex lock_;
- const size_t frames_per_buffer_;
- const size_t bytes_per_buffer_;
- std::unique_ptr<AudioBufferList> fifo_;
- size_t largest_size_;
- size_t total_written_elements_;
- size_t write_count_;
-};
-
-// Inserts periodic impulses and measures the latency between the time of
-// transmission and time of receiving the same impulse.
-// Usage requires a special hardware called Audio Loopback Dongle.
-// See http://source.android.com/devices/audio/loopback.html for details.
-class LatencyMeasuringAudioStream : public AudioStreamInterface {
- public:
- explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
- : frames_per_buffer_(frames_per_buffer),
- bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
- play_count_(0),
- rec_count_(0),
- pulse_time_(0) {}
-
- // Insert periodic impulses in first two samples of `destination`.
- void Read(void* destination, size_t num_frames) override {
- ASSERT_EQ(num_frames, frames_per_buffer_);
- if (play_count_ == 0) {
- PRINT("[");
- }
- play_count_++;
- memset(destination, 0, bytes_per_buffer_);
- if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
- if (pulse_time_ == 0) {
- pulse_time_ = rtc::TimeMillis();
- }
- PRINT(".");
- const int16_t impulse = std::numeric_limits<int16_t>::max();
- int16_t* ptr16 = static_cast<int16_t*>(destination);
- for (size_t i = 0; i < 2; ++i) {
- ptr16[i] = impulse;
- }
- }
- }
-
- // Detect received impulses in `source`, derive time between transmission and
- // detection and add the calculated delay to list of latencies.
- void Write(const void* source, size_t num_frames) override {
- ASSERT_EQ(num_frames, frames_per_buffer_);
- rec_count_++;
- if (pulse_time_ == 0) {
- // Avoid detection of new impulse response until a new impulse has
- // been transmitted (sets `pulse_time_` to value larger than zero).
- return;
- }
- const int16_t* ptr16 = static_cast<const int16_t*>(source);
- std::vector<int16_t> vec(ptr16, ptr16 + num_frames);
- // Find max value in the audio buffer.
- int max = *std::max_element(vec.begin(), vec.end());
- // Find index (element position in vector) of the max element.
- int index_of_max =
- std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
- if (max > kImpulseThreshold) {
- PRINTD("(%d,%d)", max, index_of_max);
- int64_t now_time = rtc::TimeMillis();
- int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
- PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
- PRINTD("[%d]", extra_delay);
- // Total latency is the difference between transmit time and detection
- // tome plus the extra delay within the buffer in which we detected the
- // received impulse. It is transmitted at sample 0 but can be received
- // at sample N where N > 0. The term `extra_delay` accounts for N and it
- // is a value between 0 and 10ms.
- latencies_.push_back(now_time - pulse_time_ + extra_delay);
- pulse_time_ = 0;
- } else {
- PRINTD("-");
- }
- }
-
- size_t num_latency_values() const { return latencies_.size(); }
-
- int min_latency() const {
- if (latencies_.empty())
- return 0;
- return *std::min_element(latencies_.begin(), latencies_.end());
- }
-
- int max_latency() const {
- if (latencies_.empty())
- return 0;
- return *std::max_element(latencies_.begin(), latencies_.end());
- }
-
- int average_latency() const {
- if (latencies_.empty())
- return 0;
- return 0.5 + static_cast<double>(
- std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
- latencies_.size();
- }
-
- void PrintResults() const {
- PRINT("] ");
- for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
- PRINT("%d ", *it);
- }
- PRINT("\n");
- PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(),
- max_latency(), average_latency());
- }
-
- int IndexToMilliseconds(double index) const {
- return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5);
- }
-
- private:
- const size_t frames_per_buffer_;
- const size_t bytes_per_buffer_;
- size_t play_count_;
- size_t rec_count_;
- int64_t pulse_time_;
- std::vector<int> latencies_;
-};
-
-// Mocks the AudioTransport object and proxies actions for the two callbacks
-// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
-// of AudioStreamInterface.
-class MockAudioTransportAndroid : public test::MockAudioTransport {
- public:
- explicit MockAudioTransportAndroid(int type)
- : num_callbacks_(0),
- type_(type),
- play_count_(0),
- rec_count_(0),
- audio_stream_(nullptr) {}
-
- virtual ~MockAudioTransportAndroid() {}
-
- // Set default actions of the mock object. We are delegating to fake
- // implementations (of AudioStreamInterface) here.
- void HandleCallbacks(rtc::Event* test_is_done,
- AudioStreamInterface* audio_stream,
- int num_callbacks) {
- test_is_done_ = test_is_done;
- audio_stream_ = audio_stream;
- num_callbacks_ = num_callbacks;
- if (play_mode()) {
- ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
- .WillByDefault(
- Invoke(this, &MockAudioTransportAndroid::RealNeedMorePlayData));
- }
- if (rec_mode()) {
- ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
- .WillByDefault(Invoke(
- this, &MockAudioTransportAndroid::RealRecordedDataIsAvailable));
- }
- }
-
- int32_t RealRecordedDataIsAvailable(const void* audioSamples,
- const size_t nSamples,
- const size_t nBytesPerSample,
- const size_t nChannels,
- const uint32_t samplesPerSec,
- const uint32_t totalDelayMS,
- const int32_t clockDrift,
- const uint32_t currentMicLevel,
- const bool keyPressed,
- uint32_t& newMicLevel) { // NOLINT
- EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
- rec_count_++;
- // Process the recorded audio stream if an AudioStreamInterface
- // implementation exists.
- if (audio_stream_) {
- audio_stream_->Write(audioSamples, nSamples);
- }
- if (ReceivedEnoughCallbacks()) {
- test_is_done_->Set();
- }
- return 0;
- }
-
- int32_t RealNeedMorePlayData(const size_t nSamples,
- const size_t nBytesPerSample,
- const size_t nChannels,
- const uint32_t samplesPerSec,
- void* audioSamples,
- size_t& nSamplesOut, // NOLINT
- int64_t* elapsed_time_ms,
- int64_t* ntp_time_ms) {
- EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
- play_count_++;
- nSamplesOut = nSamples;
- // Read (possibly processed) audio stream samples to be played out if an
- // AudioStreamInterface implementation exists.
- if (audio_stream_) {
- audio_stream_->Read(audioSamples, nSamples);
- }
- if (ReceivedEnoughCallbacks()) {
- test_is_done_->Set();
- }
- return 0;
- }
-
- bool ReceivedEnoughCallbacks() {
- bool recording_done = false;
- if (rec_mode())
- recording_done = rec_count_ >= num_callbacks_;
- else
- recording_done = true;
-
- bool playout_done = false;
- if (play_mode())
- playout_done = play_count_ >= num_callbacks_;
- else
- playout_done = true;
-
- return recording_done && playout_done;
- }
-
- bool play_mode() const { return type_ & kPlayout; }
- bool rec_mode() const { return type_ & kRecording; }
-
- private:
- rtc::Event* test_is_done_;
- size_t num_callbacks_;
- int type_;
- size_t play_count_;
- size_t rec_count_;
- AudioStreamInterface* audio_stream_;
- std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
-};
-
-// AudioDeviceTest test fixture.
-class AudioDeviceTest : public ::testing::Test {
- protected:
- AudioDeviceTest() : task_queue_factory_(CreateDefaultTaskQueueFactory()) {
- // One-time initialization of JVM and application context. Ensures that we
- // can do calls between C++ and Java. Initializes both Java and OpenSL ES
- // implementations.
- webrtc::audiodevicemodule::EnsureInitialized();
- // Creates an audio device using a default audio layer.
- audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
- EXPECT_NE(audio_device_.get(), nullptr);
- EXPECT_EQ(0, audio_device_->Init());
- playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
- record_parameters_ = audio_manager()->GetRecordAudioParameters();
- build_info_.reset(new BuildInfo());
- }
- virtual ~AudioDeviceTest() { EXPECT_EQ(0, audio_device_->Terminate()); }
-
- int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
- int record_sample_rate() const { return record_parameters_.sample_rate(); }
- size_t playout_channels() const { return playout_parameters_.channels(); }
- size_t record_channels() const { return record_parameters_.channels(); }
- size_t playout_frames_per_10ms_buffer() const {
- return playout_parameters_.frames_per_10ms_buffer();
- }
- size_t record_frames_per_10ms_buffer() const {
- return record_parameters_.frames_per_10ms_buffer();
- }
-
- int total_delay_ms() const {
- return audio_manager()->GetDelayEstimateInMilliseconds();
- }
-
- rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
- return audio_device_;
- }
-
- AudioDeviceModuleImpl* audio_device_impl() const {
- return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
- }
-
- AudioManager* audio_manager() const {
- return audio_device_impl()->GetAndroidAudioManagerForTest();
- }
-
- AudioManager* GetAudioManager(AudioDeviceModule* adm) const {
- return static_cast<AudioDeviceModuleImpl*>(adm)
- ->GetAndroidAudioManagerForTest();
- }
-
- AudioDeviceBuffer* audio_device_buffer() const {
- return audio_device_impl()->GetAudioDeviceBuffer();
- }
-
- rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
- AudioDeviceModule::AudioLayer audio_layer) {
- rtc::scoped_refptr<AudioDeviceModule> module(
- AudioDeviceModule::Create(audio_layer, task_queue_factory_.get()));
- return module;
- }
-
- // Returns file name relative to the resource root given a sample rate.
- std::string GetFileName(int sample_rate) {
- EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100);
- char fname[64];
- snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
- sample_rate / 1000);
- std::string file_name(webrtc::test::ResourcePath(fname, "pcm"));
- EXPECT_TRUE(test::FileExists(file_name));
-#ifdef ENABLE_PRINTF
- PRINT("file name: %s\n", file_name.c_str());
- const size_t bytes = test::GetFileSize(file_name);
- PRINT("file size: %zu [bytes]\n", bytes);
- PRINT("file size: %zu [samples]\n", bytes / kBytesPerSample);
- const int seconds =
- static_cast<int>(bytes / (sample_rate * kBytesPerSample));
- PRINT("file size: %d [secs]\n", seconds);
- PRINT("file size: %zu [callbacks]\n", seconds * kNumCallbacksPerSecond);
-#endif
- return file_name;
- }
-
- AudioDeviceModule::AudioLayer GetActiveAudioLayer() const {
- AudioDeviceModule::AudioLayer audio_layer;
- EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
- return audio_layer;
- }
-
- int TestDelayOnAudioLayer(
- const AudioDeviceModule::AudioLayer& layer_to_test) {
- rtc::scoped_refptr<AudioDeviceModule> audio_device;
- audio_device = CreateAudioDevice(layer_to_test);
- EXPECT_NE(audio_device.get(), nullptr);
- AudioManager* audio_manager = GetAudioManager(audio_device.get());
- EXPECT_NE(audio_manager, nullptr);
- return audio_manager->GetDelayEstimateInMilliseconds();
- }
-
- AudioDeviceModule::AudioLayer TestActiveAudioLayer(
- const AudioDeviceModule::AudioLayer& layer_to_test) {
- rtc::scoped_refptr<AudioDeviceModule> audio_device;
- audio_device = CreateAudioDevice(layer_to_test);
- EXPECT_NE(audio_device.get(), nullptr);
- AudioDeviceModule::AudioLayer active;
- EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active));
- return active;
- }
-
- bool DisableTestForThisDevice(const std::string& model) {
- return (build_info_->GetDeviceModel() == model);
- }
-
- // Volume control is currently only supported for the Java output audio layer.
- // For OpenSL ES, the internal stream volume is always on max level and there
- // is no need for this test to set it to max.
- bool AudioLayerSupportsVolumeControl() const {
- return GetActiveAudioLayer() == AudioDeviceModule::kAndroidJavaAudio;
- }
-
- void SetMaxPlayoutVolume() {
- if (!AudioLayerSupportsVolumeControl())
- return;
- uint32_t max_volume;
- EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
- EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
- }
-
- void DisableBuiltInAECIfAvailable() {
- if (audio_device()->BuiltInAECIsAvailable()) {
- EXPECT_EQ(0, audio_device()->EnableBuiltInAEC(false));
- }
- }
-
- void StartPlayout() {
- EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
- EXPECT_FALSE(audio_device()->Playing());
- EXPECT_EQ(0, audio_device()->InitPlayout());
- EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
- EXPECT_EQ(0, audio_device()->StartPlayout());
- EXPECT_TRUE(audio_device()->Playing());
- }
-
- void StopPlayout() {
- EXPECT_EQ(0, audio_device()->StopPlayout());
- EXPECT_FALSE(audio_device()->Playing());
- EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
- }
-
- void StartRecording() {
- EXPECT_FALSE(audio_device()->RecordingIsInitialized());
- EXPECT_FALSE(audio_device()->Recording());
- EXPECT_EQ(0, audio_device()->InitRecording());
- EXPECT_TRUE(audio_device()->RecordingIsInitialized());
- EXPECT_EQ(0, audio_device()->StartRecording());
- EXPECT_TRUE(audio_device()->Recording());
- }
-
- void StopRecording() {
- EXPECT_EQ(0, audio_device()->StopRecording());
- EXPECT_FALSE(audio_device()->Recording());
- }
-
- int GetMaxSpeakerVolume() const {
- uint32_t max_volume(0);
- EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
- return max_volume;
- }
-
- int GetMinSpeakerVolume() const {
- uint32_t min_volume(0);
- EXPECT_EQ(0, audio_device()->MinSpeakerVolume(&min_volume));
- return min_volume;
- }
-
- int GetSpeakerVolume() const {
- uint32_t volume(0);
- EXPECT_EQ(0, audio_device()->SpeakerVolume(&volume));
- return volume;
- }
-
- rtc::Event test_is_done_;
- std::unique_ptr<TaskQueueFactory> task_queue_factory_;
- rtc::scoped_refptr<AudioDeviceModule> audio_device_;
- AudioParameters playout_parameters_;
- AudioParameters record_parameters_;
- std::unique_ptr<BuildInfo> build_info_;
-};
-
-TEST_F(AudioDeviceTest, ConstructDestruct) {
- // Using the test fixture to create and destruct the audio device module.
-}
-
-// We always ask for a default audio layer when the ADM is constructed. But the
-// ADM will then internally set the best suitable combination of audio layers,
-// for input and output based on if low-latency output and/or input audio in
-// combination with OpenSL ES is supported or not. This test ensures that the
-// correct selection is done.
-TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
- const AudioDeviceModule::AudioLayer audio_layer = GetActiveAudioLayer();
- bool low_latency_output = audio_manager()->IsLowLatencyPlayoutSupported();
- bool low_latency_input = audio_manager()->IsLowLatencyRecordSupported();
- bool aaudio = audio_manager()->IsAAudioSupported();
- AudioDeviceModule::AudioLayer expected_audio_layer;
- if (aaudio) {
- expected_audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
- } else if (low_latency_output && low_latency_input) {
- expected_audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
- } else if (low_latency_output && !low_latency_input) {
- expected_audio_layer =
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
- } else {
- expected_audio_layer = AudioDeviceModule::kAndroidJavaAudio;
- }
- EXPECT_EQ(expected_audio_layer, audio_layer);
-}
-
-// Verify that it is possible to explicitly create the two types of supported
-// ADMs. These two tests overrides the default selection of native audio layer
-// by ignoring if the device supports low-latency output or not.
-TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) {
- AudioDeviceModule::AudioLayer expected_layer =
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
- AudioDeviceModule::AudioLayer active_layer =
- TestActiveAudioLayer(expected_layer);
- EXPECT_EQ(expected_layer, active_layer);
-}
-
-TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) {
- AudioDeviceModule::AudioLayer expected_layer =
- AudioDeviceModule::kAndroidJavaAudio;
- AudioDeviceModule::AudioLayer active_layer =
- TestActiveAudioLayer(expected_layer);
- EXPECT_EQ(expected_layer, active_layer);
-}
-
-TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForOpenSLInBothDirections) {
- AudioDeviceModule::AudioLayer expected_layer =
- AudioDeviceModule::kAndroidOpenSLESAudio;
- AudioDeviceModule::AudioLayer active_layer =
- TestActiveAudioLayer(expected_layer);
- EXPECT_EQ(expected_layer, active_layer);
-}
-
-// TODO(bugs.webrtc.org/8914)
-#if !defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
-#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
- DISABLED_CorrectAudioLayerIsUsedForAAudioInBothDirections
-#else
-#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
- CorrectAudioLayerIsUsedForAAudioInBothDirections
-#endif
-TEST_F(AudioDeviceTest,
- MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections) {
- AudioDeviceModule::AudioLayer expected_layer =
- AudioDeviceModule::kAndroidAAudioAudio;
- AudioDeviceModule::AudioLayer active_layer =
- TestActiveAudioLayer(expected_layer);
- EXPECT_EQ(expected_layer, active_layer);
-}
-
-// TODO(bugs.webrtc.org/8914)
-#if !defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
-#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
- DISABLED_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
-#else
-#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
- CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
-#endif
-TEST_F(AudioDeviceTest,
- MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo) {
- AudioDeviceModule::AudioLayer expected_layer =
- AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio;
- AudioDeviceModule::AudioLayer active_layer =
- TestActiveAudioLayer(expected_layer);
- EXPECT_EQ(expected_layer, active_layer);
-}
-
-// The Android ADM supports two different delay reporting modes. One for the
-// low-latency output path (in combination with OpenSL ES), and one for the
-// high-latency output path (Java backends in both directions). These two tests
-// verifies that the audio manager reports correct delay estimate given the
-// selected audio layer. Note that, this delay estimate will only be utilized
-// if the HW AEC is disabled.
-TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
- EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds,
- TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio));
-}
-
-TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
- EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
- TestDelayOnAudioLayer(
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
-}
-
-// Ensure that the ADM internal audio device buffer is configured to use the
-// correct set of parameters.
-TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) {
- EXPECT_EQ(playout_parameters_.sample_rate(),
- static_cast<int>(audio_device_buffer()->PlayoutSampleRate()));
- EXPECT_EQ(record_parameters_.sample_rate(),
- static_cast<int>(audio_device_buffer()->RecordingSampleRate()));
- EXPECT_EQ(playout_parameters_.channels(),
- audio_device_buffer()->PlayoutChannels());
- EXPECT_EQ(record_parameters_.channels(),
- audio_device_buffer()->RecordingChannels());
-}
-
-TEST_F(AudioDeviceTest, InitTerminate) {
- // Initialization is part of the test fixture.
- EXPECT_TRUE(audio_device()->Initialized());
- EXPECT_EQ(0, audio_device()->Terminate());
- EXPECT_FALSE(audio_device()->Initialized());
-}
-
-TEST_F(AudioDeviceTest, Devices) {
- // Device enumeration is not supported. Verify fixed values only.
- EXPECT_EQ(1, audio_device()->PlayoutDevices());
- EXPECT_EQ(1, audio_device()->RecordingDevices());
-}
-
-TEST_F(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) {
- // The OpenSL ES output audio path does not support volume control.
- if (!AudioLayerSupportsVolumeControl())
- return;
- bool available;
- EXPECT_EQ(0, audio_device()->SpeakerVolumeIsAvailable(&available));
- EXPECT_TRUE(available);
-}
-
-TEST_F(AudioDeviceTest, MaxSpeakerVolumeIsPositive) {
- // The OpenSL ES output audio path does not support volume control.
- if (!AudioLayerSupportsVolumeControl())
- return;
- StartPlayout();
- EXPECT_GT(GetMaxSpeakerVolume(), 0);
- StopPlayout();
-}
-
-TEST_F(AudioDeviceTest, MinSpeakerVolumeIsZero) {
- // The OpenSL ES output audio path does not support volume control.
- if (!AudioLayerSupportsVolumeControl())
- return;
- EXPECT_EQ(GetMinSpeakerVolume(), 0);
-}
-
-TEST_F(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) {
- // The OpenSL ES output audio path does not support volume control.
- if (!AudioLayerSupportsVolumeControl())
- return;
- const int default_volume = GetSpeakerVolume();
- EXPECT_GE(default_volume, GetMinSpeakerVolume());
- EXPECT_LE(default_volume, GetMaxSpeakerVolume());
-}
-
-TEST_F(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
- // The OpenSL ES output audio path does not support volume control.
- if (!AudioLayerSupportsVolumeControl())
- return;
- const int default_volume = GetSpeakerVolume();
- const int max_volume = GetMaxSpeakerVolume();
- EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
- int new_volume = GetSpeakerVolume();
- EXPECT_EQ(new_volume, max_volume);
- EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume));
-}
-
-// Tests that playout can be initiated, started and stopped. No audio callback
-// is registered in this test.
-TEST_F(AudioDeviceTest, StartStopPlayout) {
- StartPlayout();
- StopPlayout();
- StartPlayout();
- StopPlayout();
-}
-
-// Tests that recording can be initiated, started and stopped. No audio callback
-// is registered in this test.
-TEST_F(AudioDeviceTest, StartStopRecording) {
- StartRecording();
- StopRecording();
- StartRecording();
- StopRecording();
-}
-
-// Verify that calling StopPlayout() will leave us in an uninitialized state
-// which will require a new call to InitPlayout(). This test does not call
-// StartPlayout() while being uninitialized since doing so will hit a
-// RTC_DCHECK and death tests are not supported on Android.
-TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
- EXPECT_EQ(0, audio_device()->InitPlayout());
- EXPECT_EQ(0, audio_device()->StartPlayout());
- EXPECT_EQ(0, audio_device()->StopPlayout());
- EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
-}
-
-// Verify that calling StopRecording() will leave us in an uninitialized state
-// which will require a new call to InitRecording(). This test does not call
-// StartRecording() while being uninitialized since doing so will hit a
-// RTC_DCHECK and death tests are not supported on Android.
-TEST_F(AudioDeviceTest, StopRecordingRequiresInitToRestart) {
- EXPECT_EQ(0, audio_device()->InitRecording());
- EXPECT_EQ(0, audio_device()->StartRecording());
- EXPECT_EQ(0, audio_device()->StopRecording());
- EXPECT_FALSE(audio_device()->RecordingIsInitialized());
-}
-
-// Start playout and verify that the native audio layer starts asking for real
-// audio samples to play out using the NeedMorePlayData callback.
-TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
- MockAudioTransportAndroid mock(kPlayout);
- mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
- EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
- kBytesPerSample, playout_channels(),
- playout_sample_rate(), NotNull(), _, _, _))
- .Times(AtLeast(kNumCallbacks));
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- StartPlayout();
- test_is_done_.Wait(kTestTimeOutInMilliseconds);
- StopPlayout();
-}
-
-// Start recording and verify that the native audio layer starts feeding real
-// audio samples via the RecordedDataIsAvailable callback.
-// TODO(henrika): investigate if it is possible to perform a sanity check of
-// delay estimates as well (argument #6).
-TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
- MockAudioTransportAndroid mock(kRecording);
- mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
- EXPECT_CALL(
- mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
- kBytesPerSample, record_channels(),
- record_sample_rate(), _, 0, 0, false, _, _))
- .Times(AtLeast(kNumCallbacks));
-
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- StartRecording();
- test_is_done_.Wait(kTestTimeOutInMilliseconds);
- StopRecording();
-}
-
-// Start playout and recording (full-duplex audio) and verify that audio is
-// active in both directions.
-TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
- MockAudioTransportAndroid mock(kPlayout | kRecording);
- mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
- EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
- kBytesPerSample, playout_channels(),
- playout_sample_rate(), NotNull(), _, _, _))
- .Times(AtLeast(kNumCallbacks));
- EXPECT_CALL(
- mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
- kBytesPerSample, record_channels(),
- record_sample_rate(), _, 0, 0, false, _, _))
- .Times(AtLeast(kNumCallbacks));
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- StartPlayout();
- StartRecording();
- test_is_done_.Wait(kTestTimeOutInMilliseconds);
- StopRecording();
- StopPlayout();
-}
-
-// Start playout and read audio from an external PCM file when the audio layer
-// asks for data to play out. Real audio is played out in this test but it does
-// not contain any explicit verification that the audio quality is perfect.
-TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
- // TODO(henrika): extend test when mono output is supported.
- EXPECT_EQ(1u, playout_channels());
- NiceMock<MockAudioTransportAndroid> mock(kPlayout);
- const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
- std::string file_name = GetFileName(playout_sample_rate());
- std::unique_ptr<FileAudioStream> file_audio_stream(
- new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
- mock.HandleCallbacks(&test_is_done_, file_audio_stream.get(), num_callbacks);
- // SetMaxPlayoutVolume();
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- StartPlayout();
- test_is_done_.Wait(kTestTimeOutInMilliseconds);
- StopPlayout();
-}
-
-// Start playout and recording and store recorded data in an intermediate FIFO
-// buffer from which the playout side then reads its samples in the same order
-// as they were stored. Under ideal circumstances, a callback sequence would
-// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
-// means 'packet played'. Under such conditions, the FIFO would only contain
-// one packet on average. However, under more realistic conditions, the size
-// of the FIFO will vary more due to an unbalance between the two sides.
-// This test tries to verify that the device maintains a balanced callback-
-// sequence by running in loopback for ten seconds while measuring the size
-// (max and average) of the FIFO. The size of the FIFO is increased by the
-// recording side and decreased by the playout side.
-// TODO(henrika): tune the final test parameters after running tests on several
-// different devices.
-// Disabling this test on bots since it is difficult to come up with a robust
-// test condition that all worked as intended. The main issue is that, when
-// swarming is used, an initial latency can be built up when the both sides
-// starts at different times. Hence, the test can fail even if audio works
-// as intended. Keeping the test so it can be enabled manually.
-// http://bugs.webrtc.org/7744
-TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
- EXPECT_EQ(record_channels(), playout_channels());
- EXPECT_EQ(record_sample_rate(), playout_sample_rate());
- NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
- std::unique_ptr<FifoAudioStream> fifo_audio_stream(
- new FifoAudioStream(playout_frames_per_10ms_buffer()));
- mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
- kFullDuplexTimeInSec * kNumCallbacksPerSecond);
- SetMaxPlayoutVolume();
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- StartRecording();
- StartPlayout();
- test_is_done_.Wait(
- std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
- StopPlayout();
- StopRecording();
-
- // These thresholds are set rather high to accomodate differences in hardware
- // in several devices, so this test can be used in swarming.
- // See http://bugs.webrtc.org/6464
- EXPECT_LE(fifo_audio_stream->average_size(), 60u);
- EXPECT_LE(fifo_audio_stream->largest_size(), 70u);
-}
-
-// Measures loopback latency and reports the min, max and average values for
-// a full duplex audio session.
-// The latency is measured like so:
-// - Insert impulses periodically on the output side.
-// - Detect the impulses on the input side.
-// - Measure the time difference between the transmit time and receive time.
-// - Store time differences in a vector and calculate min, max and average.
-// This test requires a special hardware called Audio Loopback Dongle.
-// See http://source.android.com/devices/audio/loopback.html for details.
-TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
- EXPECT_EQ(record_channels(), playout_channels());
- EXPECT_EQ(record_sample_rate(), playout_sample_rate());
- NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
- std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
- new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
- mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
- kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
- EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
- SetMaxPlayoutVolume();
- DisableBuiltInAECIfAvailable();
- StartRecording();
- StartPlayout();
- test_is_done_.Wait(
- std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
- StopPlayout();
- StopRecording();
- // Verify that the correct number of transmitted impulses are detected.
- EXPECT_EQ(latency_audio_stream->num_latency_values(),
- static_cast<size_t>(
- kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
- latency_audio_stream->PrintResults();
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/audio_manager.cc b/modules/audio_device/android/audio_manager.cc
deleted file mode 100644
index 0b55496..0000000
--- a/modules/audio_device/android/audio_manager.cc
+++ /dev/null
@@ -1,318 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/android/audio_manager.h"
-
-#include <utility>
-
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/utility/include/helpers_android.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/platform_thread.h"
-
-namespace webrtc {
-
-// AudioManager::JavaAudioManager implementation
-AudioManager::JavaAudioManager::JavaAudioManager(
- NativeRegistration* native_reg,
- std::unique_ptr<GlobalRef> audio_manager)
- : audio_manager_(std::move(audio_manager)),
- init_(native_reg->GetMethodId("init", "()Z")),
- dispose_(native_reg->GetMethodId("dispose", "()V")),
- is_communication_mode_enabled_(
- native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
- is_device_blacklisted_for_open_sles_usage_(
- native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
- "()Z")) {
- RTC_LOG(LS_INFO) << "JavaAudioManager::ctor";
-}
-
-AudioManager::JavaAudioManager::~JavaAudioManager() {
- RTC_LOG(LS_INFO) << "JavaAudioManager::~dtor";
-}
-
-bool AudioManager::JavaAudioManager::Init() {
- return audio_manager_->CallBooleanMethod(init_);
-}
-
-void AudioManager::JavaAudioManager::Close() {
- audio_manager_->CallVoidMethod(dispose_);
-}
-
-bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() {
- return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_);
-}
-
-bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
- return audio_manager_->CallBooleanMethod(
- is_device_blacklisted_for_open_sles_usage_);
-}
-
-// AudioManager implementation
-AudioManager::AudioManager()
- : j_environment_(JVM::GetInstance()->environment()),
- audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
- initialized_(false),
- hardware_aec_(false),
- hardware_agc_(false),
- hardware_ns_(false),
- low_latency_playout_(false),
- low_latency_record_(false),
- delay_estimate_in_milliseconds_(0) {
- RTC_LOG(LS_INFO) << "ctor";
- RTC_CHECK(j_environment_);
- JNINativeMethod native_methods[] = {
- {"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V",
- reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
- j_native_registration_ = j_environment_->RegisterNatives(
- "org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
- arraysize(native_methods));
- j_audio_manager_.reset(
- new JavaAudioManager(j_native_registration_.get(),
- j_native_registration_->NewObject(
- "<init>", "(J)V", PointerTojlong(this))));
-}
-
-AudioManager::~AudioManager() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK(thread_checker_.IsCurrent());
- Close();
-}
-
-void AudioManager::SetActiveAudioLayer(
- AudioDeviceModule::AudioLayer audio_layer) {
- RTC_LOG(LS_INFO) << "SetActiveAudioLayer: " << audio_layer;
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- // Store the currently utilized audio layer.
- audio_layer_ = audio_layer;
- // The delay estimate can take one of two fixed values depending on if the
- // device supports low-latency output or not. However, it is also possible
- // that the user explicitly selects the high-latency audio path, hence we use
- // the selected `audio_layer` here to set the delay estimate.
- delay_estimate_in_milliseconds_ =
- (audio_layer == AudioDeviceModule::kAndroidJavaAudio)
- ? kHighLatencyModeDelayEstimateInMilliseconds
- : kLowLatencyModeDelayEstimateInMilliseconds;
- RTC_LOG(LS_INFO) << "delay_estimate_in_milliseconds: "
- << delay_estimate_in_milliseconds_;
-}
-
-SLObjectItf AudioManager::GetOpenSLEngine() {
- RTC_LOG(LS_INFO) << "GetOpenSLEngine";
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Only allow usage of OpenSL ES if such an audio layer has been specified.
- if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
- audio_layer_ !=
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
- RTC_LOG(LS_INFO)
- << "Unable to create OpenSL engine for the current audio layer: "
- << audio_layer_;
- return nullptr;
- }
- // OpenSL ES for Android only supports a single engine per application.
- // If one already has been created, return existing object instead of
- // creating a new.
- if (engine_object_.Get() != nullptr) {
- RTC_LOG(LS_WARNING)
- << "The OpenSL ES engine object has already been created";
- return engine_object_.Get();
- }
- // Create the engine object in thread safe mode.
- const SLEngineOption option[] = {
- {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
- SLresult result =
- slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
- if (result != SL_RESULT_SUCCESS) {
- RTC_LOG(LS_ERROR) << "slCreateEngine() failed: "
- << GetSLErrorString(result);
- engine_object_.Reset();
- return nullptr;
- }
- // Realize the SL Engine in synchronous mode.
- result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
- if (result != SL_RESULT_SUCCESS) {
- RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result);
- engine_object_.Reset();
- return nullptr;
- }
- // Finally return the SLObjectItf interface of the engine object.
- return engine_object_.Get();
-}
-
-bool AudioManager::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
- if (!j_audio_manager_->Init()) {
- RTC_LOG(LS_ERROR) << "Init() failed";
- return false;
- }
- initialized_ = true;
- return true;
-}
-
-bool AudioManager::Close() {
- RTC_LOG(LS_INFO) << "Close";
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_)
- return true;
- j_audio_manager_->Close();
- initialized_ = false;
- return true;
-}
-
-bool AudioManager::IsCommunicationModeEnabled() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return j_audio_manager_->IsCommunicationModeEnabled();
-}
-
-bool AudioManager::IsAcousticEchoCancelerSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return hardware_aec_;
-}
-
-bool AudioManager::IsAutomaticGainControlSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return hardware_agc_;
-}
-
-bool AudioManager::IsNoiseSuppressorSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return hardware_ns_;
-}
-
-bool AudioManager::IsLowLatencyPlayoutSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Some devices are blacklisted for usage of OpenSL ES even if they report
- // that low-latency playout is supported. See b/21485703 for details.
- return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage()
- ? false
- : low_latency_playout_;
-}
-
-bool AudioManager::IsLowLatencyRecordSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return low_latency_record_;
-}
-
-bool AudioManager::IsProAudioSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- // TODO(henrika): return the state independently of if OpenSL ES is
- // blacklisted or not for now. We could use the same approach as in
- // IsLowLatencyPlayoutSupported() but I can't see the need for it yet.
- return pro_audio_;
-}
-
-// TODO(henrika): improve comments...
-bool AudioManager::IsAAudioSupported() const {
-#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
- return a_audio_;
-#else
- return false;
-#endif
-}
-
-bool AudioManager::IsStereoPlayoutSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return (playout_parameters_.channels() == 2);
-}
-
-bool AudioManager::IsStereoRecordSupported() const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- return (record_parameters_.channels() == 2);
-}
-
-int AudioManager::GetDelayEstimateInMilliseconds() const {
- return delay_estimate_in_milliseconds_;
-}
-
-JNI_FUNCTION_ALIGN
-void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
- jobject obj,
- jint sample_rate,
- jint output_channels,
- jint input_channels,
- jboolean hardware_aec,
- jboolean hardware_agc,
- jboolean hardware_ns,
- jboolean low_latency_output,
- jboolean low_latency_input,
- jboolean pro_audio,
- jboolean a_audio,
- jint output_buffer_size,
- jint input_buffer_size,
- jlong native_audio_manager) {
- webrtc::AudioManager* this_object =
- reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
- this_object->OnCacheAudioParameters(
- env, sample_rate, output_channels, input_channels, hardware_aec,
- hardware_agc, hardware_ns, low_latency_output, low_latency_input,
- pro_audio, a_audio, output_buffer_size, input_buffer_size);
-}
-
-void AudioManager::OnCacheAudioParameters(JNIEnv* env,
- jint sample_rate,
- jint output_channels,
- jint input_channels,
- jboolean hardware_aec,
- jboolean hardware_agc,
- jboolean hardware_ns,
- jboolean low_latency_output,
- jboolean low_latency_input,
- jboolean pro_audio,
- jboolean a_audio,
- jint output_buffer_size,
- jint input_buffer_size) {
- RTC_LOG(LS_INFO)
- << "OnCacheAudioParameters: "
- "hardware_aec: "
- << static_cast<bool>(hardware_aec)
- << ", hardware_agc: " << static_cast<bool>(hardware_agc)
- << ", hardware_ns: " << static_cast<bool>(hardware_ns)
- << ", low_latency_output: " << static_cast<bool>(low_latency_output)
- << ", low_latency_input: " << static_cast<bool>(low_latency_input)
- << ", pro_audio: " << static_cast<bool>(pro_audio)
- << ", a_audio: " << static_cast<bool>(a_audio)
- << ", sample_rate: " << static_cast<int>(sample_rate)
- << ", output_channels: " << static_cast<int>(output_channels)
- << ", input_channels: " << static_cast<int>(input_channels)
- << ", output_buffer_size: " << static_cast<int>(output_buffer_size)
- << ", input_buffer_size: " << static_cast<int>(input_buffer_size);
- RTC_DCHECK(thread_checker_.IsCurrent());
- hardware_aec_ = hardware_aec;
- hardware_agc_ = hardware_agc;
- hardware_ns_ = hardware_ns;
- low_latency_playout_ = low_latency_output;
- low_latency_record_ = low_latency_input;
- pro_audio_ = pro_audio;
- a_audio_ = a_audio;
- playout_parameters_.reset(sample_rate, static_cast<size_t>(output_channels),
- static_cast<size_t>(output_buffer_size));
- record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels),
- static_cast<size_t>(input_buffer_size));
-}
-
-const AudioParameters& AudioManager::GetPlayoutAudioParameters() {
- RTC_CHECK(playout_parameters_.is_valid());
- RTC_DCHECK(thread_checker_.IsCurrent());
- return playout_parameters_;
-}
-
-const AudioParameters& AudioManager::GetRecordAudioParameters() {
- RTC_CHECK(record_parameters_.is_valid());
- RTC_DCHECK(thread_checker_.IsCurrent());
- return record_parameters_;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/audio_manager.h b/modules/audio_device/android/audio_manager.h
deleted file mode 100644
index 900fc78..0000000
--- a/modules/audio_device/android/audio_manager.h
+++ /dev/null
@@ -1,225 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
-
-#include <SLES/OpenSLES.h>
-#include <jni.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/opensles_common.h"
-#include "modules/audio_device/audio_device_config.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "modules/utility/include/helpers_android.h"
-#include "modules/utility/include/jvm_android.h"
-
-namespace webrtc {
-
-// Implements support for functions in the WebRTC audio stack for Android that
-// relies on the AudioManager in android.media. It also populates an
-// AudioParameter structure with native audio parameters detected at
-// construction. This class does not make any audio-related modifications
-// unless Init() is called. Caching audio parameters makes no changes but only
-// reads data from the Java side.
-class AudioManager {
- public:
- // Wraps the Java specific parts of the AudioManager into one helper class.
- // Stores method IDs for all supported methods at construction and then
- // allows calls like JavaAudioManager::Close() while hiding the Java/JNI
- // parts that are associated with this call.
- class JavaAudioManager {
- public:
- JavaAudioManager(NativeRegistration* native_registration,
- std::unique_ptr<GlobalRef> audio_manager);
- ~JavaAudioManager();
-
- bool Init();
- void Close();
- bool IsCommunicationModeEnabled();
- bool IsDeviceBlacklistedForOpenSLESUsage();
-
- private:
- std::unique_ptr<GlobalRef> audio_manager_;
- jmethodID init_;
- jmethodID dispose_;
- jmethodID is_communication_mode_enabled_;
- jmethodID is_device_blacklisted_for_open_sles_usage_;
- };
-
- AudioManager();
- ~AudioManager();
-
- // Sets the currently active audio layer combination. Must be called before
- // Init().
- void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer);
-
- // Creates and realizes the main (global) Open SL engine object and returns
- // a reference to it. The engine object is only created at the first call
- // since OpenSL ES for Android only supports a single engine per application.
- // Subsequent calls returns the already created engine. The SL engine object
- // is destroyed when the AudioManager object is deleted. It means that the
- // engine object will be the first OpenSL ES object to be created and last
- // object to be destroyed.
- // Note that NULL will be returned unless the audio layer is specified as
- // AudioDeviceModule::kAndroidOpenSLESAudio or
- // AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio.
- SLObjectItf GetOpenSLEngine();
-
- // Initializes the audio manager and stores the current audio mode.
- bool Init();
- // Revert any setting done by Init().
- bool Close();
-
- // Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION.
- bool IsCommunicationModeEnabled() const;
-
- // Native audio parameters stored during construction.
- const AudioParameters& GetPlayoutAudioParameters();
- const AudioParameters& GetRecordAudioParameters();
-
- // Returns true if the device supports built-in audio effects for AEC, AGC
- // and NS. Some devices can also be blacklisted for use in combination with
- // platform effects and these devices will return false.
- // Can currently only be used in combination with a Java based audio backend
- // for the recoring side (i.e. using the android.media.AudioRecord API).
- bool IsAcousticEchoCancelerSupported() const;
- bool IsAutomaticGainControlSupported() const;
- bool IsNoiseSuppressorSupported() const;
-
- // Returns true if the device supports the low-latency audio paths in
- // combination with OpenSL ES.
- bool IsLowLatencyPlayoutSupported() const;
- bool IsLowLatencyRecordSupported() const;
-
- // Returns true if the device supports (and has been configured for) stereo.
- // Call the Java API WebRtcAudioManager.setStereoOutput/Input() with true as
- // paramter to enable stereo. Default is mono in both directions and the
- // setting is set once and for all when the audio manager object is created.
- // TODO(henrika): stereo is not supported in combination with OpenSL ES.
- bool IsStereoPlayoutSupported() const;
- bool IsStereoRecordSupported() const;
-
- // Returns true if the device supports pro-audio features in combination with
- // OpenSL ES.
- bool IsProAudioSupported() const;
-
- // Returns true if the device supports AAudio.
- bool IsAAudioSupported() const;
-
- // Returns the estimated total delay of this device. Unit is in milliseconds.
- // The vaule is set once at construction and never changes after that.
- // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
- // webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
- int GetDelayEstimateInMilliseconds() const;
-
- private:
- // Called from Java side so we can cache the native audio parameters.
- // This method will be called by the WebRtcAudioManager constructor, i.e.
- // on the same thread that this object is created on.
- static void JNICALL CacheAudioParameters(JNIEnv* env,
- jobject obj,
- jint sample_rate,
- jint output_channels,
- jint input_channels,
- jboolean hardware_aec,
- jboolean hardware_agc,
- jboolean hardware_ns,
- jboolean low_latency_output,
- jboolean low_latency_input,
- jboolean pro_audio,
- jboolean a_audio,
- jint output_buffer_size,
- jint input_buffer_size,
- jlong native_audio_manager);
- void OnCacheAudioParameters(JNIEnv* env,
- jint sample_rate,
- jint output_channels,
- jint input_channels,
- jboolean hardware_aec,
- jboolean hardware_agc,
- jboolean hardware_ns,
- jboolean low_latency_output,
- jboolean low_latency_input,
- jboolean pro_audio,
- jboolean a_audio,
- jint output_buffer_size,
- jint input_buffer_size);
-
- // Stores thread ID in the constructor.
- // We can then use RTC_DCHECK_RUN_ON(&thread_checker_) to ensure that
- // other methods are called from the same thread.
- SequenceChecker thread_checker_;
-
- // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
- // construction.
- // Also ensures that DetachCurrentThread() is called at destruction.
- JvmThreadConnector attach_thread_if_needed_;
-
- // Wraps the JNI interface pointer and methods associated with it.
- std::unique_ptr<JNIEnvironment> j_environment_;
-
- // Contains factory method for creating the Java object.
- std::unique_ptr<NativeRegistration> j_native_registration_;
-
- // Wraps the Java specific parts of the AudioManager.
- std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
-
- // Contains the selected audio layer specified by the AudioLayer enumerator
- // in the AudioDeviceModule class.
- AudioDeviceModule::AudioLayer audio_layer_;
-
- // This object is the global entry point of the OpenSL ES API.
- // After creating the engine object, the application can obtain this object‘s
- // SLEngineItf interface. This interface contains creation methods for all
- // the other object types in the API. None of these interface are realized
- // by this class. It only provides access to the global engine object.
- webrtc::ScopedSLObjectItf engine_object_;
-
- // Set to true by Init() and false by Close().
- bool initialized_;
-
- // True if device supports hardware (or built-in) AEC.
- bool hardware_aec_;
- // True if device supports hardware (or built-in) AGC.
- bool hardware_agc_;
- // True if device supports hardware (or built-in) NS.
- bool hardware_ns_;
-
- // True if device supports the low-latency OpenSL ES audio path for output.
- bool low_latency_playout_;
-
- // True if device supports the low-latency OpenSL ES audio path for input.
- bool low_latency_record_;
-
- // True if device supports the low-latency OpenSL ES pro-audio path.
- bool pro_audio_;
-
- // True if device supports the low-latency AAudio audio path.
- bool a_audio_;
-
- // The delay estimate can take one of two fixed values depending on if the
- // device supports low-latency output or not.
- int delay_estimate_in_milliseconds_;
-
- // Contains native parameters (e.g. sample rate, channel configuration).
- // Set at construction in OnCacheAudioParameters() which is called from
- // Java on the same thread as this object is created on.
- AudioParameters playout_parameters_;
- AudioParameters record_parameters_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
diff --git a/modules/audio_device/android/audio_manager_unittest.cc b/modules/audio_device/android/audio_manager_unittest.cc
deleted file mode 100644
index 093eddd..0000000
--- a/modules/audio_device/android/audio_manager_unittest.cc
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/android/audio_manager.h"
-
-#include <SLES/OpenSLES_Android.h>
-
-#include "modules/audio_device/android/build_info.h"
-#include "modules/audio_device/android/ensure_initialized.h"
-#include "rtc_base/arraysize.h"
-#include "test/gtest.h"
-
-#define PRINT(...) fprintf(stderr, __VA_ARGS__);
-
-namespace webrtc {
-
-static const char kTag[] = " ";
-
-class AudioManagerTest : public ::testing::Test {
- protected:
- AudioManagerTest() {
- // One-time initialization of JVM and application context. Ensures that we
- // can do calls between C++ and Java.
- webrtc::audiodevicemodule::EnsureInitialized();
- audio_manager_.reset(new AudioManager());
- SetActiveAudioLayer();
- playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
- record_parameters_ = audio_manager()->GetRecordAudioParameters();
- }
-
- AudioManager* audio_manager() const { return audio_manager_.get(); }
-
- // A valid audio layer must always be set before calling Init(), hence we
- // might as well make it a part of the test fixture.
- void SetActiveAudioLayer() {
- EXPECT_EQ(0, audio_manager()->GetDelayEstimateInMilliseconds());
- audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
- EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds());
- }
-
- // One way to ensure that the engine object is valid is to create an
- // SL Engine interface since it exposes creation methods of all the OpenSL ES
- // object types and it is only supported on the engine object. This method
- // also verifies that the engine interface supports at least one interface.
- // Note that, the test below is not a full test of the SLEngineItf object
- // but only a simple sanity test to check that the global engine object is OK.
- void ValidateSLEngine(SLObjectItf engine_object) {
- EXPECT_NE(nullptr, engine_object);
- // Get the SL Engine interface which is exposed by the engine object.
- SLEngineItf engine;
- SLresult result =
- (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine);
- EXPECT_EQ(result, SL_RESULT_SUCCESS) << "GetInterface() on engine failed";
- // Ensure that the SL Engine interface exposes at least one interface.
- SLuint32 object_id = SL_OBJECTID_ENGINE;
- SLuint32 num_supported_interfaces = 0;
- result = (*engine)->QueryNumSupportedInterfaces(engine, object_id,
- &num_supported_interfaces);
- EXPECT_EQ(result, SL_RESULT_SUCCESS)
- << "QueryNumSupportedInterfaces() failed";
- EXPECT_GE(num_supported_interfaces, 1u);
- }
-
- std::unique_ptr<AudioManager> audio_manager_;
- AudioParameters playout_parameters_;
- AudioParameters record_parameters_;
-};
-
-TEST_F(AudioManagerTest, ConstructDestruct) {}
-
-// It should not be possible to create an OpenSL engine object if Java based
-// audio is requested in both directions.
-TEST_F(AudioManagerTest, GetOpenSLEngineShouldFailForJavaAudioLayer) {
- audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
- SLObjectItf engine_object = audio_manager()->GetOpenSLEngine();
- EXPECT_EQ(nullptr, engine_object);
-}
-
-// It should be possible to create an OpenSL engine object if OpenSL ES based
-// audio is requested in any direction.
-TEST_F(AudioManagerTest, GetOpenSLEngineShouldSucceedForOpenSLESAudioLayer) {
- // List of supported audio layers that uses OpenSL ES audio.
- const AudioDeviceModule::AudioLayer opensles_audio[] = {
- AudioDeviceModule::kAndroidOpenSLESAudio,
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio};
- // Verify that the global (singleton) OpenSL Engine can be acquired for all
- // audio layes that uses OpenSL ES. Note that the engine is only created once.
- for (const AudioDeviceModule::AudioLayer audio_layer : opensles_audio) {
- audio_manager()->SetActiveAudioLayer(audio_layer);
- SLObjectItf engine_object = audio_manager()->GetOpenSLEngine();
- EXPECT_NE(nullptr, engine_object);
- // Perform a simple sanity check of the created engine object.
- ValidateSLEngine(engine_object);
- }
-}
-
-TEST_F(AudioManagerTest, InitClose) {
- EXPECT_TRUE(audio_manager()->Init());
- EXPECT_TRUE(audio_manager()->Close());
-}
-
-TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) {
- PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
- audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No");
-}
-
-TEST_F(AudioManagerTest, IsAutomaticGainControlSupported) {
- EXPECT_FALSE(audio_manager()->IsAutomaticGainControlSupported());
-}
-
-TEST_F(AudioManagerTest, IsNoiseSuppressorSupported) {
- PRINT("%sNoise Suppressor support: %s\n", kTag,
- audio_manager()->IsNoiseSuppressorSupported() ? "Yes" : "No");
-}
-
-TEST_F(AudioManagerTest, IsLowLatencyPlayoutSupported) {
- PRINT("%sLow latency output support: %s\n", kTag,
- audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No");
-}
-
-TEST_F(AudioManagerTest, IsLowLatencyRecordSupported) {
- PRINT("%sLow latency input support: %s\n", kTag,
- audio_manager()->IsLowLatencyRecordSupported() ? "Yes" : "No");
-}
-
-TEST_F(AudioManagerTest, IsProAudioSupported) {
- PRINT("%sPro audio support: %s\n", kTag,
- audio_manager()->IsProAudioSupported() ? "Yes" : "No");
-}
-
-// Verify that playout side is configured for mono by default.
-TEST_F(AudioManagerTest, IsStereoPlayoutSupported) {
- EXPECT_FALSE(audio_manager()->IsStereoPlayoutSupported());
-}
-
-// Verify that recording side is configured for mono by default.
-TEST_F(AudioManagerTest, IsStereoRecordSupported) {
- EXPECT_FALSE(audio_manager()->IsStereoRecordSupported());
-}
-
-TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
- const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
- const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
- PRINT("PLAYOUT:\n");
- PRINT("%saudio layer: %s\n", kTag,
- low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
- PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
- PRINT("%schannels: %zu\n", kTag, playout_parameters_.channels());
- PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
- playout_parameters_.frames_per_buffer(),
- playout_parameters_.GetBufferSizeInMilliseconds());
- PRINT("RECORD: \n");
- PRINT("%saudio layer: %s\n", kTag,
- low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
- PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
- PRINT("%schannels: %zu\n", kTag, record_parameters_.channels());
- PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
- record_parameters_.frames_per_buffer(),
- record_parameters_.GetBufferSizeInMilliseconds());
-}
-
-// The audio device module only suppors the same sample rate in both directions.
-// In addition, in full-duplex low-latency mode (OpenSL ES), both input and
-// output must use the same native buffer size to allow for usage of the fast
-// audio track in Android.
-TEST_F(AudioManagerTest, VerifyAudioParameters) {
- const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
- const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
- EXPECT_EQ(playout_parameters_.sample_rate(),
- record_parameters_.sample_rate());
- if (low_latency_out && low_latency_in) {
- EXPECT_EQ(playout_parameters_.frames_per_buffer(),
- record_parameters_.frames_per_buffer());
- }
-}
-
-// Add device-specific information to the test for logging purposes.
-TEST_F(AudioManagerTest, ShowDeviceInfo) {
- BuildInfo build_info;
- PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str());
- PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str());
- PRINT("%smanufacturer: %s\n", kTag,
- build_info.GetDeviceManufacturer().c_str());
-}
-
-// Add Android build information to the test for logging purposes.
-TEST_F(AudioManagerTest, ShowBuildInfo) {
- BuildInfo build_info;
- PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str());
- PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str());
- PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str());
- PRINT("%sSDK version: %d\n", kTag, build_info.GetSdkVersion());
-}
-
-// Basic test of the AudioParameters class using default construction where
-// all members are set to zero.
-TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
- AudioParameters params;
- EXPECT_FALSE(params.is_valid());
- EXPECT_EQ(0, params.sample_rate());
- EXPECT_EQ(0U, params.channels());
- EXPECT_EQ(0U, params.frames_per_buffer());
- EXPECT_EQ(0U, params.frames_per_10ms_buffer());
- EXPECT_EQ(0U, params.GetBytesPerFrame());
- EXPECT_EQ(0U, params.GetBytesPerBuffer());
- EXPECT_EQ(0U, params.GetBytesPer10msBuffer());
- EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds());
-}
-
-// Basic test of the AudioParameters class using non default construction.
-TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
- const int kSampleRate = 48000;
- const size_t kChannels = 1;
- const size_t kFramesPerBuffer = 480;
- const size_t kFramesPer10msBuffer = 480;
- const size_t kBytesPerFrame = 2;
- const float kBufferSizeInMs = 10.0f;
- AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer);
- EXPECT_TRUE(params.is_valid());
- EXPECT_EQ(kSampleRate, params.sample_rate());
- EXPECT_EQ(kChannels, params.channels());
- EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
- EXPECT_EQ(static_cast<size_t>(kSampleRate / 100),
- params.frames_per_10ms_buffer());
- EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
- EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
- EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
- params.GetBytesPer10msBuffer());
- EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/audio_record_jni.cc b/modules/audio_device/android/audio_record_jni.cc
deleted file mode 100644
index 919eabb..0000000
--- a/modules/audio_device/android/audio_record_jni.cc
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#include "modules/audio_device/android/audio_record_jni.h"
-
-#include <string>
-#include <utility>
-
-#include "modules/audio_device/android/audio_common.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/platform_thread.h"
-#include "rtc_base/time_utils.h"
-#include "system_wrappers/include/metrics.h"
-
-namespace webrtc {
-
-namespace {
-// Scoped class which logs its time of life as a UMA statistic. It generates
-// a histogram which measures the time it takes for a method/scope to execute.
-class ScopedHistogramTimer {
- public:
- explicit ScopedHistogramTimer(const std::string& name)
- : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
- ~ScopedHistogramTimer() {
- const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
- RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
- RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
- }
-
- private:
- const std::string histogram_name_;
- int64_t start_time_ms_;
-};
-} // namespace
-
-// AudioRecordJni::JavaAudioRecord implementation.
-AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
- NativeRegistration* native_reg,
- std::unique_ptr<GlobalRef> audio_record)
- : audio_record_(std::move(audio_record)),
- init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
- start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
- stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
- enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
- enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
-
-AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
-
-int AudioRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
- size_t channels) {
- return audio_record_->CallIntMethod(init_recording_,
- static_cast<jint>(sample_rate),
- static_cast<jint>(channels));
-}
-
-bool AudioRecordJni::JavaAudioRecord::StartRecording() {
- return audio_record_->CallBooleanMethod(start_recording_);
-}
-
-bool AudioRecordJni::JavaAudioRecord::StopRecording() {
- return audio_record_->CallBooleanMethod(stop_recording_);
-}
-
-bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
- return audio_record_->CallBooleanMethod(enable_built_in_aec_,
- static_cast<jboolean>(enable));
-}
-
-bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
- return audio_record_->CallBooleanMethod(enable_built_in_ns_,
- static_cast<jboolean>(enable));
-}
-
-// AudioRecordJni implementation.
-AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
- : j_environment_(JVM::GetInstance()->environment()),
- audio_manager_(audio_manager),
- audio_parameters_(audio_manager->GetRecordAudioParameters()),
- total_delay_in_milliseconds_(0),
- direct_buffer_address_(nullptr),
- direct_buffer_capacity_in_bytes_(0),
- frames_per_buffer_(0),
- initialized_(false),
- recording_(false),
- audio_device_buffer_(nullptr) {
- RTC_LOG(LS_INFO) << "ctor";
- RTC_DCHECK(audio_parameters_.is_valid());
- RTC_CHECK(j_environment_);
- JNINativeMethod native_methods[] = {
- {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
- reinterpret_cast<void*>(
- &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
- {"nativeDataIsRecorded", "(IJ)V",
- reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
- j_native_registration_ = j_environment_->RegisterNatives(
- "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
- arraysize(native_methods));
- j_audio_record_.reset(
- new JavaAudioRecord(j_native_registration_.get(),
- j_native_registration_->NewObject(
- "<init>", "(J)V", PointerTojlong(this))));
- // Detach from this thread since we want to use the checker to verify calls
- // from the Java based audio thread.
- thread_checker_java_.Detach();
-}
-
-AudioRecordJni::~AudioRecordJni() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
-}
-
-int32_t AudioRecordJni::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK(thread_checker_.IsCurrent());
- return 0;
-}
-
-int32_t AudioRecordJni::Terminate() {
- RTC_LOG(LS_INFO) << "Terminate";
- RTC_DCHECK(thread_checker_.IsCurrent());
- StopRecording();
- return 0;
-}
-
-int32_t AudioRecordJni::InitRecording() {
- RTC_LOG(LS_INFO) << "InitRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!recording_);
- ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
- int frames_per_buffer = j_audio_record_->InitRecording(
- audio_parameters_.sample_rate(), audio_parameters_.channels());
- if (frames_per_buffer < 0) {
- direct_buffer_address_ = nullptr;
- RTC_LOG(LS_ERROR) << "InitRecording failed";
- return -1;
- }
- frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
- RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
- const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
- RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
- frames_per_buffer_ * bytes_per_frame);
- RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
- initialized_ = true;
- return 0;
-}
-
-int32_t AudioRecordJni::StartRecording() {
- RTC_LOG(LS_INFO) << "StartRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!recording_);
- if (!initialized_) {
- RTC_DLOG(LS_WARNING)
- << "Recording can not start since InitRecording must succeed first";
- return 0;
- }
- ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
- if (!j_audio_record_->StartRecording()) {
- RTC_LOG(LS_ERROR) << "StartRecording failed";
- return -1;
- }
- recording_ = true;
- return 0;
-}
-
-int32_t AudioRecordJni::StopRecording() {
- RTC_LOG(LS_INFO) << "StopRecording";
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_ || !recording_) {
- return 0;
- }
- if (!j_audio_record_->StopRecording()) {
- RTC_LOG(LS_ERROR) << "StopRecording failed";
- return -1;
- }
- // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
- // next time StartRecording() is called since it will create a new Java
- // thread.
- thread_checker_java_.Detach();
- initialized_ = false;
- recording_ = false;
- direct_buffer_address_ = nullptr;
- return 0;
-}
-
-void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(LS_INFO) << "AttachAudioBuffer";
- RTC_DCHECK(thread_checker_.IsCurrent());
- audio_device_buffer_ = audioBuffer;
- const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
- audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
- const size_t channels = audio_parameters_.channels();
- RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
- audio_device_buffer_->SetRecordingChannels(channels);
- total_delay_in_milliseconds_ =
- audio_manager_->GetDelayEstimateInMilliseconds();
- RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
- RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
- << total_delay_in_milliseconds_;
-}
-
-int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
- RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
- RTC_DCHECK(thread_checker_.IsCurrent());
- return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
-}
-
-int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
- // TODO(henrika): possibly remove when no longer used by any client.
- RTC_CHECK_NOTREACHED();
-}
-
-int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
- RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
- RTC_DCHECK(thread_checker_.IsCurrent());
- return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
-}
-
-JNI_FUNCTION_ALIGN
-void JNICALL AudioRecordJni::CacheDirectBufferAddress(JNIEnv* env,
- jobject obj,
- jobject byte_buffer,
- jlong nativeAudioRecord) {
- webrtc::AudioRecordJni* this_object =
- reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
- this_object->OnCacheDirectBufferAddress(env, byte_buffer);
-}
-
-void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
- jobject byte_buffer) {
- RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!direct_buffer_address_);
- direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
- jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
- RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
- direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
-}
-
-JNI_FUNCTION_ALIGN
-void JNICALL AudioRecordJni::DataIsRecorded(JNIEnv* env,
- jobject obj,
- jint length,
- jlong nativeAudioRecord) {
- webrtc::AudioRecordJni* this_object =
- reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
- this_object->OnDataIsRecorded(length);
-}
-
-// This method is called on a high-priority thread from Java. The name of
-// the thread is 'AudioRecordThread'.
-void AudioRecordJni::OnDataIsRecorded(int length) {
- RTC_DCHECK(thread_checker_java_.IsCurrent());
- if (!audio_device_buffer_) {
- RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
- return;
- }
- audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
- frames_per_buffer_);
- // We provide one (combined) fixed delay estimate for the APM and use the
- // `playDelayMs` parameter only. Components like the AEC only sees the sum
- // of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
- audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
- if (audio_device_buffer_->DeliverRecordedData() == -1) {
- RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
- }
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/audio_record_jni.h b/modules/audio_device/android/audio_record_jni.h
deleted file mode 100644
index 66a6a89..0000000
--- a/modules/audio_device/android/audio_record_jni.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
-
-#include <jni.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "modules/utility/include/helpers_android.h"
-#include "modules/utility/include/jvm_android.h"
-
-namespace webrtc {
-
-// Implements 16-bit mono PCM audio input support for Android using the Java
-// AudioRecord interface. Most of the work is done by its Java counterpart in
-// WebRtcAudioRecord.java. This class is created and lives on a thread in
-// C++-land, but recorded audio buffers are delivered on a high-priority
-// thread managed by the Java class.
-//
-// The Java class makes use of AudioEffect features (mainly AEC) which are
-// first available in Jelly Bean. If it is instantiated running against earlier
-// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
-// separately instead.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will RTC_DCHECK if any method is called on an invalid thread.
-//
-// This class uses JvmThreadConnector to attach to a Java VM if needed
-// and detach when the object goes out of scope. Additional thread checking
-// guarantees that no other (possibly non attached) thread is used.
-class AudioRecordJni {
- public:
- // Wraps the Java specific parts of the AudioRecordJni into one helper class.
- class JavaAudioRecord {
- public:
- JavaAudioRecord(NativeRegistration* native_registration,
- std::unique_ptr<GlobalRef> audio_track);
- ~JavaAudioRecord();
-
- int InitRecording(int sample_rate, size_t channels);
- bool StartRecording();
- bool StopRecording();
- bool EnableBuiltInAEC(bool enable);
- bool EnableBuiltInNS(bool enable);
-
- private:
- std::unique_ptr<GlobalRef> audio_record_;
- jmethodID init_recording_;
- jmethodID start_recording_;
- jmethodID stop_recording_;
- jmethodID enable_built_in_aec_;
- jmethodID enable_built_in_ns_;
- };
-
- explicit AudioRecordJni(AudioManager* audio_manager);
- ~AudioRecordJni();
-
- int32_t Init();
- int32_t Terminate();
-
- int32_t InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
-
- int32_t StartRecording();
- int32_t StopRecording();
- bool Recording() const { return recording_; }
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- int32_t EnableBuiltInAEC(bool enable);
- int32_t EnableBuiltInAGC(bool enable);
- int32_t EnableBuiltInNS(bool enable);
-
- private:
- // Called from Java side so we can cache the address of the Java-manged
- // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
- // is also stored in `direct_buffer_capacity_in_bytes_`.
- // This method will be called by the WebRtcAudioRecord constructor, i.e.,
- // on the same thread that this object is created on.
- static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
- jobject obj,
- jobject byte_buffer,
- jlong nativeAudioRecord);
- void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
-
- // Called periodically by the Java based WebRtcAudioRecord object when
- // recording has started. Each call indicates that there are `length` new
- // bytes recorded in the memory area `direct_buffer_address_` and it is
- // now time to send these to the consumer.
- // This method is called on a high-priority thread from Java. The name of
- // the thread is 'AudioRecordThread'.
- static void JNICALL DataIsRecorded(JNIEnv* env,
- jobject obj,
- jint length,
- jlong nativeAudioRecord);
- void OnDataIsRecorded(int length);
-
- // Stores thread ID in constructor.
- SequenceChecker thread_checker_;
-
- // Stores thread ID in first call to OnDataIsRecorded() from high-priority
- // thread in Java. Detached during construction of this object.
- SequenceChecker thread_checker_java_;
-
- // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
- // construction.
- // Also ensures that DetachCurrentThread() is called at destruction.
- JvmThreadConnector attach_thread_if_needed_;
-
- // Wraps the JNI interface pointer and methods associated with it.
- std::unique_ptr<JNIEnvironment> j_environment_;
-
- // Contains factory method for creating the Java object.
- std::unique_ptr<NativeRegistration> j_native_registration_;
-
- // Wraps the Java specific parts of the AudioRecordJni class.
- std::unique_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
-
- // Raw pointer to the audio manger.
- const AudioManager* audio_manager_;
-
- // Contains audio parameters provided to this class at construction by the
- // AudioManager.
- const AudioParameters audio_parameters_;
-
- // Delay estimate of the total round-trip delay (input + output).
- // Fixed value set once in AttachAudioBuffer() and it can take one out of two
- // possible values. See audio_common.h for details.
- int total_delay_in_milliseconds_;
-
- // Cached copy of address to direct audio buffer owned by `j_audio_record_`.
- void* direct_buffer_address_;
-
- // Number of bytes in the direct audio buffer owned by `j_audio_record_`.
- size_t direct_buffer_capacity_in_bytes_;
-
- // Number audio frames per audio buffer. Each audio frame corresponds to
- // one sample of PCM mono data at 16 bits per sample. Hence, each audio
- // frame contains 2 bytes (given that the Java layer only supports mono).
- // Example: 480 for 48000 Hz or 441 for 44100 Hz.
- size_t frames_per_buffer_;
-
- bool initialized_;
-
- bool recording_;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
- AudioDeviceBuffer* audio_device_buffer_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
diff --git a/modules/audio_device/android/audio_track_jni.cc b/modules/audio_device/android/audio_track_jni.cc
deleted file mode 100644
index 5afa1ec..0000000
--- a/modules/audio_device/android/audio_track_jni.cc
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#include "modules/audio_device/android/audio_track_jni.h"
-
-#include <utility>
-
-#include "modules/audio_device/android/audio_manager.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/platform_thread.h"
-#include "system_wrappers/include/field_trial.h"
-#include "system_wrappers/include/metrics.h"
-
-namespace webrtc {
-
-// AudioTrackJni::JavaAudioTrack implementation.
-AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
- NativeRegistration* native_reg,
- std::unique_ptr<GlobalRef> audio_track)
- : audio_track_(std::move(audio_track)),
- init_playout_(native_reg->GetMethodId("initPlayout", "(IID)I")),
- start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
- stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
- set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
- get_stream_max_volume_(
- native_reg->GetMethodId("getStreamMaxVolume", "()I")),
- get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")),
- get_buffer_size_in_frames_(
- native_reg->GetMethodId("getBufferSizeInFrames", "()I")) {}
-
-AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
-
-bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
- double buffer_size_factor =
- strtod(webrtc::field_trial::FindFullName(
- "WebRTC-AudioDevicePlayoutBufferSizeFactor")
- .c_str(),
- nullptr);
- if (buffer_size_factor == 0)
- buffer_size_factor = 1.0;
- int requested_buffer_size_bytes = audio_track_->CallIntMethod(
- init_playout_, sample_rate, channels, buffer_size_factor);
- // Update UMA histograms for both the requested and actual buffer size.
- if (requested_buffer_size_bytes >= 0) {
- // To avoid division by zero, we assume the sample rate is 48k if an invalid
- // value is found.
- sample_rate = sample_rate <= 0 ? 48000 : sample_rate;
- // This calculation assumes that audio is mono.
- const int requested_buffer_size_ms =
- (requested_buffer_size_bytes * 1000) / (2 * sample_rate);
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeRequestedAudioBufferSizeMs",
- requested_buffer_size_ms, 0, 1000, 100);
- int actual_buffer_size_frames =
- audio_track_->CallIntMethod(get_buffer_size_in_frames_);
- if (actual_buffer_size_frames >= 0) {
- const int actual_buffer_size_ms =
- actual_buffer_size_frames * 1000 / sample_rate;
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeAudioBufferSizeMs",
- actual_buffer_size_ms, 0, 1000, 100);
- }
- return true;
- }
- return false;
-}
-
-bool AudioTrackJni::JavaAudioTrack::StartPlayout() {
- return audio_track_->CallBooleanMethod(start_playout_);
-}
-
-bool AudioTrackJni::JavaAudioTrack::StopPlayout() {
- return audio_track_->CallBooleanMethod(stop_playout_);
-}
-
-bool AudioTrackJni::JavaAudioTrack::SetStreamVolume(int volume) {
- return audio_track_->CallBooleanMethod(set_stream_volume_, volume);
-}
-
-int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() {
- return audio_track_->CallIntMethod(get_stream_max_volume_);
-}
-
-int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
- return audio_track_->CallIntMethod(get_stream_volume_);
-}
-
-// TODO(henrika): possible extend usage of AudioManager and add it as member.
-AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
- : j_environment_(JVM::GetInstance()->environment()),
- audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
- direct_buffer_address_(nullptr),
- direct_buffer_capacity_in_bytes_(0),
- frames_per_buffer_(0),
- initialized_(false),
- playing_(false),
- audio_device_buffer_(nullptr) {
- RTC_LOG(LS_INFO) << "ctor";
- RTC_DCHECK(audio_parameters_.is_valid());
- RTC_CHECK(j_environment_);
- JNINativeMethod native_methods[] = {
- {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
- reinterpret_cast<void*>(
- &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
- {"nativeGetPlayoutData", "(IJ)V",
- reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
- j_native_registration_ = j_environment_->RegisterNatives(
- "org/webrtc/voiceengine/WebRtcAudioTrack", native_methods,
- arraysize(native_methods));
- j_audio_track_.reset(
- new JavaAudioTrack(j_native_registration_.get(),
- j_native_registration_->NewObject(
- "<init>", "(J)V", PointerTojlong(this))));
- // Detach from this thread since we want to use the checker to verify calls
- // from the Java based audio thread.
- thread_checker_java_.Detach();
-}
-
-AudioTrackJni::~AudioTrackJni() {
- RTC_LOG(LS_INFO) << "dtor";
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
-}
-
-int32_t AudioTrackJni::Init() {
- RTC_LOG(LS_INFO) << "Init";
- RTC_DCHECK(thread_checker_.IsCurrent());
- return 0;
-}
-
-int32_t AudioTrackJni::Terminate() {
- RTC_LOG(LS_INFO) << "Terminate";
- RTC_DCHECK(thread_checker_.IsCurrent());
- StopPlayout();
- return 0;
-}
-
-int32_t AudioTrackJni::InitPlayout() {
- RTC_LOG(LS_INFO) << "InitPlayout";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!playing_);
- if (!j_audio_track_->InitPlayout(audio_parameters_.sample_rate(),
- audio_parameters_.channels())) {
- RTC_LOG(LS_ERROR) << "InitPlayout failed";
- return -1;
- }
- initialized_ = true;
- return 0;
-}
-
-int32_t AudioTrackJni::StartPlayout() {
- RTC_LOG(LS_INFO) << "StartPlayout";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!playing_);
- if (!initialized_) {
- RTC_DLOG(LS_WARNING)
- << "Playout can not start since InitPlayout must succeed first";
- return 0;
- }
- if (!j_audio_track_->StartPlayout()) {
- RTC_LOG(LS_ERROR) << "StartPlayout failed";
- return -1;
- }
- playing_ = true;
- return 0;
-}
-
-int32_t AudioTrackJni::StopPlayout() {
- RTC_LOG(LS_INFO) << "StopPlayout";
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_ || !playing_) {
- return 0;
- }
- if (!j_audio_track_->StopPlayout()) {
- RTC_LOG(LS_ERROR) << "StopPlayout failed";
- return -1;
- }
- // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
- // next time StartRecording() is called since it will create a new Java
- // thread.
- thread_checker_java_.Detach();
- initialized_ = false;
- playing_ = false;
- direct_buffer_address_ = nullptr;
- return 0;
-}
-
-int AudioTrackJni::SpeakerVolumeIsAvailable(bool& available) {
- available = true;
- return 0;
-}
-
-int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
- RTC_LOG(LS_INFO) << "SetSpeakerVolume(" << volume << ")";
- RTC_DCHECK(thread_checker_.IsCurrent());
- return j_audio_track_->SetStreamVolume(volume) ? 0 : -1;
-}
-
-int AudioTrackJni::MaxSpeakerVolume(uint32_t& max_volume) const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- max_volume = j_audio_track_->GetStreamMaxVolume();
- return 0;
-}
-
-int AudioTrackJni::MinSpeakerVolume(uint32_t& min_volume) const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- min_volume = 0;
- return 0;
-}
-
-int AudioTrackJni::SpeakerVolume(uint32_t& volume) const {
- RTC_DCHECK(thread_checker_.IsCurrent());
- volume = j_audio_track_->GetStreamVolume();
- RTC_LOG(LS_INFO) << "SpeakerVolume: " << volume;
- return 0;
-}
-
-// TODO(henrika): possibly add stereo support.
-void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(LS_INFO) << "AttachAudioBuffer";
- RTC_DCHECK(thread_checker_.IsCurrent());
- audio_device_buffer_ = audioBuffer;
- const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
- audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
- const size_t channels = audio_parameters_.channels();
- RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
- audio_device_buffer_->SetPlayoutChannels(channels);
-}
-
-JNI_FUNCTION_ALIGN
-void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env,
- jobject obj,
- jobject byte_buffer,
- jlong nativeAudioTrack) {
- webrtc::AudioTrackJni* this_object =
- reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
- this_object->OnCacheDirectBufferAddress(env, byte_buffer);
-}
-
-void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
- jobject byte_buffer) {
- RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!direct_buffer_address_);
- direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
- jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
- RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
- direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
- const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
- frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
- RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
-}
-
-JNI_FUNCTION_ALIGN
-void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env,
- jobject obj,
- jint length,
- jlong nativeAudioTrack) {
- webrtc::AudioTrackJni* this_object =
- reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
- this_object->OnGetPlayoutData(static_cast<size_t>(length));
-}
-
-// This method is called on a high-priority thread from Java. The name of
-// the thread is 'AudioRecordTrack'.
-void AudioTrackJni::OnGetPlayoutData(size_t length) {
- RTC_DCHECK(thread_checker_java_.IsCurrent());
- const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
- RTC_DCHECK_EQ(frames_per_buffer_, length / bytes_per_frame);
- if (!audio_device_buffer_) {
- RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
- return;
- }
- // Pull decoded data (in 16-bit PCM format) from jitter buffer.
- int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
- if (samples <= 0) {
- RTC_LOG(LS_ERROR) << "AudioDeviceBuffer::RequestPlayoutData failed";
- return;
- }
- RTC_DCHECK_EQ(samples, frames_per_buffer_);
- // Copy decoded data into common byte buffer to ensure that it can be
- // written to the Java based audio track.
- samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
- RTC_DCHECK_EQ(length, bytes_per_frame * samples);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/audio_track_jni.h b/modules/audio_device/android/audio_track_jni.h
deleted file mode 100644
index 7eb6908..0000000
--- a/modules/audio_device/android/audio_track_jni.h
+++ /dev/null
@@ -1,161 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
-
-#include <jni.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "modules/utility/include/helpers_android.h"
-#include "modules/utility/include/jvm_android.h"
-
-namespace webrtc {
-
-// Implements 16-bit mono PCM audio output support for Android using the Java
-// AudioTrack interface. Most of the work is done by its Java counterpart in
-// WebRtcAudioTrack.java. This class is created and lives on a thread in
-// C++-land, but decoded audio buffers are requested on a high-priority
-// thread managed by the Java class.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will RTC_DCHECK if any method is called on an invalid thread.
-//
-// This class uses JvmThreadConnector to attach to a Java VM if needed
-// and detach when the object goes out of scope. Additional thread checking
-// guarantees that no other (possibly non attached) thread is used.
-class AudioTrackJni {
- public:
- // Wraps the Java specific parts of the AudioTrackJni into one helper class.
- class JavaAudioTrack {
- public:
- JavaAudioTrack(NativeRegistration* native_registration,
- std::unique_ptr<GlobalRef> audio_track);
- ~JavaAudioTrack();
-
- bool InitPlayout(int sample_rate, int channels);
- bool StartPlayout();
- bool StopPlayout();
- bool SetStreamVolume(int volume);
- int GetStreamMaxVolume();
- int GetStreamVolume();
-
- private:
- std::unique_ptr<GlobalRef> audio_track_;
- jmethodID init_playout_;
- jmethodID start_playout_;
- jmethodID stop_playout_;
- jmethodID set_stream_volume_;
- jmethodID get_stream_max_volume_;
- jmethodID get_stream_volume_;
- jmethodID get_buffer_size_in_frames_;
- };
-
- explicit AudioTrackJni(AudioManager* audio_manager);
- ~AudioTrackJni();
-
- int32_t Init();
- int32_t Terminate();
-
- int32_t InitPlayout();
- bool PlayoutIsInitialized() const { return initialized_; }
-
- int32_t StartPlayout();
- int32_t StopPlayout();
- bool Playing() const { return playing_; }
-
- int SpeakerVolumeIsAvailable(bool& available);
- int SetSpeakerVolume(uint32_t volume);
- int SpeakerVolume(uint32_t& volume) const;
- int MaxSpeakerVolume(uint32_t& max_volume) const;
- int MinSpeakerVolume(uint32_t& min_volume) const;
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- private:
- // Called from Java side so we can cache the address of the Java-manged
- // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
- // is also stored in `direct_buffer_capacity_in_bytes_`.
- // Called on the same thread as the creating thread.
- static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
- jobject obj,
- jobject byte_buffer,
- jlong nativeAudioTrack);
- void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
-
- // Called periodically by the Java based WebRtcAudioTrack object when
- // playout has started. Each call indicates that `length` new bytes should
- // be written to the memory area `direct_buffer_address_` for playout.
- // This method is called on a high-priority thread from Java. The name of
- // the thread is 'AudioTrackThread'.
- static void JNICALL GetPlayoutData(JNIEnv* env,
- jobject obj,
- jint length,
- jlong nativeAudioTrack);
- void OnGetPlayoutData(size_t length);
-
- // Stores thread ID in constructor.
- SequenceChecker thread_checker_;
-
- // Stores thread ID in first call to OnGetPlayoutData() from high-priority
- // thread in Java. Detached during construction of this object.
- SequenceChecker thread_checker_java_;
-
- // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
- // construction.
- // Also ensures that DetachCurrentThread() is called at destruction.
- JvmThreadConnector attach_thread_if_needed_;
-
- // Wraps the JNI interface pointer and methods associated with it.
- std::unique_ptr<JNIEnvironment> j_environment_;
-
- // Contains factory method for creating the Java object.
- std::unique_ptr<NativeRegistration> j_native_registration_;
-
- // Wraps the Java specific parts of the AudioTrackJni class.
- std::unique_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
-
- // Contains audio parameters provided to this class at construction by the
- // AudioManager.
- const AudioParameters audio_parameters_;
-
- // Cached copy of address to direct audio buffer owned by `j_audio_track_`.
- void* direct_buffer_address_;
-
- // Number of bytes in the direct audio buffer owned by `j_audio_track_`.
- size_t direct_buffer_capacity_in_bytes_;
-
- // Number of audio frames per audio buffer. Each audio frame corresponds to
- // one sample of PCM mono data at 16 bits per sample. Hence, each audio
- // frame contains 2 bytes (given that the Java layer only supports mono).
- // Example: 480 for 48000 Hz or 441 for 44100 Hz.
- size_t frames_per_buffer_;
-
- bool initialized_;
-
- bool playing_;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
- // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
- // and therefore outlives this object.
- AudioDeviceBuffer* audio_device_buffer_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
diff --git a/modules/audio_device/android/build_info.cc b/modules/audio_device/android/build_info.cc
deleted file mode 100644
index 916be82..0000000
--- a/modules/audio_device/android/build_info.cc
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/android/build_info.h"
-
-#include "modules/utility/include/helpers_android.h"
-
-namespace webrtc {
-
-BuildInfo::BuildInfo()
- : j_environment_(JVM::GetInstance()->environment()),
- j_build_info_(
- JVM::GetInstance()->GetClass("org/webrtc/voiceengine/BuildInfo")) {}
-
-std::string BuildInfo::GetStringFromJava(const char* name) {
- jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
- jstring j_string =
- static_cast<jstring>(j_build_info_.CallStaticObjectMethod(id));
- return j_environment_->JavaToStdString(j_string);
-}
-
-std::string BuildInfo::GetDeviceModel() {
- return GetStringFromJava("getDeviceModel");
-}
-
-std::string BuildInfo::GetBrand() {
- return GetStringFromJava("getBrand");
-}
-
-std::string BuildInfo::GetDeviceManufacturer() {
- return GetStringFromJava("getDeviceManufacturer");
-}
-
-std::string BuildInfo::GetAndroidBuildId() {
- return GetStringFromJava("getAndroidBuildId");
-}
-
-std::string BuildInfo::GetBuildType() {
- return GetStringFromJava("getBuildType");
-}
-
-std::string BuildInfo::GetBuildRelease() {
- return GetStringFromJava("getBuildRelease");
-}
-
-SdkCode BuildInfo::GetSdkVersion() {
- jmethodID id = j_build_info_.GetStaticMethodId("getSdkVersion", "()I");
- jint j_version = j_build_info_.CallStaticIntMethod(id);
- return static_cast<SdkCode>(j_version);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/build_info.h b/modules/audio_device/android/build_info.h
deleted file mode 100644
index 3647e56..0000000
--- a/modules/audio_device/android/build_info.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
-
-#include <jni.h>
-
-#include <memory>
-#include <string>
-
-#include "modules/utility/include/jvm_android.h"
-
-namespace webrtc {
-
-// This enumeration maps to the values returned by BuildInfo::GetSdkVersion(),
-// indicating the Android release associated with a given SDK version.
-// See https://developer.android.com/guide/topics/manifest/uses-sdk-element.html
-// for details.
-enum SdkCode {
- SDK_CODE_JELLY_BEAN = 16, // Android 4.1
- SDK_CODE_JELLY_BEAN_MR1 = 17, // Android 4.2
- SDK_CODE_JELLY_BEAN_MR2 = 18, // Android 4.3
- SDK_CODE_KITKAT = 19, // Android 4.4
- SDK_CODE_WATCH = 20, // Android 4.4W
- SDK_CODE_LOLLIPOP = 21, // Android 5.0
- SDK_CODE_LOLLIPOP_MR1 = 22, // Android 5.1
- SDK_CODE_MARSHMALLOW = 23, // Android 6.0
- SDK_CODE_N = 24,
-};
-
-// Utility class used to query the Java class (org/webrtc/voiceengine/BuildInfo)
-// for device and Android build information.
-// The calling thread is attached to the JVM at construction if needed and a
-// valid Java environment object is also created.
-// All Get methods must be called on the creating thread. If not, the code will
-// hit RTC_DCHECKs when calling JNIEnvironment::JavaToStdString().
-class BuildInfo {
- public:
- BuildInfo();
- ~BuildInfo() {}
-
- // End-user-visible name for the end product (e.g. "Nexus 6").
- std::string GetDeviceModel();
- // Consumer-visible brand (e.g. "google").
- std::string GetBrand();
- // Manufacturer of the product/hardware (e.g. "motorola").
- std::string GetDeviceManufacturer();
- // Android build ID (e.g. LMY47D).
- std::string GetAndroidBuildId();
- // The type of build (e.g. "user" or "eng").
- std::string GetBuildType();
- // The user-visible version string (e.g. "5.1").
- std::string GetBuildRelease();
- // The user-visible SDK version of the framework (e.g. 21). See SdkCode enum
- // for translation.
- SdkCode GetSdkVersion();
-
- private:
- // Helper method which calls a static getter method with `name` and returns
- // a string from Java.
- std::string GetStringFromJava(const char* name);
-
- // Ensures that this class can access a valid JNI interface pointer even
- // if the creating thread was not attached to the JVM.
- JvmThreadConnector attach_thread_if_needed_;
-
- // Provides access to the JNIEnv interface pointer and the JavaToStdString()
- // method which is used to translate Java strings to std strings.
- std::unique_ptr<JNIEnvironment> j_environment_;
-
- // Holds the jclass object and provides access to CallStaticObjectMethod().
- // Used by GetStringFromJava() during construction only.
- JavaClass j_build_info_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
diff --git a/modules/audio_device/android/ensure_initialized.cc b/modules/audio_device/android/ensure_initialized.cc
deleted file mode 100644
index 59e9c8f..0000000
--- a/modules/audio_device/android/ensure_initialized.cc
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/android/ensure_initialized.h"
-
-#include <jni.h>
-#include <pthread.h>
-#include <stddef.h>
-
-#include "modules/utility/include/jvm_android.h"
-#include "rtc_base/checks.h"
-#include "sdk/android/src/jni/jvm.h"
-
-namespace webrtc {
-namespace audiodevicemodule {
-
-static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
-
-void EnsureInitializedOnce() {
- RTC_CHECK(::webrtc::jni::GetJVM() != nullptr);
-
- JNIEnv* jni = ::webrtc::jni::AttachCurrentThreadIfNeeded();
- JavaVM* jvm = NULL;
- RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
-
- // Initialize the Java environment (currently only used by the audio manager).
- webrtc::JVM::Initialize(jvm);
-}
-
-void EnsureInitialized() {
- RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
-}
-
-} // namespace audiodevicemodule
-} // namespace webrtc
diff --git a/modules/audio_device/android/ensure_initialized.h b/modules/audio_device/android/ensure_initialized.h
deleted file mode 100644
index c1997b4..0000000
--- a/modules/audio_device/android/ensure_initialized.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * 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.
- */
-
-namespace webrtc {
-namespace audiodevicemodule {
-
-void EnsureInitialized();
-
-} // namespace audiodevicemodule
-} // namespace webrtc
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
deleted file mode 100644
index aed8a06..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * 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.voiceengine;
-
-import android.os.Build;
-
-public final class BuildInfo {
- public static String getDevice() {
- return Build.DEVICE;
- }
-
- public static String getDeviceModel() {
- return Build.MODEL;
- }
-
- public static String getProduct() {
- return Build.PRODUCT;
- }
-
- public static String getBrand() {
- return Build.BRAND;
- }
-
- public static String getDeviceManufacturer() {
- return Build.MANUFACTURER;
- }
-
- public static String getAndroidBuildId() {
- return Build.ID;
- }
-
- public static String getBuildType() {
- return Build.TYPE;
- }
-
- public static String getBuildRelease() {
- return Build.VERSION.RELEASE;
- }
-
- public static int getSdkVersion() {
- return Build.VERSION.SDK_INT;
- }
-}
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
deleted file mode 100644
index 92f1c93..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
+++ /dev/null
@@ -1,312 +0,0 @@
-/*
- * 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.voiceengine;
-
-import android.media.audiofx.AcousticEchoCanceler;
-import android.media.audiofx.AudioEffect;
-import android.media.audiofx.AudioEffect.Descriptor;
-import android.media.audiofx.NoiseSuppressor;
-import android.os.Build;
-import androidx.annotation.Nullable;
-import java.util.List;
-import java.util.UUID;
-import org.webrtc.Logging;
-
-// This class wraps control of three different platform effects. Supported
-// effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
-// Calling enable() will active all effects that are
-// supported by the device if the corresponding `shouldEnableXXX` member is set.
-public class WebRtcAudioEffects {
- private static final boolean DEBUG = false;
-
- private static final String TAG = "WebRtcAudioEffects";
-
- // UUIDs for Software Audio Effects that we want to avoid using.
- // The implementor field will be set to "The Android Open Source Project".
- private static final UUID AOSP_ACOUSTIC_ECHO_CANCELER =
- UUID.fromString("bb392ec0-8d4d-11e0-a896-0002a5d5c51b");
- private static final UUID AOSP_NOISE_SUPPRESSOR =
- UUID.fromString("c06c8400-8e06-11e0-9cb6-0002a5d5c51b");
-
- // Contains the available effect descriptors returned from the
- // AudioEffect.getEffects() call. This result is cached to avoid doing the
- // slow OS call multiple times.
- private static @Nullable Descriptor[] cachedEffects;
-
- // Contains the audio effect objects. Created in enable() and destroyed
- // in release().
- private @Nullable AcousticEchoCanceler aec;
- private @Nullable NoiseSuppressor ns;
-
- // Affects the final state given to the setEnabled() method on each effect.
- // The default state is set to "disabled" but each effect can also be enabled
- // by calling setAEC() and setNS().
- // To enable an effect, both the shouldEnableXXX member and the static
- // canUseXXX() must be true.
- private boolean shouldEnableAec;
- private boolean shouldEnableNs;
-
- // Checks if the device implements Acoustic Echo Cancellation (AEC).
- // Returns true if the device implements AEC, false otherwise.
- public static boolean isAcousticEchoCancelerSupported() {
- // Note: we're using isAcousticEchoCancelerEffectAvailable() instead of
- // AcousticEchoCanceler.isAvailable() to avoid the expensive getEffects()
- // OS API call.
- return isAcousticEchoCancelerEffectAvailable();
- }
-
- // Checks if the device implements Noise Suppression (NS).
- // Returns true if the device implements NS, false otherwise.
- public static boolean isNoiseSuppressorSupported() {
- // Note: we're using isNoiseSuppressorEffectAvailable() instead of
- // NoiseSuppressor.isAvailable() to avoid the expensive getEffects()
- // OS API call.
- return isNoiseSuppressorEffectAvailable();
- }
-
- // Returns true if the device is blacklisted for HW AEC usage.
- public static boolean isAcousticEchoCancelerBlacklisted() {
- List<String> blackListedModels = WebRtcAudioUtils.getBlackListedModelsForAecUsage();
- boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
- if (isBlacklisted) {
- Logging.w(TAG, Build.MODEL + " is blacklisted for HW AEC usage!");
- }
- return isBlacklisted;
- }
-
- // Returns true if the device is blacklisted for HW NS usage.
- public static boolean isNoiseSuppressorBlacklisted() {
- List<String> blackListedModels = WebRtcAudioUtils.getBlackListedModelsForNsUsage();
- boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
- if (isBlacklisted) {
- Logging.w(TAG, Build.MODEL + " is blacklisted for HW NS usage!");
- }
- return isBlacklisted;
- }
-
- // Returns true if the platform AEC should be excluded based on its UUID.
- // AudioEffect.queryEffects() can throw IllegalStateException.
- private static boolean isAcousticEchoCancelerExcludedByUUID() {
- for (Descriptor d : getAvailableEffects()) {
- if (d.type.equals(AudioEffect.EFFECT_TYPE_AEC)
- && d.uuid.equals(AOSP_ACOUSTIC_ECHO_CANCELER)) {
- return true;
- }
- }
- return false;
- }
-
- // Returns true if the platform NS should be excluded based on its UUID.
- // AudioEffect.queryEffects() can throw IllegalStateException.
- private static boolean isNoiseSuppressorExcludedByUUID() {
- for (Descriptor d : getAvailableEffects()) {
- if (d.type.equals(AudioEffect.EFFECT_TYPE_NS) && d.uuid.equals(AOSP_NOISE_SUPPRESSOR)) {
- return true;
- }
- }
- return false;
- }
-
- // Returns true if the device supports Acoustic Echo Cancellation (AEC).
- private static boolean isAcousticEchoCancelerEffectAvailable() {
- return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_AEC);
- }
-
- // Returns true if the device supports Noise Suppression (NS).
- private static boolean isNoiseSuppressorEffectAvailable() {
- return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_NS);
- }
-
- // Returns true if all conditions for supporting the HW AEC are fulfilled.
- // It will not be possible to enable the HW AEC if this method returns false.
- public static boolean canUseAcousticEchoCanceler() {
- boolean canUseAcousticEchoCanceler = isAcousticEchoCancelerSupported()
- && !WebRtcAudioUtils.useWebRtcBasedAcousticEchoCanceler()
- && !isAcousticEchoCancelerBlacklisted() && !isAcousticEchoCancelerExcludedByUUID();
- Logging.d(TAG, "canUseAcousticEchoCanceler: " + canUseAcousticEchoCanceler);
- return canUseAcousticEchoCanceler;
- }
-
- // Returns true if all conditions for supporting the HW NS are fulfilled.
- // It will not be possible to enable the HW NS if this method returns false.
- public static boolean canUseNoiseSuppressor() {
- boolean canUseNoiseSuppressor = isNoiseSuppressorSupported()
- && !WebRtcAudioUtils.useWebRtcBasedNoiseSuppressor() && !isNoiseSuppressorBlacklisted()
- && !isNoiseSuppressorExcludedByUUID();
- Logging.d(TAG, "canUseNoiseSuppressor: " + canUseNoiseSuppressor);
- return canUseNoiseSuppressor;
- }
-
- public static WebRtcAudioEffects create() {
- return new WebRtcAudioEffects();
- }
-
- private WebRtcAudioEffects() {
- Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
- }
-
- // Call this method to enable or disable the platform AEC. It modifies
- // `shouldEnableAec` which is used in enable() where the actual state
- // of the AEC effect is modified. Returns true if HW AEC is supported and
- // false otherwise.
- public boolean setAEC(boolean enable) {
- Logging.d(TAG, "setAEC(" + enable + ")");
- if (!canUseAcousticEchoCanceler()) {
- Logging.w(TAG, "Platform AEC is not supported");
- shouldEnableAec = false;
- return false;
- }
- if (aec != null && (enable != shouldEnableAec)) {
- Logging.e(TAG, "Platform AEC state can't be modified while recording");
- return false;
- }
- shouldEnableAec = enable;
- return true;
- }
-
- // Call this method to enable or disable the platform NS. It modifies
- // `shouldEnableNs` which is used in enable() where the actual state
- // of the NS effect is modified. Returns true if HW NS is supported and
- // false otherwise.
- public boolean setNS(boolean enable) {
- Logging.d(TAG, "setNS(" + enable + ")");
- if (!canUseNoiseSuppressor()) {
- Logging.w(TAG, "Platform NS is not supported");
- shouldEnableNs = false;
- return false;
- }
- if (ns != null && (enable != shouldEnableNs)) {
- Logging.e(TAG, "Platform NS state can't be modified while recording");
- return false;
- }
- shouldEnableNs = enable;
- return true;
- }
-
- public void enable(int audioSession) {
- Logging.d(TAG, "enable(audioSession=" + audioSession + ")");
- assertTrue(aec == null);
- assertTrue(ns == null);
-
- if (DEBUG) {
- // Add logging of supported effects but filter out "VoIP effects", i.e.,
- // AEC, AEC and NS. Avoid calling AudioEffect.queryEffects() unless the
- // DEBUG flag is set since we have seen crashes in this API.
- for (Descriptor d : AudioEffect.queryEffects()) {
- if (effectTypeIsVoIP(d.type)) {
- Logging.d(TAG, "name: " + d.name + ", "
- + "mode: " + d.connectMode + ", "
- + "implementor: " + d.implementor + ", "
- + "UUID: " + d.uuid);
- }
- }
- }
-
- if (isAcousticEchoCancelerSupported()) {
- // Create an AcousticEchoCanceler and attach it to the AudioRecord on
- // the specified audio session.
- aec = AcousticEchoCanceler.create(audioSession);
- if (aec != null) {
- boolean enabled = aec.getEnabled();
- boolean enable = shouldEnableAec && canUseAcousticEchoCanceler();
- if (aec.setEnabled(enable) != AudioEffect.SUCCESS) {
- Logging.e(TAG, "Failed to set the AcousticEchoCanceler state");
- }
- Logging.d(TAG, "AcousticEchoCanceler: was " + (enabled ? "enabled" : "disabled")
- + ", enable: " + enable + ", is now: "
- + (aec.getEnabled() ? "enabled" : "disabled"));
- } else {
- Logging.e(TAG, "Failed to create the AcousticEchoCanceler instance");
- }
- }
-
- if (isNoiseSuppressorSupported()) {
- // Create an NoiseSuppressor and attach it to the AudioRecord on the
- // specified audio session.
- ns = NoiseSuppressor.create(audioSession);
- if (ns != null) {
- boolean enabled = ns.getEnabled();
- boolean enable = shouldEnableNs && canUseNoiseSuppressor();
- if (ns.setEnabled(enable) != AudioEffect.SUCCESS) {
- Logging.e(TAG, "Failed to set the NoiseSuppressor state");
- }
- Logging.d(TAG, "NoiseSuppressor: was " + (enabled ? "enabled" : "disabled") + ", enable: "
- + enable + ", is now: " + (ns.getEnabled() ? "enabled" : "disabled"));
- } else {
- Logging.e(TAG, "Failed to create the NoiseSuppressor instance");
- }
- }
- }
-
- // Releases all native audio effect resources. It is a good practice to
- // release the effect engine when not in use as control can be returned
- // to other applications or the native resources released.
- public void release() {
- Logging.d(TAG, "release");
- if (aec != null) {
- aec.release();
- aec = null;
- }
- if (ns != null) {
- ns.release();
- ns = null;
- }
- }
-
- // Returns true for effect types in `type` that are of "VoIP" types:
- // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
- // Noise Suppressor (NS). Note that, an extra check for support is needed
- // in each comparison since some devices includes effects in the
- // AudioEffect.Descriptor array that are actually not available on the device.
- // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
- // AutomaticGainControl.isAvailable() returns false.
- private boolean effectTypeIsVoIP(UUID type) {
- return (AudioEffect.EFFECT_TYPE_AEC.equals(type) && isAcousticEchoCancelerSupported())
- || (AudioEffect.EFFECT_TYPE_NS.equals(type) && isNoiseSuppressorSupported());
- }
-
- // 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");
- }
- }
-
- // Returns the cached copy of the audio effects array, if available, or
- // queries the operating system for the list of effects.
- private static @Nullable Descriptor[] getAvailableEffects() {
- if (cachedEffects != null) {
- return cachedEffects;
- }
- // The caching is best effort only - if this method is called from several
- // threads in parallel, they may end up doing the underlying OS call
- // multiple times. It's normally only called on one thread so there's no
- // real need to optimize for the multiple threads case.
- cachedEffects = AudioEffect.queryEffects();
- return cachedEffects;
- }
-
- // Returns true if an effect of the specified type is available. Functionally
- // equivalent to (NoiseSuppressor`AutomaticGainControl`...).isAvailable(), but
- // faster as it avoids the expensive OS call to enumerate effects.
- private static boolean isEffectTypeAvailable(UUID effectType) {
- Descriptor[] effects = getAvailableEffects();
- if (effects == null) {
- return false;
- }
- for (Descriptor d : effects) {
- if (d.type.equals(effectType)) {
- return true;
- }
- }
- return false;
- }
-}
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
deleted file mode 100644
index 43c416f..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
+++ /dev/null
@@ -1,371 +0,0 @@
-/*
- * 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.voiceengine;
-
-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 androidx.annotation.Nullable;
-import java.util.Timer;
-import java.util.TimerTask;
-import org.webrtc.ContextUtils;
-import org.webrtc.Logging;
-
-// WebRtcAudioManager handles tasks that uses android.media.AudioManager.
-// At construction, storeAudioParameters() is called and it retrieves
-// fundamental audio parameters like native sample rate and number of channels.
-// The result is then provided to the caller by nativeCacheAudioParameters().
-// It is also possible to call init() to set up the audio environment for best
-// possible "VoIP performance". All settings done in init() are reverted by
-// dispose(). This class can also be used without calling init() if the user
-// prefers to set up the audio environment separately. However, it is
-// recommended to always use AudioManager.MODE_IN_COMMUNICATION.
-public class WebRtcAudioManager {
- private static final boolean DEBUG = false;
-
- 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;
- private static boolean useStereoInput;
-
- private static boolean blacklistDeviceForOpenSLESUsage;
- private static boolean blacklistDeviceForOpenSLESUsageIsOverridden;
-
- // Call this method to override the default list of blacklisted devices
- // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
- // Allows an app to take control over which devices to exclude from using
- // the OpenSL ES audio output path
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setBlacklistDeviceForOpenSLESUsage(boolean enable) {
- blacklistDeviceForOpenSLESUsageIsOverridden = true;
- blacklistDeviceForOpenSLESUsage = enable;
- }
-
- // Call these methods to override the default mono audio modes for the specified direction(s)
- // (input and/or output).
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setStereoOutput(boolean enable) {
- Logging.w(TAG, "Overriding default output behavior: setStereoOutput(" + enable + ')');
- useStereoOutput = enable;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setStereoInput(boolean enable) {
- Logging.w(TAG, "Overriding default input behavior: setStereoInput(" + enable + ')');
- useStereoInput = enable;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean getStereoOutput() {
- return useStereoOutput;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean getStereoInput() {
- return useStereoInput;
- }
-
- // 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;
-
- // Private utility class that periodically checks and logs the volume level
- // of the audio stream that is currently controlled by the volume control.
- // A timer triggers logs once every 30 seconds and the timer's associated
- // thread is named "WebRtcVolumeLevelLoggerThread".
- private static class VolumeLogger {
- private static final String THREAD_NAME = "WebRtcVolumeLevelLoggerThread";
- private static final int TIMER_PERIOD_IN_SECONDS = 30;
-
- private final AudioManager audioManager;
- private @Nullable Timer timer;
-
- public VolumeLogger(AudioManager audioManager) {
- this.audioManager = audioManager;
- }
-
- public void start() {
- timer = new Timer(THREAD_NAME);
- timer.schedule(new LogVolumeTask(audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
- audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
- 0, TIMER_PERIOD_IN_SECONDS * 1000);
- }
-
- private class LogVolumeTask extends TimerTask {
- private final int maxRingVolume;
- private final int maxVoiceCallVolume;
-
- LogVolumeTask(int maxRingVolume, int maxVoiceCallVolume) {
- this.maxRingVolume = maxRingVolume;
- this.maxVoiceCallVolume = maxVoiceCallVolume;
- }
-
- @Override
- public void run() {
- final int mode = audioManager.getMode();
- if (mode == AudioManager.MODE_RINGTONE) {
- Logging.d(TAG, "STREAM_RING stream volume: "
- + audioManager.getStreamVolume(AudioManager.STREAM_RING) + " (max="
- + maxRingVolume + ")");
- } else if (mode == AudioManager.MODE_IN_COMMUNICATION) {
- Logging.d(TAG, "VOICE_CALL stream volume: "
- + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL) + " (max="
- + maxVoiceCallVolume + ")");
- }
- }
- }
-
- private void stop() {
- if (timer != null) {
- timer.cancel();
- timer = null;
- }
- }
- }
-
- private final long nativeAudioManager;
- private final AudioManager audioManager;
-
- private boolean initialized;
- 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 VolumeLogger volumeLogger;
-
- WebRtcAudioManager(long nativeAudioManager) {
- Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
- this.nativeAudioManager = nativeAudioManager;
- audioManager =
- (AudioManager) ContextUtils.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
- if (DEBUG) {
- WebRtcAudioUtils.logDeviceInfo(TAG);
- }
- volumeLogger = new VolumeLogger(audioManager);
- storeAudioParameters();
- nativeCacheAudioParameters(sampleRate, outputChannels, inputChannels, hardwareAEC, hardwareAGC,
- hardwareNS, lowLatencyOutput, lowLatencyInput, proAudio, aAudio, outputBufferSize,
- inputBufferSize, nativeAudioManager);
- WebRtcAudioUtils.logAudioState(TAG);
- }
-
- private boolean init() {
- Logging.d(TAG, "init" + WebRtcAudioUtils.getThreadInfo());
- if (initialized) {
- return true;
- }
- Logging.d(TAG, "audio mode is: "
- + WebRtcAudioUtils.modeToString(audioManager.getMode()));
- initialized = true;
- volumeLogger.start();
- return true;
- }
-
- private void dispose() {
- Logging.d(TAG, "dispose" + WebRtcAudioUtils.getThreadInfo());
- if (!initialized) {
- return;
- }
- volumeLogger.stop();
- }
-
- private boolean isCommunicationModeEnabled() {
- return (audioManager.getMode() == AudioManager.MODE_IN_COMMUNICATION);
- }
-
- private boolean isDeviceBlacklistedForOpenSLESUsage() {
- boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
- ? blacklistDeviceForOpenSLESUsage
- : WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
- if (blacklisted) {
- Logging.d(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
- }
- 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);
- }
-
- // 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() {
- // 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();
- }
-
- // Returns true if the device has professional audio level of functionality
- // and therefore supports the lowest possible round-trip latency.
- private boolean isProAudioSupported() {
- return Build.VERSION.SDK_INT >= 23
- && 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 && Build.VERSION.SDK_INT >= 27;
- }
-
- // Returns the native output sample rate for this device's output stream.
- private int getNativeOutputSampleRate() {
- // 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;
- }
- // Default can be overriden by WebRtcAudioUtils.setDefaultSampleRateHz().
- // If so, use that value and return here.
- if (WebRtcAudioUtils.isDefaultSampleRateOverridden()) {
- Logging.d(TAG, "Default sample rate is overriden to "
- + WebRtcAudioUtils.getDefaultSampleRateHz() + " Hz");
- return WebRtcAudioUtils.getDefaultSampleRateHz();
- }
- // No overrides available. Deliver best possible estimate based on default
- // Android AudioManager APIs.
- final int sampleRateHz = getSampleRateForApiLevel();
- Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
- return sampleRateHz;
- }
-
- private int getSampleRateForApiLevel() {
- String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
- return (sampleRateString == null) ? WebRtcAudioUtils.getDefaultSampleRateHz()
- : Integer.parseInt(sampleRateString);
- }
-
- // Returns the native output buffer size for low-latency output streams.
- private int getLowLatencyOutputFramesPerBuffer() {
- assertTrue(isLowLatencyOutputSupported());
- String framesPerBuffer =
- audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
- return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
- }
-
- // Returns true if the device supports an audio effect (AEC or NS).
- // Four conditions must be fulfilled if functions are to return true:
- // 1) the platform must support the built-in (HW) effect,
- // 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() {
- return WebRtcAudioEffects.canUseAcousticEchoCanceler();
- }
- private static boolean isNoiseSuppressorSupported() {
- return WebRtcAudioEffects.canUseNoiseSuppressor();
- }
-
- // 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 native input buffer size for input streams.
- private int getLowLatencyInputFramesPerBuffer() {
- assertTrue(isLowLatencyInputSupported());
- return getLowLatencyOutputFramesPerBuffer();
- }
-
- // 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;
- }
-
- // 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");
- }
- }
-
- private native void nativeCacheAudioParameters(int sampleRate, int outputChannels,
- int inputChannels, boolean hardwareAEC, boolean hardwareAGC, boolean hardwareNS,
- boolean lowLatencyOutput, boolean lowLatencyInput, boolean proAudio, boolean aAudio,
- int outputBufferSize, int inputBufferSize, long nativeAudioManager);
-}
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
deleted file mode 100644
index 8eab01c..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
+++ /dev/null
@@ -1,409 +0,0 @@
-/*
- * 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.voiceengine;
-
-import android.media.AudioFormat;
-import android.media.AudioRecord;
-import android.media.MediaRecorder.AudioSource;
-import android.os.Build;
-import android.os.Process;
-import androidx.annotation.Nullable;
-import java.lang.System;
-import java.nio.ByteBuffer;
-import java.util.Arrays;
-import java.util.concurrent.TimeUnit;
-import org.webrtc.Logging;
-import org.webrtc.ThreadUtils;
-
-public class WebRtcAudioRecord {
- private static final boolean DEBUG = false;
-
- private static final String TAG = "WebRtcAudioRecord";
-
- // 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;
-
- // Requested size of each recorded buffer provided to the client.
- private static final int CALLBACK_BUFFER_SIZE_MS = 10;
-
- // Average number of callbacks per second.
- private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
-
- // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
- // buffer size). The extra space is allocated to guard against glitches under
- // high load.
- private static final int BUFFER_SIZE_FACTOR = 2;
-
- // The AudioRecordJavaThread is allowed to wait for successful call to join()
- // but the wait times out afther this amount of time.
- private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;
-
- private static final int DEFAULT_AUDIO_SOURCE = getDefaultAudioSource();
- private static int audioSource = DEFAULT_AUDIO_SOURCE;
-
- private final long nativeAudioRecord;
-
- private @Nullable WebRtcAudioEffects effects;
-
- private ByteBuffer byteBuffer;
-
- private @Nullable AudioRecord audioRecord;
- private @Nullable AudioRecordThread audioThread;
-
- private static volatile boolean microphoneMute;
- private byte[] emptyBytes;
-
- // Audio recording error handler functions.
- public enum AudioRecordStartErrorCode {
- AUDIO_RECORD_START_EXCEPTION,
- AUDIO_RECORD_START_STATE_MISMATCH,
- }
-
- public static interface WebRtcAudioRecordErrorCallback {
- void onWebRtcAudioRecordInitError(String errorMessage);
- void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
- void onWebRtcAudioRecordError(String errorMessage);
- }
-
- private static @Nullable WebRtcAudioRecordErrorCallback errorCallback;
-
- public static void setErrorCallback(WebRtcAudioRecordErrorCallback errorCallback) {
- Logging.d(TAG, "Set error callback");
- WebRtcAudioRecord.errorCallback = errorCallback;
- }
-
- /**
- * Contains audio sample information. Object is passed using {@link
- * WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback}
- */
- 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;
-
- private AudioSamples(AudioRecord audioRecord, byte[] data) {
- this.audioFormat = audioRecord.getAudioFormat();
- this.channelCount = audioRecord.getChannelCount();
- this.sampleRate = audioRecord.getSampleRate();
- 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 WebRtcAudioRecordSamplesReadyCallback {
- void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
- }
-
- private static @Nullable WebRtcAudioRecordSamplesReadyCallback audioSamplesReadyCallback;
-
- public static void setOnAudioSamplesReady(WebRtcAudioRecordSamplesReadyCallback callback) {
- audioSamplesReadyCallback = callback;
- }
-
- /**
- * Audio thread which keeps calling ByteBuffer.read() waiting for audio
- * to be recorded. Feeds recorded data to the native counterpart as a
- * periodic sequence of callbacks using DataIsRecorded().
- * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
- */
- private class AudioRecordThread extends Thread {
- private volatile boolean keepAlive = true;
-
- public AudioRecordThread(String name) {
- super(name);
- }
-
- // TODO(titovartem) make correct fix during webrtc:9175
- @SuppressWarnings("ByteBufferBackingArray")
- @Override
- public void run() {
- Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
- Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
- assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
-
- long lastTime = System.nanoTime();
- while (keepAlive) {
- int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
- if (bytesRead == byteBuffer.capacity()) {
- if (microphoneMute) {
- byteBuffer.clear();
- byteBuffer.put(emptyBytes);
- }
- // It's possible we've been shut down during the read, and stopRecording() tried and
- // failed to join this thread. To be a bit safer, try to avoid calling any native methods
- // in case they've been unregistered after stopRecording() returned.
- if (keepAlive) {
- nativeDataIsRecorded(bytesRead, nativeAudioRecord);
- }
- if (audioSamplesReadyCallback != null) {
- // Copy the entire byte buffer array. Assume that the start of the byteBuffer is
- // at index 0.
- byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
- audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
- new AudioSamples(audioRecord, data));
- }
- } else {
- String errorMessage = "AudioRecord.read failed: " + bytesRead;
- Logging.e(TAG, errorMessage);
- if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
- keepAlive = false;
- reportWebRtcAudioRecordError(errorMessage);
- }
- }
- if (DEBUG) {
- long nowTime = System.nanoTime();
- long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
- lastTime = nowTime;
- Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
- }
- }
-
- try {
- if (audioRecord != null) {
- audioRecord.stop();
- }
- } catch (IllegalStateException e) {
- Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
- }
- }
-
- // Stops the inner thread loop and also calls AudioRecord.stop().
- // Does not block the calling thread.
- public void stopThread() {
- Logging.d(TAG, "stopThread");
- keepAlive = false;
- }
- }
-
- WebRtcAudioRecord(long nativeAudioRecord) {
- Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
- this.nativeAudioRecord = nativeAudioRecord;
- if (DEBUG) {
- WebRtcAudioUtils.logDeviceInfo(TAG);
- }
- effects = WebRtcAudioEffects.create();
- }
-
- private boolean enableBuiltInAEC(boolean enable) {
- Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
- if (effects == null) {
- Logging.e(TAG, "Built-in AEC is not supported on this platform");
- return false;
- }
- return effects.setAEC(enable);
- }
-
- private boolean enableBuiltInNS(boolean enable) {
- Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
- if (effects == null) {
- Logging.e(TAG, "Built-in NS is not supported on this platform");
- return false;
- }
- return effects.setNS(enable);
- }
-
- private int initRecording(int sampleRate, int channels) {
- Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
- if (audioRecord != null) {
- reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
- return -1;
- }
- final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
- final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
- byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
- Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
- emptyBytes = new byte[byteBuffer.capacity()];
- // Rather than passing the ByteBuffer with every callback (requiring
- // the potentially expensive GetDirectBufferAddress) we simply have the
- // the native class cache the address to the memory once.
- nativeCacheDirectBufferAddress(byteBuffer, nativeAudioRecord);
-
- // Get the minimum buffer size required for the successful creation of
- // an AudioRecord object, in byte units.
- // Note that this size doesn't guarantee a smooth recording under load.
- final int channelConfig = channelCountToConfiguration(channels);
- int minBufferSize =
- AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
- if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
- reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
- return -1;
- }
- Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
-
- // Use a larger buffer size than the minimum required when creating the
- // AudioRecord instance to ensure smooth recording under load. It has been
- // verified that it does not increase the actual recording latency.
- int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
- Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
- try {
- audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
- AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
- } catch (IllegalArgumentException e) {
- reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
- releaseAudioResources();
- return -1;
- }
- if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
- reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
- releaseAudioResources();
- return -1;
- }
- if (effects != null) {
- effects.enable(audioRecord.getAudioSessionId());
- }
- logMainParameters();
- logMainParametersExtended();
- return framesPerBuffer;
- }
-
- private boolean startRecording() {
- Logging.d(TAG, "startRecording");
- assertTrue(audioRecord != null);
- assertTrue(audioThread == null);
- try {
- audioRecord.startRecording();
- } catch (IllegalStateException e) {
- reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
- "AudioRecord.startRecording failed: " + e.getMessage());
- return false;
- }
- if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
- reportWebRtcAudioRecordStartError(
- AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
- "AudioRecord.startRecording failed - incorrect state :"
- + audioRecord.getRecordingState());
- return false;
- }
- audioThread = new AudioRecordThread("AudioRecordJavaThread");
- audioThread.start();
- return true;
- }
-
- private boolean stopRecording() {
- Logging.d(TAG, "stopRecording");
- assertTrue(audioThread != null);
- audioThread.stopThread();
- if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
- Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
- WebRtcAudioUtils.logAudioState(TAG);
- }
- audioThread = null;
- if (effects != null) {
- effects.release();
- }
- releaseAudioResources();
- return true;
- }
-
- private void logMainParameters() {
- Logging.d(TAG, "AudioRecord: "
- + "session ID: " + audioRecord.getAudioSessionId() + ", "
- + "channels: " + audioRecord.getChannelCount() + ", "
- + "sample rate: " + audioRecord.getSampleRate());
- }
-
- private void logMainParametersExtended() {
- if (Build.VERSION.SDK_INT >= 23) {
- Logging.d(TAG, "AudioRecord: "
- // The frame count of the native AudioRecord buffer.
- + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
- }
- }
-
- // 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");
- }
- }
-
- private int channelCountToConfiguration(int channels) {
- return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
- }
-
- private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);
-
- private native void nativeDataIsRecorded(int bytes, long nativeAudioRecord);
-
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setAudioSource(int source) {
- Logging.w(TAG, "Audio source is changed from: " + audioSource
- + " to " + source);
- audioSource = source;
- }
-
- private static int getDefaultAudioSource() {
- return AudioSource.VOICE_COMMUNICATION;
- }
-
- // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
- // the microphone is muted.
- public static void setMicrophoneMute(boolean mute) {
- Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
- microphoneMute = mute;
- }
-
- // Releases the native AudioRecord resources.
- private void releaseAudioResources() {
- Logging.d(TAG, "releaseAudioResources");
- if (audioRecord != null) {
- audioRecord.release();
- audioRecord = null;
- }
- }
-
- private void reportWebRtcAudioRecordInitError(String errorMessage) {
- Logging.e(TAG, "Init recording error: " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioRecordInitError(errorMessage);
- }
- }
-
- private void reportWebRtcAudioRecordStartError(
- AudioRecordStartErrorCode errorCode, String errorMessage) {
- Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
- }
- }
-
- private void reportWebRtcAudioRecordError(String errorMessage) {
- Logging.e(TAG, "Run-time recording error: " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioRecordError(errorMessage);
- }
- }
-}
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
deleted file mode 100644
index 3e1875c..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
+++ /dev/null
@@ -1,494 +0,0 @@
-/*
- * 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.voiceengine;
-
-import android.content.Context;
-import android.media.AudioAttributes;
-import android.media.AudioFormat;
-import android.media.AudioManager;
-import android.media.AudioTrack;
-import android.os.Build;
-import android.os.Process;
-import androidx.annotation.Nullable;
-import java.lang.Thread;
-import java.nio.ByteBuffer;
-import org.webrtc.ContextUtils;
-import org.webrtc.Logging;
-import org.webrtc.ThreadUtils;
-
-public class WebRtcAudioTrack {
- private static final boolean DEBUG = false;
-
- private static final String TAG = "WebRtcAudioTrack";
-
- // 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;
-
- // Requested size of each recorded buffer provided to the client.
- private static final int CALLBACK_BUFFER_SIZE_MS = 10;
-
- // Average number of callbacks per second.
- private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
-
- // The AudioTrackThread is allowed to wait for successful call to join()
- // but the wait times out afther this amount of time.
- private static final long AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS = 2000;
-
- // By default, WebRTC creates audio tracks with a usage attribute
- // corresponding to voice communications, such as telephony or VoIP.
- private static final int DEFAULT_USAGE = AudioAttributes.USAGE_VOICE_COMMUNICATION;
- private static int usageAttribute = DEFAULT_USAGE;
-
- // This method overrides the default usage attribute and allows the user
- // to set it to something else than AudioAttributes.USAGE_VOICE_COMMUNICATION.
- // NOTE: calling this method will most likely break existing VoIP tuning.
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setAudioTrackUsageAttribute(int usage) {
- Logging.w(TAG, "Default usage attribute is changed from: "
- + DEFAULT_USAGE + " to " + usage);
- usageAttribute = usage;
- }
-
- private final long nativeAudioTrack;
- private final AudioManager audioManager;
- private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
-
- private ByteBuffer byteBuffer;
-
- private @Nullable AudioTrack audioTrack;
- private @Nullable AudioTrackThread audioThread;
-
- // Samples to be played are replaced by zeros if `speakerMute` is set to true.
- // Can be used to ensure that the speaker is fully muted.
- private static volatile boolean speakerMute;
- private byte[] emptyBytes;
-
- // Audio playout/track error handler functions.
- public enum AudioTrackStartErrorCode {
- AUDIO_TRACK_START_EXCEPTION,
- AUDIO_TRACK_START_STATE_MISMATCH,
- }
-
- @Deprecated
- public static interface WebRtcAudioTrackErrorCallback {
- void onWebRtcAudioTrackInitError(String errorMessage);
- void onWebRtcAudioTrackStartError(String errorMessage);
- void onWebRtcAudioTrackError(String errorMessage);
- }
-
- // TODO(henrika): upgrade all clients to use this new interface instead.
- public static interface ErrorCallback {
- void onWebRtcAudioTrackInitError(String errorMessage);
- void onWebRtcAudioTrackStartError(AudioTrackStartErrorCode errorCode, String errorMessage);
- void onWebRtcAudioTrackError(String errorMessage);
- }
-
- private static @Nullable WebRtcAudioTrackErrorCallback errorCallbackOld;
- private static @Nullable ErrorCallback errorCallback;
-
- @Deprecated
- public static void setErrorCallback(WebRtcAudioTrackErrorCallback errorCallback) {
- Logging.d(TAG, "Set error callback (deprecated");
- WebRtcAudioTrack.errorCallbackOld = errorCallback;
- }
-
- public static void setErrorCallback(ErrorCallback errorCallback) {
- Logging.d(TAG, "Set extended error callback");
- WebRtcAudioTrack.errorCallback = errorCallback;
- }
-
- /**
- * Audio thread which keeps calling AudioTrack.write() to stream audio.
- * Data is periodically acquired from the native WebRTC layer using the
- * nativeGetPlayoutData callback function.
- * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
- */
- private class AudioTrackThread extends Thread {
- private volatile boolean keepAlive = true;
-
- public AudioTrackThread(String name) {
- super(name);
- }
-
- @Override
- public void run() {
- Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
- Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
- assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
-
- // Fixed size in bytes of each 10ms block of audio data that we ask for
- // using callbacks to the native WebRTC client.
- final int sizeInBytes = byteBuffer.capacity();
-
- while (keepAlive) {
- // Get 10ms of PCM data from the native WebRTC client. Audio data is
- // written into the common ByteBuffer using the address that was
- // cached at construction.
- nativeGetPlayoutData(sizeInBytes, nativeAudioTrack);
- // Write data until all data has been written to the audio sink.
- // Upon return, the buffer position will have been advanced to reflect
- // the amount of data that was successfully written to the AudioTrack.
- assertTrue(sizeInBytes <= byteBuffer.remaining());
- if (speakerMute) {
- byteBuffer.clear();
- byteBuffer.put(emptyBytes);
- byteBuffer.position(0);
- }
- int bytesWritten = audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
- if (bytesWritten != sizeInBytes) {
- Logging.e(TAG, "AudioTrack.write played invalid number of bytes: " + bytesWritten);
- // If a write() returns a negative value, an error has occurred.
- // Stop playing and report an error in this case.
- if (bytesWritten < 0) {
- keepAlive = false;
- reportWebRtcAudioTrackError("AudioTrack.write failed: " + bytesWritten);
- }
- }
- // The byte buffer must be rewinded since byteBuffer.position() is
- // increased at each call to AudioTrack.write(). If we don't do this,
- // next call to AudioTrack.write() will fail.
- byteBuffer.rewind();
-
- // TODO(henrika): it is possible to create a delay estimate here by
- // counting number of written frames and subtracting the result from
- // audioTrack.getPlaybackHeadPosition().
- }
-
- // Stops playing the audio data. Since the instance was created in
- // MODE_STREAM mode, audio will stop playing after the last buffer that
- // was written has been played.
- if (audioTrack != null) {
- Logging.d(TAG, "Calling AudioTrack.stop...");
- try {
- audioTrack.stop();
- Logging.d(TAG, "AudioTrack.stop is done.");
- } catch (IllegalStateException e) {
- Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
- }
- }
- }
-
- // Stops the inner thread loop which results in calling AudioTrack.stop().
- // Does not block the calling thread.
- public void stopThread() {
- Logging.d(TAG, "stopThread");
- keepAlive = false;
- }
- }
-
- WebRtcAudioTrack(long nativeAudioTrack) {
- threadChecker.checkIsOnValidThread();
- Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
- this.nativeAudioTrack = nativeAudioTrack;
- audioManager =
- (AudioManager) ContextUtils.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
- if (DEBUG) {
- WebRtcAudioUtils.logDeviceInfo(TAG);
- }
- }
-
- private int initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
- threadChecker.checkIsOnValidThread();
- Logging.d(TAG,
- "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
- + ", bufferSizeFactor=" + bufferSizeFactor + ")");
- final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
- byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
- Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
- emptyBytes = new byte[byteBuffer.capacity()];
- // Rather than passing the ByteBuffer with every callback (requiring
- // the potentially expensive GetDirectBufferAddress) we simply have the
- // the native class cache the address to the memory once.
- nativeCacheDirectBufferAddress(byteBuffer, nativeAudioTrack);
-
- // Get the minimum buffer size required for the successful creation of an
- // AudioTrack object to be created in the MODE_STREAM mode.
- // Note that this size doesn't guarantee a smooth playback under load.
- final int channelConfig = channelCountToConfiguration(channels);
- final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig,
- AudioFormat.ENCODING_PCM_16BIT)
- * bufferSizeFactor);
- Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
- // For the streaming mode, data must be written to the audio sink in
- // chunks of size (given by byteBuffer.capacity()) less than or equal
- // to the total buffer size `minBufferSizeInBytes`. But, we have seen
- // reports of "getMinBufferSize(): error querying hardware". Hence, it
- // can happen that `minBufferSizeInBytes` contains an invalid value.
- if (minBufferSizeInBytes < byteBuffer.capacity()) {
- reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
- return -1;
- }
-
- // Ensure that prevision audio session was stopped correctly before trying
- // to create a new AudioTrack.
- if (audioTrack != null) {
- reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack.");
- return -1;
- }
- try {
- // Create an AudioTrack object and initialize its associated audio buffer.
- // The size of this buffer determines how long an AudioTrack can play
- // before running out of data.
- // As we are on API level 21 or higher, it is possible to use a special AudioTrack
- // constructor that uses AudioAttributes and AudioFormat as input. It allows us to
- // supersede the notion of stream types for defining the behavior of audio playback,
- // and to allow certain platforms or routing policies to use this information for more
- // refined volume or routing decisions.
- audioTrack = createAudioTrack(sampleRate, channelConfig, minBufferSizeInBytes);
- } catch (IllegalArgumentException e) {
- reportWebRtcAudioTrackInitError(e.getMessage());
- releaseAudioResources();
- return -1;
- }
-
- // It can happen that an AudioTrack is created but it was not successfully
- // initialized upon creation. Seems to be the case e.g. when the maximum
- // number of globally available audio tracks is exceeded.
- if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
- reportWebRtcAudioTrackInitError("Initialization of audio track failed.");
- releaseAudioResources();
- return -1;
- }
- logMainParameters();
- logMainParametersExtended();
- return minBufferSizeInBytes;
- }
-
- private boolean startPlayout() {
- threadChecker.checkIsOnValidThread();
- Logging.d(TAG, "startPlayout");
- assertTrue(audioTrack != null);
- assertTrue(audioThread == null);
-
- // Starts playing an audio track.
- try {
- audioTrack.play();
- } catch (IllegalStateException e) {
- reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION,
- "AudioTrack.play failed: " + e.getMessage());
- releaseAudioResources();
- return false;
- }
- if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
- reportWebRtcAudioTrackStartError(
- AudioTrackStartErrorCode.AUDIO_TRACK_START_STATE_MISMATCH,
- "AudioTrack.play failed - incorrect state :"
- + audioTrack.getPlayState());
- releaseAudioResources();
- return false;
- }
-
- // Create and start new high-priority thread which calls AudioTrack.write()
- // and where we also call the native nativeGetPlayoutData() callback to
- // request decoded audio from WebRTC.
- audioThread = new AudioTrackThread("AudioTrackJavaThread");
- audioThread.start();
- return true;
- }
-
- private boolean stopPlayout() {
- threadChecker.checkIsOnValidThread();
- Logging.d(TAG, "stopPlayout");
- assertTrue(audioThread != null);
- logUnderrunCount();
- audioThread.stopThread();
-
- Logging.d(TAG, "Stopping the AudioTrackThread...");
- audioThread.interrupt();
- if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) {
- Logging.e(TAG, "Join of AudioTrackThread timed out.");
- WebRtcAudioUtils.logAudioState(TAG);
- }
- Logging.d(TAG, "AudioTrackThread has now been stopped.");
- audioThread = null;
- releaseAudioResources();
- return true;
- }
-
- // 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 (audioManager.isVolumeFixed()) {
- Logging.e(TAG, "The device implements a fixed volume policy.");
- return false;
- }
- audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
- return true;
- }
-
- /** Get current volume level for a phone call audio stream. */
- private int getStreamVolume() {
- threadChecker.checkIsOnValidThread();
- Logging.d(TAG, "getStreamVolume");
- assertTrue(audioManager != null);
- return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
- }
-
- private void logMainParameters() {
- Logging.d(TAG, "AudioTrack: "
- + "session ID: " + audioTrack.getAudioSessionId() + ", "
- + "channels: " + audioTrack.getChannelCount() + ", "
- + "sample rate: " + audioTrack.getSampleRate() + ", "
- // Gain (>=1.0) expressed as linear multiplier on sample values.
- + "max gain: " + AudioTrack.getMaxVolume());
- }
-
- // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
- // It allows certain platforms or routing policies to use this information for more
- // refined volume or routing decisions.
- private static AudioTrack createAudioTrack(
- int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
- Logging.d(TAG, "createAudioTrack");
- // TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
- // performance when Android O is supported. Add some logging in the mean time.
- final int nativeOutputSampleRate =
- AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
- Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
- if (sampleRateInHz != nativeOutputSampleRate) {
- Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
- }
- if (usageAttribute != DEFAULT_USAGE) {
- Logging.w(TAG, "A non default usage attribute is used: " + usageAttribute);
- }
- // Create an audio track where the audio usage is for VoIP and the content type is speech.
- return new AudioTrack(
- new AudioAttributes.Builder()
- .setUsage(usageAttribute)
- .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
- .build(),
- new AudioFormat.Builder()
- .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
- .setSampleRate(sampleRateInHz)
- .setChannelMask(channelConfig)
- .build(),
- bufferSizeInBytes,
- AudioTrack.MODE_STREAM,
- AudioManager.AUDIO_SESSION_ID_GENERATE);
- }
-
- private void logBufferSizeInFrames() {
- if (Build.VERSION.SDK_INT >= 23) {
- Logging.d(TAG, "AudioTrack: "
- // The effective size of the AudioTrack buffer that the app writes to.
- + "buffer size in frames: " + audioTrack.getBufferSizeInFrames());
- }
- }
-
- private int getBufferSizeInFrames() {
- if (Build.VERSION.SDK_INT >= 23) {
- return audioTrack.getBufferSizeInFrames();
- }
- return -1;
- }
-
- private void logBufferCapacityInFrames() {
- if (Build.VERSION.SDK_INT >= 24) {
- Logging.d(TAG,
- "AudioTrack: "
- // Maximum size of the AudioTrack buffer in frames.
- + "buffer capacity in frames: " + audioTrack.getBufferCapacityInFrames());
- }
- }
-
- private void logMainParametersExtended() {
- logBufferSizeInFrames();
- logBufferCapacityInFrames();
- }
-
- // Prints the number of underrun occurrences in the application-level write
- // buffer since the AudioTrack was created. An underrun occurs if the app does
- // not write audio data quickly enough, causing the buffer to underflow and a
- // potential audio glitch.
- // TODO(henrika): keep track of this value in the field and possibly add new
- // UMA stat if needed.
- private void logUnderrunCount() {
- if (Build.VERSION.SDK_INT >= 24) {
- Logging.d(TAG, "underrun count: " + audioTrack.getUnderrunCount());
- }
- }
-
- // 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");
- }
- }
-
- private int channelCountToConfiguration(int channels) {
- return (channels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
- }
-
- private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);
-
- private native void nativeGetPlayoutData(int bytes, long nativeAudioRecord);
-
- // Sets all samples to be played out to zero if `mute` is true, i.e.,
- // ensures that the speaker is muted.
- public static void setSpeakerMute(boolean mute) {
- Logging.w(TAG, "setSpeakerMute(" + mute + ")");
- speakerMute = mute;
- }
-
- // Releases the native AudioTrack resources.
- private void releaseAudioResources() {
- Logging.d(TAG, "releaseAudioResources");
- if (audioTrack != null) {
- audioTrack.release();
- audioTrack = null;
- }
- }
-
- private void reportWebRtcAudioTrackInitError(String errorMessage) {
- Logging.e(TAG, "Init playout error: " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallbackOld != null) {
- errorCallbackOld.onWebRtcAudioTrackInitError(errorMessage);
- }
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioTrackInitError(errorMessage);
- }
- }
-
- private void reportWebRtcAudioTrackStartError(
- AudioTrackStartErrorCode errorCode, String errorMessage) {
- Logging.e(TAG, "Start playout error: " + errorCode + ". " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallbackOld != null) {
- errorCallbackOld.onWebRtcAudioTrackStartError(errorMessage);
- }
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioTrackStartError(errorCode, errorMessage);
- }
- }
-
- private void reportWebRtcAudioTrackError(String errorMessage) {
- Logging.e(TAG, "Run-time playback error: " + errorMessage);
- WebRtcAudioUtils.logAudioState(TAG);
- if (errorCallbackOld != null) {
- errorCallbackOld.onWebRtcAudioTrackError(errorMessage);
- }
- if (errorCallback != null) {
- errorCallback.onWebRtcAudioTrackError(errorMessage);
- }
- }
-}
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
deleted file mode 100644
index 0472114..0000000
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
+++ /dev/null
@@ -1,377 +0,0 @@
-/*
- * 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.voiceengine;
-
-import static android.media.AudioManager.MODE_IN_CALL;
-import static android.media.AudioManager.MODE_IN_COMMUNICATION;
-import static android.media.AudioManager.MODE_NORMAL;
-import static android.media.AudioManager.MODE_RINGTONE;
-
-import android.content.Context;
-import android.content.pm.PackageManager;
-import android.media.AudioDeviceInfo;
-import android.media.AudioManager;
-import android.os.Build;
-import java.lang.Thread;
-import java.util.Arrays;
-import java.util.List;
-import org.webrtc.ContextUtils;
-import org.webrtc.Logging;
-
-public final class WebRtcAudioUtils {
- private static final String TAG = "WebRtcAudioUtils";
-
- // List of devices where we have seen issues (e.g. bad audio quality) using
- // the low latency output mode in combination with OpenSL ES.
- // The device name is given by Build.MODEL.
- private static final String[] BLACKLISTED_OPEN_SL_ES_MODELS = new String[] {
- // It is recommended to maintain a list of blacklisted models outside
- // this package and instead call
- // WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(true)
- // from the client for devices where OpenSL ES shall be disabled.
- };
-
- // List of devices where it has been verified that the built-in effect
- // bad and where it makes sense to avoid using it and instead rely on the
- // native WebRTC version instead. The device name is given by Build.MODEL.
- private static final String[] BLACKLISTED_AEC_MODELS = new String[] {
- // It is recommended to maintain a list of blacklisted models outside
- // this package and instead call setWebRtcBasedAcousticEchoCanceler(true)
- // from the client for devices where the built-in AEC shall be disabled.
- };
- private static final String[] BLACKLISTED_NS_MODELS = new String[] {
- // It is recommended to maintain a list of blacklisted models outside
- // this package and instead call setWebRtcBasedNoiseSuppressor(true)
- // from the client for devices where the built-in NS shall be disabled.
- };
-
- // Use 16kHz as the default sample rate. A higher sample rate might prevent
- // us from supporting communication mode on some older (e.g. ICS) devices.
- private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
- private static int defaultSampleRateHz = DEFAULT_SAMPLE_RATE_HZ;
- // Set to true if setDefaultSampleRateHz() has been called.
- private static boolean isDefaultSampleRateOverridden;
-
- // By default, utilize hardware based audio effects for AEC and NS when
- // available.
- private static boolean useWebRtcBasedAcousticEchoCanceler;
- private static boolean useWebRtcBasedNoiseSuppressor;
-
- // Call these methods if any hardware based effect shall be replaced by a
- // software based version provided by the WebRTC stack instead.
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setWebRtcBasedAcousticEchoCanceler(boolean enable) {
- useWebRtcBasedAcousticEchoCanceler = enable;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setWebRtcBasedNoiseSuppressor(boolean enable) {
- useWebRtcBasedNoiseSuppressor = enable;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setWebRtcBasedAutomaticGainControl(boolean enable) {
- // TODO(henrika): deprecated; remove when no longer used by any client.
- Logging.w(TAG, "setWebRtcBasedAutomaticGainControl() is deprecated");
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean useWebRtcBasedAcousticEchoCanceler() {
- if (useWebRtcBasedAcousticEchoCanceler) {
- Logging.w(TAG, "Overriding default behavior; now using WebRTC AEC!");
- }
- return useWebRtcBasedAcousticEchoCanceler;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean useWebRtcBasedNoiseSuppressor() {
- if (useWebRtcBasedNoiseSuppressor) {
- Logging.w(TAG, "Overriding default behavior; now using WebRTC NS!");
- }
- return useWebRtcBasedNoiseSuppressor;
- }
-
- // TODO(henrika): deprecated; remove when no longer used by any client.
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean useWebRtcBasedAutomaticGainControl() {
- // Always return true here to avoid trying to use any built-in AGC.
- return true;
- }
-
- // Returns true if the device supports an audio effect (AEC or NS).
- // Four conditions must be fulfilled if functions are to return true:
- // 1) the platform must support the built-in (HW) effect,
- // 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).
- public static boolean isAcousticEchoCancelerSupported() {
- return WebRtcAudioEffects.canUseAcousticEchoCanceler();
- }
- 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
- // available Android APIs are known to return invalid results.
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setDefaultSampleRateHz(int sampleRateHz) {
- isDefaultSampleRateOverridden = true;
- defaultSampleRateHz = sampleRateHz;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized boolean isDefaultSampleRateOverridden() {
- return isDefaultSampleRateOverridden;
- }
-
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized int getDefaultSampleRateHz() {
- return defaultSampleRateHz;
- }
-
- public static List<String> getBlackListedModelsForAecUsage() {
- return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_AEC_MODELS);
- }
-
- public static List<String> getBlackListedModelsForNsUsage() {
- return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_NS_MODELS);
- }
-
- // Helper method for building a string of thread information.
- public static String getThreadInfo() {
- return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
- + "]";
- }
-
- // Returns true if we're running on emulator.
- public static boolean runningOnEmulator() {
- return Build.HARDWARE.equals("goldfish") && Build.BRAND.startsWith("generic_");
- }
-
- // Returns true if the device is blacklisted for OpenSL ES usage.
- public static boolean deviceIsBlacklistedForOpenSLESUsage() {
- List<String> blackListedModels = Arrays.asList(BLACKLISTED_OPEN_SL_ES_MODELS);
- return blackListedModels.contains(Build.MODEL);
- }
-
- // Information about the current build, taken from system properties.
- static void logDeviceInfo(String tag) {
- Logging.d(tag, "Android SDK: " + Build.VERSION.SDK_INT + ", "
- + "Release: " + Build.VERSION.RELEASE + ", "
- + "Brand: " + Build.BRAND + ", "
- + "Device: " + Build.DEVICE + ", "
- + "Id: " + Build.ID + ", "
- + "Hardware: " + Build.HARDWARE + ", "
- + "Manufacturer: " + Build.MANUFACTURER + ", "
- + "Model: " + Build.MODEL + ", "
- + "Product: " + Build.PRODUCT);
- }
-
- // Logs information about the current audio state. The idea is to call this
- // method when errors are detected to log under what conditions the error
- // occurred. Hopefully it will provide clues to what might be the root cause.
- static void logAudioState(String tag) {
- logDeviceInfo(tag);
- final Context context = ContextUtils.getApplicationContext();
- final AudioManager audioManager =
- (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
- logAudioStateBasic(tag, audioManager);
- logAudioStateVolume(tag, audioManager);
- logAudioDeviceInfo(tag, audioManager);
- }
-
- // Reports basic audio statistics.
- private static void logAudioStateBasic(String tag, AudioManager audioManager) {
- Logging.d(tag, "Audio State: "
- + "audio mode: " + modeToString(audioManager.getMode()) + ", "
- + "has mic: " + hasMicrophone() + ", "
- + "mic muted: " + audioManager.isMicrophoneMute() + ", "
- + "music active: " + audioManager.isMusicActive() + ", "
- + "speakerphone: " + audioManager.isSpeakerphoneOn() + ", "
- + "BT SCO: " + audioManager.isBluetoothScoOn());
- }
-
- // Adds volume information for all possible stream types.
- private static void logAudioStateVolume(String tag, AudioManager audioManager) {
- final int[] streams = {
- AudioManager.STREAM_VOICE_CALL,
- AudioManager.STREAM_MUSIC,
- AudioManager.STREAM_RING,
- AudioManager.STREAM_ALARM,
- AudioManager.STREAM_NOTIFICATION,
- AudioManager.STREAM_SYSTEM
- };
- Logging.d(tag, "Audio State: ");
- // Some devices may not have volume controls and might use a fixed volume.
- boolean fixedVolume = audioManager.isVolumeFixed();
- Logging.d(tag, " fixed volume=" + fixedVolume);
- if (!fixedVolume) {
- for (int stream : streams) {
- StringBuilder info = new StringBuilder();
- info.append(" " + streamTypeToString(stream) + ": ");
- info.append("volume=").append(audioManager.getStreamVolume(stream));
- info.append(", max=").append(audioManager.getStreamMaxVolume(stream));
- logIsStreamMute(tag, audioManager, stream, info);
- Logging.d(tag, info.toString());
- }
- }
- }
-
- private static void logIsStreamMute(
- String tag, AudioManager audioManager, int stream, StringBuilder info) {
- if (Build.VERSION.SDK_INT >= 23) {
- info.append(", muted=").append(audioManager.isStreamMute(stream));
- }
- }
-
- private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
- if (Build.VERSION.SDK_INT < 23) {
- return;
- }
- final AudioDeviceInfo[] devices =
- audioManager.getDevices(AudioManager.GET_DEVICES_ALL);
- if (devices.length == 0) {
- return;
- }
- Logging.d(tag, "Audio Devices: ");
- for (AudioDeviceInfo device : devices) {
- StringBuilder info = new StringBuilder();
- info.append(" ").append(deviceTypeToString(device.getType()));
- info.append(device.isSource() ? "(in): " : "(out): ");
- // An empty array indicates that the device supports arbitrary channel counts.
- if (device.getChannelCounts().length > 0) {
- info.append("channels=").append(Arrays.toString(device.getChannelCounts()));
- info.append(", ");
- }
- if (device.getEncodings().length > 0) {
- // Examples: ENCODING_PCM_16BIT = 2, ENCODING_PCM_FLOAT = 4.
- info.append("encodings=").append(Arrays.toString(device.getEncodings()));
- info.append(", ");
- }
- if (device.getSampleRates().length > 0) {
- info.append("sample rates=").append(Arrays.toString(device.getSampleRates()));
- info.append(", ");
- }
- info.append("id=").append(device.getId());
- Logging.d(tag, info.toString());
- }
- }
-
- // Converts media.AudioManager modes into local string representation.
- static String modeToString(int mode) {
- switch (mode) {
- case MODE_IN_CALL:
- return "MODE_IN_CALL";
- case MODE_IN_COMMUNICATION:
- return "MODE_IN_COMMUNICATION";
- case MODE_NORMAL:
- return "MODE_NORMAL";
- case MODE_RINGTONE:
- return "MODE_RINGTONE";
- default:
- return "MODE_INVALID";
- }
- }
-
- private static String streamTypeToString(int stream) {
- switch(stream) {
- case AudioManager.STREAM_VOICE_CALL:
- return "STREAM_VOICE_CALL";
- case AudioManager.STREAM_MUSIC:
- return "STREAM_MUSIC";
- case AudioManager.STREAM_RING:
- return "STREAM_RING";
- case AudioManager.STREAM_ALARM:
- return "STREAM_ALARM";
- case AudioManager.STREAM_NOTIFICATION:
- return "STREAM_NOTIFICATION";
- case AudioManager.STREAM_SYSTEM:
- return "STREAM_SYSTEM";
- default:
- return "STREAM_INVALID";
- }
- }
-
- // Converts AudioDeviceInfo types to local string representation.
- private static String deviceTypeToString(int type) {
- switch (type) {
- case AudioDeviceInfo.TYPE_UNKNOWN:
- return "TYPE_UNKNOWN";
- case AudioDeviceInfo.TYPE_BUILTIN_EARPIECE:
- return "TYPE_BUILTIN_EARPIECE";
- case AudioDeviceInfo.TYPE_BUILTIN_SPEAKER:
- return "TYPE_BUILTIN_SPEAKER";
- case AudioDeviceInfo.TYPE_WIRED_HEADSET:
- return "TYPE_WIRED_HEADSET";
- case AudioDeviceInfo.TYPE_WIRED_HEADPHONES:
- return "TYPE_WIRED_HEADPHONES";
- case AudioDeviceInfo.TYPE_LINE_ANALOG:
- return "TYPE_LINE_ANALOG";
- case AudioDeviceInfo.TYPE_LINE_DIGITAL:
- return "TYPE_LINE_DIGITAL";
- case AudioDeviceInfo.TYPE_BLUETOOTH_SCO:
- return "TYPE_BLUETOOTH_SCO";
- case AudioDeviceInfo.TYPE_BLUETOOTH_A2DP:
- return "TYPE_BLUETOOTH_A2DP";
- case AudioDeviceInfo.TYPE_HDMI:
- return "TYPE_HDMI";
- case AudioDeviceInfo.TYPE_HDMI_ARC:
- return "TYPE_HDMI_ARC";
- case AudioDeviceInfo.TYPE_USB_DEVICE:
- return "TYPE_USB_DEVICE";
- case AudioDeviceInfo.TYPE_USB_ACCESSORY:
- return "TYPE_USB_ACCESSORY";
- case AudioDeviceInfo.TYPE_DOCK:
- return "TYPE_DOCK";
- case AudioDeviceInfo.TYPE_FM:
- return "TYPE_FM";
- case AudioDeviceInfo.TYPE_BUILTIN_MIC:
- return "TYPE_BUILTIN_MIC";
- case AudioDeviceInfo.TYPE_FM_TUNER:
- return "TYPE_FM_TUNER";
- case AudioDeviceInfo.TYPE_TV_TUNER:
- return "TYPE_TV_TUNER";
- case AudioDeviceInfo.TYPE_TELEPHONY:
- return "TYPE_TELEPHONY";
- case AudioDeviceInfo.TYPE_AUX_LINE:
- return "TYPE_AUX_LINE";
- case AudioDeviceInfo.TYPE_IP:
- return "TYPE_IP";
- case AudioDeviceInfo.TYPE_BUS:
- return "TYPE_BUS";
- case AudioDeviceInfo.TYPE_USB_HEADSET:
- return "TYPE_USB_HEADSET";
- default:
- return "TYPE_UNKNOWN";
- }
- }
-
- // Returns true if the device can record audio via a microphone.
- private static boolean hasMicrophone() {
- return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
- PackageManager.FEATURE_MICROPHONE);
- }
-}
diff --git a/modules/audio_device/android/opensles_common.cc b/modules/audio_device/android/opensles_common.cc
deleted file mode 100644
index 019714d..0000000
--- a/modules/audio_device/android/opensles_common.cc
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#include "modules/audio_device/android/opensles_common.h"
-
-#include <SLES/OpenSLES.h>
-
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-
-// Returns a string representation given an integer SL_RESULT_XXX code.
-// The mapping can be found in <SLES/OpenSLES.h>.
-const char* GetSLErrorString(size_t code) {
- static const char* sl_error_strings[] = {
- "SL_RESULT_SUCCESS", // 0
- "SL_RESULT_PRECONDITIONS_VIOLATED", // 1
- "SL_RESULT_PARAMETER_INVALID", // 2
- "SL_RESULT_MEMORY_FAILURE", // 3
- "SL_RESULT_RESOURCE_ERROR", // 4
- "SL_RESULT_RESOURCE_LOST", // 5
- "SL_RESULT_IO_ERROR", // 6
- "SL_RESULT_BUFFER_INSUFFICIENT", // 7
- "SL_RESULT_CONTENT_CORRUPTED", // 8
- "SL_RESULT_CONTENT_UNSUPPORTED", // 9
- "SL_RESULT_CONTENT_NOT_FOUND", // 10
- "SL_RESULT_PERMISSION_DENIED", // 11
- "SL_RESULT_FEATURE_UNSUPPORTED", // 12
- "SL_RESULT_INTERNAL_ERROR", // 13
- "SL_RESULT_UNKNOWN_ERROR", // 14
- "SL_RESULT_OPERATION_ABORTED", // 15
- "SL_RESULT_CONTROL_LOST", // 16
- };
-
- if (code >= arraysize(sl_error_strings)) {
- return "SL_RESULT_UNKNOWN_ERROR";
- }
- return sl_error_strings[code];
-}
-
-SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
- int sample_rate,
- size_t bits_per_sample) {
- RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
- SLDataFormat_PCM format;
- format.formatType = SL_DATAFORMAT_PCM;
- format.numChannels = static_cast<SLuint32>(channels);
- // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
- switch (sample_rate) {
- case 8000:
- format.samplesPerSec = SL_SAMPLINGRATE_8;
- break;
- case 16000:
- format.samplesPerSec = SL_SAMPLINGRATE_16;
- break;
- case 22050:
- format.samplesPerSec = SL_SAMPLINGRATE_22_05;
- break;
- case 32000:
- format.samplesPerSec = SL_SAMPLINGRATE_32;
- break;
- case 44100:
- format.samplesPerSec = SL_SAMPLINGRATE_44_1;
- break;
- case 48000:
- format.samplesPerSec = SL_SAMPLINGRATE_48;
- break;
- case 64000:
- format.samplesPerSec = SL_SAMPLINGRATE_64;
- break;
- case 88200:
- format.samplesPerSec = SL_SAMPLINGRATE_88_2;
- break;
- case 96000:
- format.samplesPerSec = SL_SAMPLINGRATE_96;
- break;
- default:
- RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
- break;
- }
- format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
- format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
- format.endianness = SL_BYTEORDER_LITTLEENDIAN;
- if (format.numChannels == 1) {
- format.channelMask = SL_SPEAKER_FRONT_CENTER;
- } else if (format.numChannels == 2) {
- format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
- } else {
- RTC_CHECK(false) << "Unsupported number of channels: "
- << format.numChannels;
- }
- return format;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/opensles_common.h b/modules/audio_device/android/opensles_common.h
deleted file mode 100644
index 438c522..0000000
--- a/modules/audio_device/android/opensles_common.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2013 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
-
-#include <SLES/OpenSLES.h>
-#include <stddef.h>
-
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-
-// Returns a string representation given an integer SL_RESULT_XXX code.
-// The mapping can be found in <SLES/OpenSLES.h>.
-const char* GetSLErrorString(size_t code);
-
-// Configures an SL_DATAFORMAT_PCM structure based on native audio parameters.
-SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
- int sample_rate,
- size_t bits_per_sample);
-
-// Helper class for using SLObjectItf interfaces.
-template <typename SLType, typename SLDerefType>
-class ScopedSLObject {
- public:
- ScopedSLObject() : obj_(nullptr) {}
-
- ~ScopedSLObject() { Reset(); }
-
- SLType* Receive() {
- RTC_DCHECK(!obj_);
- return &obj_;
- }
-
- SLDerefType operator->() { return *obj_; }
-
- SLType Get() const { return obj_; }
-
- void Reset() {
- if (obj_) {
- (*obj_)->Destroy(obj_);
- obj_ = nullptr;
- }
- }
-
- private:
- SLType obj_;
-};
-
-typedef ScopedSLObject<SLObjectItf, const SLObjectItf_*> ScopedSLObjectItf;
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
diff --git a/modules/audio_device/android/opensles_player.cc b/modules/audio_device/android/opensles_player.cc
deleted file mode 100644
index f2b3a37..0000000
--- a/modules/audio_device/android/opensles_player.cc
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_device/android/opensles_player.h"
-
-#include <android/log.h>
-
-#include <memory>
-
-#include "api/array_view.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/fine_audio_buffer.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/platform_thread.h"
-#include "rtc_base/time_utils.h"
-
-#define TAG "OpenSLESPlayer"
-#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
-#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
-#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
-#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
-#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
-
-#define RETURN_ON_ERROR(op, ...) \
- do { \
- SLresult err = (op); \
- if (err != SL_RESULT_SUCCESS) { \
- ALOGE("%s failed: %s", #op, GetSLErrorString(err)); \
- return __VA_ARGS__; \
- } \
- } while (0)
-
-namespace webrtc {
-
-OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
- : audio_manager_(audio_manager),
- audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
- audio_device_buffer_(nullptr),
- initialized_(false),
- playing_(false),
- buffer_index_(0),
- engine_(nullptr),
- player_(nullptr),
- simple_buffer_queue_(nullptr),
- volume_(nullptr),
- last_play_time_(0) {
- ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
- // Use native audio output parameters provided by the audio manager and
- // define the PCM format structure.
- pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
- audio_parameters_.sample_rate(),
- audio_parameters_.bits_per_sample());
- // Detach from this thread since we want to use the checker to verify calls
- // from the internal audio thread.
- thread_checker_opensles_.Detach();
-}
-
-OpenSLESPlayer::~OpenSLESPlayer() {
- ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
- DestroyAudioPlayer();
- DestroyMix();
- engine_ = nullptr;
- RTC_DCHECK(!engine_);
- RTC_DCHECK(!output_mix_.Get());
- RTC_DCHECK(!player_);
- RTC_DCHECK(!simple_buffer_queue_);
- RTC_DCHECK(!volume_);
-}
-
-int OpenSLESPlayer::Init() {
- ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (audio_parameters_.channels() == 2) {
- ALOGW("Stereo mode is enabled");
- }
- return 0;
-}
-
-int OpenSLESPlayer::Terminate() {
- ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- StopPlayout();
- return 0;
-}
-
-int OpenSLESPlayer::InitPlayout() {
- ALOGD("InitPlayout[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!playing_);
- if (!ObtainEngineInterface()) {
- ALOGE("Failed to obtain SL Engine interface");
- return -1;
- }
- CreateMix();
- initialized_ = true;
- buffer_index_ = 0;
- return 0;
-}
-
-int OpenSLESPlayer::StartPlayout() {
- ALOGD("StartPlayout[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(initialized_);
- RTC_DCHECK(!playing_);
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetPlayout();
- }
- // The number of lower latency audio players is limited, hence we create the
- // audio player in Start() and destroy it in Stop().
- CreateAudioPlayer();
- // Fill up audio buffers to avoid initial glitch and to ensure that playback
- // starts when mode is later changed to SL_PLAYSTATE_PLAYING.
- // TODO(henrika): we can save some delay by only making one call to
- // EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
- last_play_time_ = rtc::Time();
- for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
- EnqueuePlayoutData(true);
- }
- // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
- // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
- // state, adding buffers will implicitly start playback.
- RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
- playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
- RTC_DCHECK(playing_);
- return 0;
-}
-
-int OpenSLESPlayer::StopPlayout() {
- ALOGD("StopPlayout[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_ || !playing_) {
- return 0;
- }
- // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
- RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
- // Clear the buffer queue to flush out any remaining data.
- RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
-#if RTC_DCHECK_IS_ON
- // Verify that the buffer queue is in fact cleared as it should.
- SLAndroidSimpleBufferQueueState buffer_queue_state;
- (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
- RTC_DCHECK_EQ(0, buffer_queue_state.count);
- RTC_DCHECK_EQ(0, buffer_queue_state.index);
-#endif
- // The number of lower latency audio players is limited, hence we create the
- // audio player in Start() and destroy it in Stop().
- DestroyAudioPlayer();
- thread_checker_opensles_.Detach();
- initialized_ = false;
- playing_ = false;
- return 0;
-}
-
-int OpenSLESPlayer::SpeakerVolumeIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int OpenSLESPlayer::MaxSpeakerVolume(uint32_t& maxVolume) const {
- return -1;
-}
-
-int OpenSLESPlayer::MinSpeakerVolume(uint32_t& minVolume) const {
- return -1;
-}
-
-int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
- return -1;
-}
-
-int OpenSLESPlayer::SpeakerVolume(uint32_t& volume) const {
- return -1;
-}
-
-void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- ALOGD("AttachAudioBuffer");
- RTC_DCHECK(thread_checker_.IsCurrent());
- audio_device_buffer_ = audioBuffer;
- const int sample_rate_hz = audio_parameters_.sample_rate();
- ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
- audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
- const size_t channels = audio_parameters_.channels();
- ALOGD("SetPlayoutChannels(%zu)", channels);
- audio_device_buffer_->SetPlayoutChannels(channels);
- RTC_CHECK(audio_device_buffer_);
- AllocateDataBuffers();
-}
-
-void OpenSLESPlayer::AllocateDataBuffers() {
- ALOGD("AllocateDataBuffers");
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!simple_buffer_queue_);
- RTC_CHECK(audio_device_buffer_);
- // Create a modified audio buffer class which allows us to ask for any number
- // of samples (and not only multiple of 10ms) to match the native OpenSL ES
- // buffer size. The native buffer size corresponds to the
- // PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
- // frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
- // recommended to construct audio buffers so that they contain an exact
- // multiple of this number. If so, callbacks will occur at regular intervals,
- // which reduces jitter.
- const size_t buffer_size_in_samples =
- audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
- ALOGD("native buffer size: %zu", buffer_size_in_samples);
- ALOGD("native buffer size in ms: %.2f",
- audio_parameters_.GetBufferSizeInMilliseconds());
- fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
- // Allocated memory for audio buffers.
- for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
- audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);
- }
-}
-
-bool OpenSLESPlayer::ObtainEngineInterface() {
- ALOGD("ObtainEngineInterface");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (engine_)
- return true;
- // Get access to (or create if not already existing) the global OpenSL Engine
- // object.
- SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
- if (engine_object == nullptr) {
- ALOGE("Failed to access the global OpenSL engine");
- return false;
- }
- // Get the SL Engine Interface which is implicit.
- RETURN_ON_ERROR(
- (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine_),
- false);
- return true;
-}
-
-bool OpenSLESPlayer::CreateMix() {
- ALOGD("CreateMix");
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(engine_);
- if (output_mix_.Get())
- return true;
-
- // Create the ouput mix on the engine object. No interfaces will be used.
- RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
- nullptr, nullptr),
- false);
- RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
- false);
- return true;
-}
-
-void OpenSLESPlayer::DestroyMix() {
- ALOGD("DestroyMix");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!output_mix_.Get())
- return;
- output_mix_.Reset();
-}
-
-bool OpenSLESPlayer::CreateAudioPlayer() {
- ALOGD("CreateAudioPlayer");
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(output_mix_.Get());
- if (player_object_.Get())
- return true;
- RTC_DCHECK(!player_);
- RTC_DCHECK(!simple_buffer_queue_);
- RTC_DCHECK(!volume_);
-
- // source: Android Simple Buffer Queue Data Locator is source.
- SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
- SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
- static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
- SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
-
- // sink: OutputMix-based data is sink.
- SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
- output_mix_.Get()};
- SLDataSink audio_sink = {&locator_output_mix, nullptr};
-
- // Define interfaces that we indend to use and realize.
- const SLInterfaceID interface_ids[] = {SL_IID_ANDROIDCONFIGURATION,
- SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
- const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
- SL_BOOLEAN_TRUE};
-
- // Create the audio player on the engine interface.
- RETURN_ON_ERROR(
- (*engine_)->CreateAudioPlayer(
- engine_, player_object_.Receive(), &audio_source, &audio_sink,
- arraysize(interface_ids), interface_ids, interface_required),
- false);
-
- // Use the Android configuration interface to set platform-specific
- // parameters. Should be done before player is realized.
- SLAndroidConfigurationItf player_config;
- RETURN_ON_ERROR(
- player_object_->GetInterface(player_object_.Get(),
- SL_IID_ANDROIDCONFIGURATION, &player_config),
- false);
- // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
- // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
- SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
- RETURN_ON_ERROR(
- (*player_config)
- ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
- &stream_type, sizeof(SLint32)),
- false);
-
- // Realize the audio player object after configuration has been set.
- RETURN_ON_ERROR(
- player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
-
- // Get the SLPlayItf interface on the audio player.
- RETURN_ON_ERROR(
- player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY, &player_),
- false);
-
- // Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
- RETURN_ON_ERROR(
- player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE,
- &simple_buffer_queue_),
- false);
-
- // Register callback method for the Android Simple Buffer Queue interface.
- // This method will be called when the native audio layer needs audio data.
- RETURN_ON_ERROR((*simple_buffer_queue_)
- ->RegisterCallback(simple_buffer_queue_,
- SimpleBufferQueueCallback, this),
- false);
-
- // Get the SLVolumeItf interface on the audio player.
- RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
- SL_IID_VOLUME, &volume_),
- false);
-
- // TODO(henrika): might not be required to set volume to max here since it
- // seems to be default on most devices. Might be required for unit tests.
- // RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
-
- return true;
-}
-
-void OpenSLESPlayer::DestroyAudioPlayer() {
- ALOGD("DestroyAudioPlayer");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!player_object_.Get())
- return;
- (*simple_buffer_queue_)
- ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
- player_object_.Reset();
- player_ = nullptr;
- simple_buffer_queue_ = nullptr;
- volume_ = nullptr;
-}
-
-// static
-void OpenSLESPlayer::SimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf caller,
- void* context) {
- OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
- stream->FillBufferQueue();
-}
-
-void OpenSLESPlayer::FillBufferQueue() {
- RTC_DCHECK(thread_checker_opensles_.IsCurrent());
- SLuint32 state = GetPlayState();
- if (state != SL_PLAYSTATE_PLAYING) {
- ALOGW("Buffer callback in non-playing state!");
- return;
- }
- EnqueuePlayoutData(false);
-}
-
-void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
- // Check delta time between two successive callbacks and provide a warning
- // if it becomes very large.
- // TODO(henrika): using 150ms as upper limit but this value is rather random.
- const uint32_t current_time = rtc::Time();
- const uint32_t diff = current_time - last_play_time_;
- if (diff > 150) {
- ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
- }
- last_play_time_ = current_time;
- SLint8* audio_ptr8 =
- reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get());
- if (silence) {
- RTC_DCHECK(thread_checker_.IsCurrent());
- // Avoid acquiring real audio data from WebRTC and fill the buffer with
- // zeros instead. Used to prime the buffer with silence and to avoid asking
- // for audio data from two different threads.
- memset(audio_ptr8, 0, audio_parameters_.GetBytesPerBuffer());
- } else {
- RTC_DCHECK(thread_checker_opensles_.IsCurrent());
- // Read audio data from the WebRTC source using the FineAudioBuffer object
- // to adjust for differences in buffer size between WebRTC (10ms) and native
- // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
- // delay estimation.
- fine_audio_buffer_->GetPlayoutData(
- rtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
- audio_parameters_.frames_per_buffer() *
- audio_parameters_.channels()),
- 25);
- }
- // Enqueue the decoded audio buffer for playback.
- SLresult err = (*simple_buffer_queue_)
- ->Enqueue(simple_buffer_queue_, audio_ptr8,
- audio_parameters_.GetBytesPerBuffer());
- if (SL_RESULT_SUCCESS != err) {
- ALOGE("Enqueue failed: %d", err);
- }
- buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
-}
-
-SLuint32 OpenSLESPlayer::GetPlayState() const {
- RTC_DCHECK(player_);
- SLuint32 state;
- SLresult err = (*player_)->GetPlayState(player_, &state);
- if (SL_RESULT_SUCCESS != err) {
- ALOGE("GetPlayState failed: %d", err);
- }
- return state;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/opensles_player.h b/modules/audio_device/android/opensles_player.h
deleted file mode 100644
index 41593a4..0000000
--- a/modules/audio_device/android/opensles_player.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
-
-#include <SLES/OpenSLES.h>
-#include <SLES/OpenSLES_Android.h>
-#include <SLES/OpenSLES_AndroidConfiguration.h>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/android/opensles_common.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "modules/utility/include/helpers_android.h"
-
-namespace webrtc {
-
-class FineAudioBuffer;
-
-// Implements 16-bit mono PCM audio output support for Android using the
-// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will RTC_DCHECK if any method is called on an invalid thread. Decoded audio
-// buffers are requested on a dedicated internal thread managed by the OpenSL
-// ES layer.
-//
-// The existing design forces the user to call InitPlayout() after Stoplayout()
-// to be able to call StartPlayout() again. This is inline with how the Java-
-// based implementation works.
-//
-// OpenSL ES is a native C API which have no Dalvik-related overhead such as
-// garbage collection pauses and it supports reduced audio output latency.
-// If the device doesn't claim this feature but supports API level 9 (Android
-// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
-// the output latency may be higher.
-class OpenSLESPlayer {
- public:
- // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
- // required for lower latency. Beginning with API level 18 (Android 4.3), a
- // buffer count of 1 is sufficient for lower latency. In addition, the buffer
- // size and sample rate must be compatible with the device's native output
- // configuration provided via the audio manager at construction.
- // TODO(henrika): perhaps set this value dynamically based on OS version.
- static const int kNumOfOpenSLESBuffers = 2;
-
- explicit OpenSLESPlayer(AudioManager* audio_manager);
- ~OpenSLESPlayer();
-
- int Init();
- int Terminate();
-
- int InitPlayout();
- bool PlayoutIsInitialized() const { return initialized_; }
-
- int StartPlayout();
- int StopPlayout();
- bool Playing() const { return playing_; }
-
- int SpeakerVolumeIsAvailable(bool& available);
- int SetSpeakerVolume(uint32_t volume);
- int SpeakerVolume(uint32_t& volume) const;
- int MaxSpeakerVolume(uint32_t& maxVolume) const;
- int MinSpeakerVolume(uint32_t& minVolume) const;
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- private:
- // These callback methods are called when data is required for playout.
- // They are both called from an internal "OpenSL ES thread" which is not
- // attached to the Dalvik VM.
- static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
- void* context);
- void FillBufferQueue();
- // Reads audio data in PCM format using the AudioDeviceBuffer.
- // Can be called both on the main thread (during Start()) and from the
- // internal audio thread while output streaming is active.
- // If the `silence` flag is set, the audio is filled with zeros instead of
- // asking the WebRTC layer for real audio data. This procedure is also known
- // as audio priming.
- void EnqueuePlayoutData(bool silence);
-
- // Allocate memory for audio buffers which will be used to render audio
- // via the SLAndroidSimpleBufferQueueItf interface.
- void AllocateDataBuffers();
-
- // Obtaines the SL Engine Interface from the existing global Engine object.
- // The interface exposes creation methods of all the OpenSL ES object types.
- // This method defines the `engine_` member variable.
- bool ObtainEngineInterface();
-
- // Creates/destroys the output mix object.
- bool CreateMix();
- void DestroyMix();
-
- // Creates/destroys the audio player and the simple-buffer object.
- // Also creates the volume object.
- bool CreateAudioPlayer();
- void DestroyAudioPlayer();
-
- SLuint32 GetPlayState() const;
-
- // Ensures that methods are called from the same thread as this object is
- // created on.
- SequenceChecker thread_checker_;
-
- // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
- // non-application thread which is not attached to the Dalvik JVM.
- // Detached during construction of this object.
- SequenceChecker thread_checker_opensles_;
-
- // Raw pointer to the audio manager injected at construction. Used to cache
- // audio parameters and to access the global SL engine object needed by the
- // ObtainEngineInterface() method. The audio manager outlives any instance of
- // this class.
- AudioManager* audio_manager_;
-
- // Contains audio parameters provided to this class at construction by the
- // AudioManager.
- const AudioParameters audio_parameters_;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
- AudioDeviceBuffer* audio_device_buffer_;
-
- bool initialized_;
- bool playing_;
-
- // PCM-type format definition.
- // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
- // 32-bit float representation is needed.
- SLDataFormat_PCM pcm_format_;
-
- // Queue of audio buffers to be used by the player object for rendering
- // audio.
- std::unique_ptr<SLint16[]> audio_buffers_[kNumOfOpenSLESBuffers];
-
- // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
- // in chunks of 10ms. It then allows for this data to be pulled in
- // a finer or coarser granularity. I.e. interacting with this class instead
- // of directly with the AudioDeviceBuffer one can ask for any number of
- // audio data samples.
- // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
- // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 192
- // in each callback (one every 4th ms). This class can then ask for 192 and
- // the FineAudioBuffer will ask WebRTC for new data approximately only every
- // second callback and also cache non-utilized audio.
- std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
-
- // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
- // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
- int buffer_index_;
-
- // This interface exposes creation methods for all the OpenSL ES object types.
- // It is the OpenSL ES API entry point.
- SLEngineItf engine_;
-
- // Output mix object to be used by the player object.
- webrtc::ScopedSLObjectItf output_mix_;
-
- // The audio player media object plays out audio to the speakers. It also
- // supports volume control.
- webrtc::ScopedSLObjectItf player_object_;
-
- // This interface is supported on the audio player and it controls the state
- // of the audio player.
- SLPlayItf player_;
-
- // The Android Simple Buffer Queue interface is supported on the audio player
- // and it provides methods to send audio data from the source to the audio
- // player for rendering.
- SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
-
- // This interface exposes controls for manipulating the object’s audio volume
- // properties. This interface is supported on the Audio Player object.
- SLVolumeItf volume_;
-
- // Last time the OpenSL ES layer asked for audio data to play out.
- uint32_t last_play_time_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
diff --git a/modules/audio_device/android/opensles_recorder.cc b/modules/audio_device/android/opensles_recorder.cc
deleted file mode 100644
index 4e0c26d..0000000
--- a/modules/audio_device/android/opensles_recorder.cc
+++ /dev/null
@@ -1,431 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_device/android/opensles_recorder.h"
-
-#include <android/log.h>
-
-#include <memory>
-
-#include "api/array_view.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/fine_audio_buffer.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/platform_thread.h"
-#include "rtc_base/time_utils.h"
-
-#define TAG "OpenSLESRecorder"
-#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
-#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
-#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
-#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
-#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
-
-#define LOG_ON_ERROR(op) \
- [](SLresult err) { \
- if (err != SL_RESULT_SUCCESS) { \
- ALOGE("%s:%d %s failed: %s", __FILE__, __LINE__, #op, \
- GetSLErrorString(err)); \
- return true; \
- } \
- return false; \
- }(op)
-
-namespace webrtc {
-
-OpenSLESRecorder::OpenSLESRecorder(AudioManager* audio_manager)
- : audio_manager_(audio_manager),
- audio_parameters_(audio_manager->GetRecordAudioParameters()),
- audio_device_buffer_(nullptr),
- initialized_(false),
- recording_(false),
- engine_(nullptr),
- recorder_(nullptr),
- simple_buffer_queue_(nullptr),
- buffer_index_(0),
- last_rec_time_(0) {
- ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
- // Detach from this thread since we want to use the checker to verify calls
- // from the internal audio thread.
- thread_checker_opensles_.Detach();
- // Use native audio output parameters provided by the audio manager and
- // define the PCM format structure.
- pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
- audio_parameters_.sample_rate(),
- audio_parameters_.bits_per_sample());
-}
-
-OpenSLESRecorder::~OpenSLESRecorder() {
- ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
- DestroyAudioRecorder();
- engine_ = nullptr;
- RTC_DCHECK(!engine_);
- RTC_DCHECK(!recorder_);
- RTC_DCHECK(!simple_buffer_queue_);
-}
-
-int OpenSLESRecorder::Init() {
- ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (audio_parameters_.channels() == 2) {
- ALOGD("Stereo mode is enabled");
- }
- return 0;
-}
-
-int OpenSLESRecorder::Terminate() {
- ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- StopRecording();
- return 0;
-}
-
-int OpenSLESRecorder::InitRecording() {
- ALOGD("InitRecording[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!initialized_);
- RTC_DCHECK(!recording_);
- if (!ObtainEngineInterface()) {
- ALOGE("Failed to obtain SL Engine interface");
- return -1;
- }
- CreateAudioRecorder();
- initialized_ = true;
- buffer_index_ = 0;
- return 0;
-}
-
-int OpenSLESRecorder::StartRecording() {
- ALOGD("StartRecording[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(initialized_);
- RTC_DCHECK(!recording_);
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetRecord();
- }
- // Add buffers to the queue before changing state to SL_RECORDSTATE_RECORDING
- // to ensure that recording starts as soon as the state is modified. On some
- // devices, SLAndroidSimpleBufferQueue::Clear() used in Stop() does not flush
- // the buffers as intended and we therefore check the number of buffers
- // already queued first. Enqueue() can return SL_RESULT_BUFFER_INSUFFICIENT
- // otherwise.
- int num_buffers_in_queue = GetBufferCount();
- for (int i = 0; i < kNumOfOpenSLESBuffers - num_buffers_in_queue; ++i) {
- if (!EnqueueAudioBuffer()) {
- recording_ = false;
- return -1;
- }
- }
- num_buffers_in_queue = GetBufferCount();
- RTC_DCHECK_EQ(num_buffers_in_queue, kNumOfOpenSLESBuffers);
- LogBufferState();
- // Start audio recording by changing the state to SL_RECORDSTATE_RECORDING.
- // Given that buffers are already enqueued, recording should start at once.
- // The macro returns -1 if recording fails to start.
- last_rec_time_ = rtc::Time();
- if (LOG_ON_ERROR(
- (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_RECORDING))) {
- return -1;
- }
- recording_ = (GetRecordState() == SL_RECORDSTATE_RECORDING);
- RTC_DCHECK(recording_);
- return 0;
-}
-
-int OpenSLESRecorder::StopRecording() {
- ALOGD("StopRecording[tid=%d]", rtc::CurrentThreadId());
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!initialized_ || !recording_) {
- return 0;
- }
- // Stop recording by setting the record state to SL_RECORDSTATE_STOPPED.
- if (LOG_ON_ERROR(
- (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_STOPPED))) {
- return -1;
- }
- // Clear the buffer queue to get rid of old data when resuming recording.
- if (LOG_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_))) {
- return -1;
- }
- thread_checker_opensles_.Detach();
- initialized_ = false;
- recording_ = false;
- return 0;
-}
-
-void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
- ALOGD("AttachAudioBuffer");
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_CHECK(audio_buffer);
- audio_device_buffer_ = audio_buffer;
- // Ensure that the audio device buffer is informed about the native sample
- // rate used on the recording side.
- const int sample_rate_hz = audio_parameters_.sample_rate();
- ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
- audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
- // Ensure that the audio device buffer is informed about the number of
- // channels preferred by the OS on the recording side.
- const size_t channels = audio_parameters_.channels();
- ALOGD("SetRecordingChannels(%zu)", channels);
- audio_device_buffer_->SetRecordingChannels(channels);
- // Allocated memory for internal data buffers given existing audio parameters.
- AllocateDataBuffers();
-}
-
-int OpenSLESRecorder::EnableBuiltInAEC(bool enable) {
- ALOGD("EnableBuiltInAEC(%d)", enable);
- RTC_DCHECK(thread_checker_.IsCurrent());
- ALOGE("Not implemented");
- return 0;
-}
-
-int OpenSLESRecorder::EnableBuiltInAGC(bool enable) {
- ALOGD("EnableBuiltInAGC(%d)", enable);
- RTC_DCHECK(thread_checker_.IsCurrent());
- ALOGE("Not implemented");
- return 0;
-}
-
-int OpenSLESRecorder::EnableBuiltInNS(bool enable) {
- ALOGD("EnableBuiltInNS(%d)", enable);
- RTC_DCHECK(thread_checker_.IsCurrent());
- ALOGE("Not implemented");
- return 0;
-}
-
-bool OpenSLESRecorder::ObtainEngineInterface() {
- ALOGD("ObtainEngineInterface");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (engine_)
- return true;
- // Get access to (or create if not already existing) the global OpenSL Engine
- // object.
- SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
- if (engine_object == nullptr) {
- ALOGE("Failed to access the global OpenSL engine");
- return false;
- }
- // Get the SL Engine Interface which is implicit.
- if (LOG_ON_ERROR(
- (*engine_object)
- ->GetInterface(engine_object, SL_IID_ENGINE, &engine_))) {
- return false;
- }
- return true;
-}
-
-bool OpenSLESRecorder::CreateAudioRecorder() {
- ALOGD("CreateAudioRecorder");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (recorder_object_.Get())
- return true;
- RTC_DCHECK(!recorder_);
- RTC_DCHECK(!simple_buffer_queue_);
-
- // Audio source configuration.
- SLDataLocator_IODevice mic_locator = {SL_DATALOCATOR_IODEVICE,
- SL_IODEVICE_AUDIOINPUT,
- SL_DEFAULTDEVICEID_AUDIOINPUT, NULL};
- SLDataSource audio_source = {&mic_locator, NULL};
-
- // Audio sink configuration.
- SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {
- SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
- static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
- SLDataSink audio_sink = {&buffer_queue, &pcm_format_};
-
- // Create the audio recorder object (requires the RECORD_AUDIO permission).
- // Do not realize the recorder yet. Set the configuration first.
- const SLInterfaceID interface_id[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
- SL_IID_ANDROIDCONFIGURATION};
- const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
- if (LOG_ON_ERROR((*engine_)->CreateAudioRecorder(
- engine_, recorder_object_.Receive(), &audio_source, &audio_sink,
- arraysize(interface_id), interface_id, interface_required))) {
- return false;
- }
-
- // Configure the audio recorder (before it is realized).
- SLAndroidConfigurationItf recorder_config;
- if (LOG_ON_ERROR((recorder_object_->GetInterface(recorder_object_.Get(),
- SL_IID_ANDROIDCONFIGURATION,
- &recorder_config)))) {
- return false;
- }
-
- // Uses the default microphone tuned for audio communication.
- // Note that, SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION leads to a fast
- // track but also excludes usage of required effects like AEC, AGC and NS.
- // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION
- SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
- if (LOG_ON_ERROR(((*recorder_config)
- ->SetConfiguration(recorder_config,
- SL_ANDROID_KEY_RECORDING_PRESET,
- &stream_type, sizeof(SLint32))))) {
- return false;
- }
-
- // The audio recorder can now be realized (in synchronous mode).
- if (LOG_ON_ERROR((recorder_object_->Realize(recorder_object_.Get(),
- SL_BOOLEAN_FALSE)))) {
- return false;
- }
-
- // Get the implicit recorder interface (SL_IID_RECORD).
- if (LOG_ON_ERROR((recorder_object_->GetInterface(
- recorder_object_.Get(), SL_IID_RECORD, &recorder_)))) {
- return false;
- }
-
- // Get the simple buffer queue interface (SL_IID_ANDROIDSIMPLEBUFFERQUEUE).
- // It was explicitly requested.
- if (LOG_ON_ERROR((recorder_object_->GetInterface(
- recorder_object_.Get(), SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
- &simple_buffer_queue_)))) {
- return false;
- }
-
- // Register the input callback for the simple buffer queue.
- // This callback will be called when receiving new data from the device.
- if (LOG_ON_ERROR(((*simple_buffer_queue_)
- ->RegisterCallback(simple_buffer_queue_,
- SimpleBufferQueueCallback, this)))) {
- return false;
- }
- return true;
-}
-
-void OpenSLESRecorder::DestroyAudioRecorder() {
- ALOGD("DestroyAudioRecorder");
- RTC_DCHECK(thread_checker_.IsCurrent());
- if (!recorder_object_.Get())
- return;
- (*simple_buffer_queue_)
- ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
- recorder_object_.Reset();
- recorder_ = nullptr;
- simple_buffer_queue_ = nullptr;
-}
-
-void OpenSLESRecorder::SimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf buffer_queue,
- void* context) {
- OpenSLESRecorder* stream = static_cast<OpenSLESRecorder*>(context);
- stream->ReadBufferQueue();
-}
-
-void OpenSLESRecorder::AllocateDataBuffers() {
- ALOGD("AllocateDataBuffers");
- RTC_DCHECK(thread_checker_.IsCurrent());
- RTC_DCHECK(!simple_buffer_queue_);
- RTC_CHECK(audio_device_buffer_);
- // Create a modified audio buffer class which allows us to deliver any number
- // of samples (and not only multiple of 10ms) to match the native audio unit
- // buffer size.
- ALOGD("frames per native buffer: %zu", audio_parameters_.frames_per_buffer());
- ALOGD("frames per 10ms buffer: %zu",
- audio_parameters_.frames_per_10ms_buffer());
- ALOGD("bytes per native buffer: %zu", audio_parameters_.GetBytesPerBuffer());
- ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
- RTC_DCHECK(audio_device_buffer_);
- fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
- // Allocate queue of audio buffers that stores recorded audio samples.
- const int buffer_size_samples =
- audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
- audio_buffers_.reset(new std::unique_ptr<SLint16[]>[kNumOfOpenSLESBuffers]);
- for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
- audio_buffers_[i].reset(new SLint16[buffer_size_samples]);
- }
-}
-
-void OpenSLESRecorder::ReadBufferQueue() {
- RTC_DCHECK(thread_checker_opensles_.IsCurrent());
- SLuint32 state = GetRecordState();
- if (state != SL_RECORDSTATE_RECORDING) {
- ALOGW("Buffer callback in non-recording state!");
- return;
- }
- // Check delta time between two successive callbacks and provide a warning
- // if it becomes very large.
- // TODO(henrika): using 150ms as upper limit but this value is rather random.
- const uint32_t current_time = rtc::Time();
- const uint32_t diff = current_time - last_rec_time_;
- if (diff > 150) {
- ALOGW("Bad OpenSL ES record timing, dT=%u [ms]", diff);
- }
- last_rec_time_ = current_time;
- // Send recorded audio data to the WebRTC sink.
- // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
- // since there is no support to turn off built-in EC in combination with
- // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
- // these estimates) will never be active.
- fine_audio_buffer_->DeliverRecordedData(
- rtc::ArrayView<const int16_t>(
- audio_buffers_[buffer_index_].get(),
- audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
- 25);
- // Enqueue the utilized audio buffer and use if for recording again.
- EnqueueAudioBuffer();
-}
-
-bool OpenSLESRecorder::EnqueueAudioBuffer() {
- SLresult err =
- (*simple_buffer_queue_)
- ->Enqueue(
- simple_buffer_queue_,
- reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get()),
- audio_parameters_.GetBytesPerBuffer());
- if (SL_RESULT_SUCCESS != err) {
- ALOGE("Enqueue failed: %s", GetSLErrorString(err));
- return false;
- }
- buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
- return true;
-}
-
-SLuint32 OpenSLESRecorder::GetRecordState() const {
- RTC_DCHECK(recorder_);
- SLuint32 state;
- SLresult err = (*recorder_)->GetRecordState(recorder_, &state);
- if (SL_RESULT_SUCCESS != err) {
- ALOGE("GetRecordState failed: %s", GetSLErrorString(err));
- }
- return state;
-}
-
-SLAndroidSimpleBufferQueueState OpenSLESRecorder::GetBufferQueueState() const {
- RTC_DCHECK(simple_buffer_queue_);
- // state.count: Number of buffers currently in the queue.
- // state.index: Index of the currently filling buffer. This is a linear index
- // that keeps a cumulative count of the number of buffers recorded.
- SLAndroidSimpleBufferQueueState state;
- SLresult err =
- (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &state);
- if (SL_RESULT_SUCCESS != err) {
- ALOGE("GetState failed: %s", GetSLErrorString(err));
- }
- return state;
-}
-
-void OpenSLESRecorder::LogBufferState() const {
- SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
- ALOGD("state.count:%d state.index:%d", state.count, state.index);
-}
-
-SLuint32 OpenSLESRecorder::GetBufferCount() {
- SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
- return state.count;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/android/opensles_recorder.h b/modules/audio_device/android/opensles_recorder.h
deleted file mode 100644
index e659c3c..0000000
--- a/modules/audio_device/android/opensles_recorder.h
+++ /dev/null
@@ -1,193 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
-#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
-
-#include <SLES/OpenSLES.h>
-#include <SLES/OpenSLES_Android.h>
-#include <SLES/OpenSLES_AndroidConfiguration.h>
-
-#include <memory>
-
-#include "api/sequence_checker.h"
-#include "modules/audio_device/android/audio_common.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/android/opensles_common.h"
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "modules/utility/include/helpers_android.h"
-
-namespace webrtc {
-
-class FineAudioBuffer;
-
-// Implements 16-bit mono PCM audio input support for Android using the
-// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All public methods must also be called on the same thread. A thread checker
-// will RTC_DCHECK if any method is called on an invalid thread. Recorded audio
-// buffers are provided on a dedicated internal thread managed by the OpenSL
-// ES layer.
-//
-// The existing design forces the user to call InitRecording() after
-// StopRecording() to be able to call StartRecording() again. This is inline
-// with how the Java-based implementation works.
-//
-// 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. The capability for lower latency output is a prerequisite for the
-// lower latency input feature. Then, create an AudioRecorder with the same
-// sample rate and buffer size as would be used for output. OpenSL ES interfaces
-// for input effects preclude the lower latency path.
-// See https://developer.android.com/ndk/guides/audio/opensl-prog-notes.html
-// for more details.
-class OpenSLESRecorder {
- public:
- // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
- // required for lower latency. Beginning with API level 18 (Android 4.3), a
- // buffer count of 1 is sufficient for lower latency. In addition, the buffer
- // size and sample rate must be compatible with the device's native input
- // configuration provided via the audio manager at construction.
- // TODO(henrika): perhaps set this value dynamically based on OS version.
- static const int kNumOfOpenSLESBuffers = 2;
-
- explicit OpenSLESRecorder(AudioManager* audio_manager);
- ~OpenSLESRecorder();
-
- int Init();
- int Terminate();
-
- int InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
-
- int StartRecording();
- int StopRecording();
- bool Recording() const { return recording_; }
-
- void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer);
-
- // TODO(henrika): add support using OpenSL ES APIs when available.
- int EnableBuiltInAEC(bool enable);
- int EnableBuiltInAGC(bool enable);
- int EnableBuiltInNS(bool enable);
-
- private:
- // Obtaines the SL Engine Interface from the existing global Engine object.
- // The interface exposes creation methods of all the OpenSL ES object types.
- // This method defines the `engine_` member variable.
- bool ObtainEngineInterface();
-
- // Creates/destroys the audio recorder and the simple-buffer queue object.
- bool CreateAudioRecorder();
- void DestroyAudioRecorder();
-
- // Allocate memory for audio buffers which will be used to capture audio
- // via the SLAndroidSimpleBufferQueueItf interface.
- void AllocateDataBuffers();
-
- // These callback methods are called when data has been written to the input
- // buffer queue. They are both called from an internal "OpenSL ES thread"
- // which is not attached to the Dalvik VM.
- static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
- void* context);
- void ReadBufferQueue();
-
- // Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
- // called both on the main thread (but before recording has started) and from
- // the internal audio thread while input streaming is active. It uses
- // `simple_buffer_queue_` but no lock is needed since the initial calls from
- // the main thread and the native callback thread are mutually exclusive.
- bool EnqueueAudioBuffer();
-
- // Returns the current recorder state.
- SLuint32 GetRecordState() const;
-
- // Returns the current buffer queue state.
- SLAndroidSimpleBufferQueueState GetBufferQueueState() const;
-
- // Number of buffers currently in the queue.
- SLuint32 GetBufferCount();
-
- // Prints a log message of the current queue state. Can be used for debugging
- // purposes.
- void LogBufferState() const;
-
- // Ensures that methods are called from the same thread as this object is
- // created on.
- SequenceChecker thread_checker_;
-
- // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
- // non-application thread which is not attached to the Dalvik JVM.
- // Detached during construction of this object.
- SequenceChecker thread_checker_opensles_;
-
- // Raw pointer to the audio manager injected at construction. Used to cache
- // audio parameters and to access the global SL engine object needed by the
- // ObtainEngineInterface() method. The audio manager outlives any instance of
- // this class.
- AudioManager* const audio_manager_;
-
- // Contains audio parameters provided to this class at construction by the
- // AudioManager.
- const AudioParameters audio_parameters_;
-
- // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
- // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
- AudioDeviceBuffer* audio_device_buffer_;
-
- // PCM-type format definition.
- // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
- // 32-bit float representation is needed.
- SLDataFormat_PCM pcm_format_;
-
- bool initialized_;
- bool recording_;
-
- // This interface exposes creation methods for all the OpenSL ES object types.
- // It is the OpenSL ES API entry point.
- SLEngineItf engine_;
-
- // The audio recorder media object records audio to the destination specified
- // by the data sink capturing it from the input specified by the data source.
- webrtc::ScopedSLObjectItf recorder_object_;
-
- // This interface is supported on the audio recorder object and it controls
- // the state of the audio recorder.
- SLRecordItf recorder_;
-
- // The Android Simple Buffer Queue interface is supported on the audio
- // recorder. For recording, an app should enqueue empty buffers. When a
- // registered callback sends notification that the system has finished writing
- // data to the buffer, the app can read the buffer.
- SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
-
- // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
- // chunks of audio.
- std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
-
- // Queue of audio buffers to be used by the recorder object for capturing
- // audio. They will be used in a Round-robin way and the size of each buffer
- // is given by AudioParameters::frames_per_buffer(), i.e., it corresponds to
- // the native OpenSL ES buffer size.
- std::unique_ptr<std::unique_ptr<SLint16[]>[]> audio_buffers_;
-
- // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
- // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
- int buffer_index_;
-
- // Last time the OpenSL ES layer delivered recorded audio data.
- uint32_t last_rec_time_;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index 092b98f..9da9c62 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -26,16 +26,7 @@
#endif
#elif defined(WEBRTC_ANDROID)
#include <stdlib.h>
-#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
-#include "modules/audio_device/android/aaudio_player.h"
-#include "modules/audio_device/android/aaudio_recorder.h"
-#endif
-#include "modules/audio_device/android/audio_device_template.h"
-#include "modules/audio_device/android/audio_manager.h"
-#include "modules/audio_device/android/audio_record_jni.h"
-#include "modules/audio_device/android/audio_track_jni.h"
-#include "modules/audio_device/android/opensles_player.h"
-#include "modules/audio_device/android/opensles_recorder.h"
+#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
#elif defined(WEBRTC_LINUX)
#if defined(WEBRTC_ENABLE_LINUX_ALSA)
#include "modules/audio_device/linux/audio_device_alsa_linux.h"
@@ -74,7 +65,11 @@
AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
+#if defined(WEBRTC_ANDROID)
+ return CreateAndroidAudioDeviceModule(audio_layer);
+#else
return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
+#endif
}
// static
@@ -89,6 +84,14 @@
RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() "
"factory method instead for this option.";
return nullptr;
+ } else if (audio_layer == AudioDeviceModule::kAndroidJavaAudio ||
+ audio_layer == AudioDeviceModule::kAndroidOpenSLESAudio ||
+ audio_layer == AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio ||
+ audio_layer == kAndroidAAudioAudio ||
+ audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
+ RTC_LOG(LS_ERROR) << "Use the CreateAndroidAudioDeviceModule() "
+ "factory method instead for this option.";
+ return nullptr;
}
// Create the generic reference counted (platform independent) implementation.
@@ -182,70 +185,13 @@
}
#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, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when
// WEBRTC_LINUX is defined. WEBRTC_ENABLE_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(WEBRTC_ANDROID) && defined(WEBRTC_LINUX)
#if !defined(WEBRTC_ENABLE_LINUX_PULSE)
// Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
// - kPlatformDefaultAudio => ALSA, and
diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h
index 45f73dc..1737b46 100644
--- a/modules/audio_device/audio_device_impl.h
+++ b/modules/audio_device/audio_device_impl.h
@@ -24,7 +24,6 @@
namespace webrtc {
class AudioDeviceGeneric;
-class AudioManager;
class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
public:
@@ -145,12 +144,6 @@
int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
-#if defined(WEBRTC_ANDROID)
- // Only use this acccessor for test purposes on Android.
- AudioManager* GetAndroidAudioManagerForTest() {
- return audio_manager_android_.get();
- }
-#endif
AudioDeviceBuffer* GetAudioDeviceBuffer() { return &audio_device_buffer_; }
int RestartPlayoutInternally() override { return -1; }
@@ -165,10 +158,6 @@
AudioLayer audio_layer_;
PlatformType platform_type_ = kPlatformNotSupported;
bool initialized_ = false;
-#if defined(WEBRTC_ANDROID)
- // Should be declared first to ensure that it outlives other resources.
- std::unique_ptr<AudioManager> audio_manager_android_;
-#endif
AudioDeviceBuffer audio_device_buffer_;
std::unique_ptr<AudioDeviceGeneric> audio_device_;
};
diff --git a/modules/audio_device/g3doc/audio_device_module.md b/modules/audio_device/g3doc/audio_device_module.md
index 101b2e4..84cfb41 100644
--- a/modules/audio_device/g3doc/audio_device_module.md
+++ b/modules/audio_device/g3doc/audio_device_module.md
@@ -5,8 +5,8 @@
## Overview
-The ADM is responsible for driving input (microphone) and output (speaker) audio
-in WebRTC and the API is defined in [audio_device.h][19].
+The ADM(AudioDeviceModule) is responsible for driving input (microphone) and
+output (speaker) audio in WebRTC and the API is defined in [audio_device.h][19].
Main functions of the ADM are:
diff --git a/modules/utility/source/jvm_android.cc b/modules/utility/source/jvm_android.cc
index ee9930b..e0c66d5 100644
--- a/modules/utility/source/jvm_android.cc
+++ b/modules/utility/source/jvm_android.cc
@@ -27,10 +27,6 @@
const char* name;
jclass clazz;
} loaded_classes[] = {
- {"org/webrtc/voiceengine/BuildInfo", nullptr},
- {"org/webrtc/voiceengine/WebRtcAudioManager", nullptr},
- {"org/webrtc/voiceengine/WebRtcAudioRecord", nullptr},
- {"org/webrtc/voiceengine/WebRtcAudioTrack", nullptr},
};
// Android's FindClass() is trickier than usual because the app-specific
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 8611707..d662fab 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -55,7 +55,6 @@
":swcodecs_java",
":video_api_java",
":video_java",
- "../../modules/audio_device:audio_device_java",
"../../rtc_base:base_java",
]
}
@@ -91,7 +90,6 @@
":surfaceviewrenderer_java",
":video_api_java",
":video_java",
- "//modules/audio_device:audio_device_java",
"//rtc_base:base_java",
]
}
@@ -156,6 +154,7 @@
sources = [
"api/org/webrtc/Predicate.java",
"api/org/webrtc/RefCounted.java",
+ "src/java/org/webrtc/ApplicationContextProvider.java",
"src/java/org/webrtc/CalledByNative.java",
"src/java/org/webrtc/CalledByNativeUnchecked.java",
"src/java/org/webrtc/Histogram.java",
@@ -165,7 +164,10 @@
"src/java/org/webrtc/WebRtcClassLoader.java",
]
- deps = [ "//third_party/androidx:androidx_annotation_annotation_java" ]
+ deps = [
+ "//rtc_base:base_java",
+ "//third_party/androidx:androidx_annotation_annotation_java",
+ ]
}
rtc_android_library("audio_api_java") {
@@ -317,7 +319,6 @@
":swcodecs_java",
":video_api_java",
":video_java",
- "//modules/audio_device:audio_device_java",
"//rtc_base:base_java",
"//third_party/androidx:androidx_annotation_annotation_java",
]
@@ -579,7 +580,6 @@
":internal_jni",
":native_api_jni",
"../../api:field_trials_view",
- "../../api:libjingle_peerconnection_api",
"../../api:scoped_refptr",
"../../api:sequence_checker",
"../../api/task_queue:pending_task_safety_flag",
@@ -931,6 +931,7 @@
rtc_library("native_api_jni") {
visibility = [ "*" ]
sources = [
+ "native_api/jni/application_context_provider.cc",
"native_api/jni/class_loader.cc",
"native_api/jni/java_types.cc",
"native_api/jni/jvm.cc",
@@ -939,6 +940,7 @@
]
public = [
+ "native_api/jni/application_context_provider.h",
"native_api/jni/class_loader.h",
"native_api/jni/java_types.h",
"native_api/jni/jni_int_wrapper.h",
@@ -984,10 +986,12 @@
deps = [
":base_jni",
+ ":internal_jni",
":java_audio_device_module",
+ ":native_api_jni",
":opensles_audio_device_module",
"../../api:scoped_refptr",
- "../../modules/audio_device",
+ "../../modules/audio_device:audio_device_api",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:refcount",
@@ -1197,7 +1201,7 @@
":base_jni",
":generated_java_audio_device_module_native_jni",
"../../api:sequence_checker",
- "../../modules/audio_device",
+ "../../modules/audio_device:audio_device_api",
"../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks",
"../../rtc_base:logging",
@@ -1255,7 +1259,7 @@
"../../api:refcountedbase",
"../../api:scoped_refptr",
"../../api:sequence_checker",
- "../../modules/audio_device",
+ "../../modules/audio_device:audio_device_api",
"../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks",
"../../rtc_base:logging",
@@ -1439,6 +1443,7 @@
generate_jni("generated_native_api_jni") {
sources = [
+ "src/java/org/webrtc/ApplicationContextProvider.java",
"src/java/org/webrtc/JniHelper.java",
"src/java/org/webrtc/WebRtcClassLoader.java",
]
@@ -1603,8 +1608,6 @@
sources = [
"native_unittests/android_network_monitor_unittest.cc",
- "native_unittests/application_context_provider.cc",
- "native_unittests/application_context_provider.h",
"native_unittests/audio_device/audio_device_unittest.cc",
"native_unittests/codecs/wrapper_unittest.cc",
"native_unittests/java_types_unittest.cc",
@@ -1676,7 +1679,6 @@
testonly = true
sources = [
- "native_unittests/org/webrtc/ApplicationContextProvider.java",
"native_unittests/org/webrtc/BuildInfo.java",
"native_unittests/org/webrtc/CodecsWrapperTestHelper.java",
"native_unittests/org/webrtc/FakeVideoEncoder.java",
@@ -1701,7 +1703,6 @@
testonly = true
sources = [
- "native_unittests/org/webrtc/ApplicationContextProvider.java",
"native_unittests/org/webrtc/BuildInfo.java",
"native_unittests/org/webrtc/CodecsWrapperTestHelper.java",
"native_unittests/org/webrtc/JavaTypesTestHelper.java",
diff --git a/sdk/android/native_api/DEPS b/sdk/android/native_api/DEPS
index 020e1cb..8afaebe 100644
--- a/sdk/android/native_api/DEPS
+++ b/sdk/android/native_api/DEPS
@@ -1,4 +1,5 @@
include_rules = [
"+modules/audio_device/include/audio_device.h",
+ "+modules/utility/include/jvm_android.h",
"+system_wrappers/include",
]
diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc
index 2be7f7d..6ba327a 100644
--- a/sdk/android/native_api/audio_device_module/audio_device_android.cc
+++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc
@@ -24,10 +24,12 @@
#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
#endif
+#include "sdk/android/native_api/jni/application_context_provider.h"
#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
#include "sdk/android/src/jni/audio_device/opensles_player.h"
#include "sdk/android/src/jni/audio_device/opensles_recorder.h"
+#include "sdk/android/src/jni/jvm.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@@ -70,6 +72,31 @@
std::make_unique<jni::AAudioRecorder>(input_parameters),
std::make_unique<jni::AAudioPlayer>(output_parameters));
}
+
+rtc::scoped_refptr<AudioDeviceModule>
+CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env,
+ jobject application_context) {
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
+ // Get default audio input/output parameters.
+ const JavaParamRef<jobject> j_context(application_context);
+ const ScopedJavaLocalRef<jobject> j_audio_manager =
+ jni::GetAudioManager(env, j_context);
+ AudioParameters input_parameters;
+ AudioParameters output_parameters;
+ GetDefaultAudioParameters(env, application_context, &input_parameters,
+ &output_parameters);
+ // Create ADM from AudioRecord and OpenSLESPlayer.
+ auto audio_input = std::make_unique<jni::AudioRecordJni>(
+ env, input_parameters, jni::kLowLatencyModeDelayEstimateInMilliseconds,
+ jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
+ j_audio_manager));
+
+ return CreateAudioDeviceModuleFromInputAndOutput(
+ AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio,
+ false /* use_stereo_input */, false /* use_stereo_output */,
+ jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
+ std::make_unique<jni::AAudioPlayer>(output_parameters));
+}
#endif
rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
@@ -152,4 +179,57 @@
std::move(audio_output));
}
+rtc::scoped_refptr<AudioDeviceModule> CreateAndroidAudioDeviceModule(
+ AudioDeviceModule::AudioLayer audio_layer) {
+ auto env = AttachCurrentThreadIfNeeded();
+ auto j_context = webrtc::GetAppContext(env);
+ // Select best possible combination of audio layers.
+ if (audio_layer == AudioDeviceModule::kPlatformDefaultAudio) {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+ // AAudio based audio for both input and output.
+ audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
+#else
+ if (jni::IsLowLatencyInputSupported(env, j_context) &&
+ jni::IsLowLatencyOutputSupported(env, j_context)) {
+ // Use OpenSL ES for both playout and recording.
+ audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
+ } else if (jni::IsLowLatencyOutputSupported(env, j_context) &&
+ !jni::IsLowLatencyInputSupported(env, j_context)) {
+ // Use OpenSL ES for output on devices that only supports the
+ // low-latency output audio path.
+ audio_layer = AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
+ } else {
+ // Use Java-based audio in both directions when low-latency output is
+ // not supported.
+ audio_layer = AudioDeviceModule::kAndroidJavaAudio;
+ }
+#endif
+ }
+ switch (audio_layer) {
+ case AudioDeviceModule::kAndroidJavaAudio:
+ // Java audio for both input and output audio.
+ return CreateJavaAudioDeviceModule(env, j_context.obj());
+ case AudioDeviceModule::kAndroidOpenSLESAudio:
+ // OpenSL ES based audio for both input and output audio.
+ return CreateOpenSLESAudioDeviceModule(env, j_context.obj());
+ case AudioDeviceModule::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.
+ return CreateJavaInputAndOpenSLESOutputAudioDeviceModule(
+ env, j_context.obj());
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+ case AudioDeviceModule::kAndroidAAudioAudio:
+ // AAudio based audio for both input and output.
+ return CreateAAudioAudioDeviceModule(env, j_context.obj());
+ case AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio:
+ // Java audio for input and AAudio for output audio (i.e. mixed APIs).
+ return CreateJavaInputAndAAudioOutputAudioDeviceModule(
+ env, j_context.obj());
+#endif
+ default:
+ return nullptr;
+ }
+}
+
} // namespace webrtc
diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.h b/sdk/android/native_api/audio_device_module/audio_device_android.h
index a093f8c..b687dca 100644
--- a/sdk/android/native_api/audio_device_module/audio_device_android.h
+++ b/sdk/android/native_api/audio_device_module/audio_device_android.h
@@ -32,8 +32,17 @@
jobject application_context);
rtc::scoped_refptr<AudioDeviceModule>
-CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
- jobject application_context);
+CreateJavaInputAndOpenSLESOutputAudioDeviceModule(
+ JNIEnv* env,
+ jobject application_context);
+
+rtc::scoped_refptr<AudioDeviceModule>
+CreateJavaInputAndAAudioOutputAudioDeviceModule(
+ JNIEnv* env,
+ jobject application_context);
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAndroidAudioDeviceModule(
+ AudioDeviceModule::AudioLayer audio_layer);
} // namespace webrtc
diff --git a/sdk/android/native_unittests/application_context_provider.cc b/sdk/android/native_api/jni/application_context_provider.cc
similarity index 63%
rename from sdk/android/native_unittests/application_context_provider.cc
rename to sdk/android/native_api/jni/application_context_provider.cc
index 07b3c04..de3c4a3 100644
--- a/sdk/android/native_unittests/application_context_provider.cc
+++ b/sdk/android/native_api/jni/application_context_provider.cc
@@ -7,18 +7,16 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "sdk/android/native_unittests/application_context_provider.h"
+#include "sdk/android/native_api/jni/application_context_provider.h"
-#include "sdk/android/generated_native_unittests_jni/ApplicationContextProvider_jni.h"
-#include "sdk/android/src/jni/jni_helpers.h"
+#include "sdk/android/generated_native_api_jni/ApplicationContextProvider_jni.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
namespace webrtc {
-namespace test {
-ScopedJavaLocalRef<jobject> GetAppContextForTest(JNIEnv* jni) {
+ScopedJavaLocalRef<jobject> GetAppContext(JNIEnv* jni) {
return ScopedJavaLocalRef<jobject>(
- jni::Java_ApplicationContextProvider_getApplicationContextForTest(jni));
+ jni::Java_ApplicationContextProvider_getApplicationContext(jni));
}
-} // namespace test
} // namespace webrtc
diff --git a/sdk/android/native_api/jni/application_context_provider.h b/sdk/android/native_api/jni/application_context_provider.h
new file mode 100644
index 0000000..dc3a80a
--- /dev/null
+++ b/sdk/android/native_api/jni/application_context_provider.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2019 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.
+ */
+#ifndef SDK_ANDROID_NATIVE_API_JNI_APPLICATION_CONTEXT_PROVIDER_H_
+#define SDK_ANDROID_NATIVE_API_JNI_APPLICATION_CONTEXT_PROVIDER_H_
+
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+
+namespace webrtc {
+
+ScopedJavaLocalRef<jobject> GetAppContext(JNIEnv* jni);
+
+} // namespace webrtc
+
+#endif // SDK_ANDROID_NATIVE_API_JNI_APPLICATION_CONTEXT_PROVIDER_H_
diff --git a/sdk/android/native_unittests/android_network_monitor_unittest.cc b/sdk/android/native_unittests/android_network_monitor_unittest.cc
index 9aec62d..0489cfd 100644
--- a/sdk/android/native_unittests/android_network_monitor_unittest.cc
+++ b/sdk/android/native_unittests/android_network_monitor_unittest.cc
@@ -13,7 +13,7 @@
#include "rtc_base/ip_address.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread.h"
-#include "sdk/android/native_unittests/application_context_provider.h"
+#include "sdk/android/native_api/jni/application_context_provider.h"
#include "sdk/android/src/jni/jni_helpers.h"
#include "test/gtest.h"
#include "test/scoped_key_value_config.h"
@@ -47,7 +47,7 @@
public:
AndroidNetworkMonitorTest() {
JNIEnv* env = AttachCurrentThreadIfNeeded();
- ScopedJavaLocalRef<jobject> context = test::GetAppContextForTest(env);
+ ScopedJavaLocalRef<jobject> context = GetAppContext(env);
network_monitor_ = std::make_unique<jni::AndroidNetworkMonitor>(
env, context, field_trials_);
}
diff --git a/sdk/android/native_unittests/application_context_provider.h b/sdk/android/native_unittests/application_context_provider.h
deleted file mode 100644
index 8aace02..0000000
--- a/sdk/android/native_unittests/application_context_provider.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright 2019 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.
- */
-#ifndef SDK_ANDROID_NATIVE_UNITTESTS_APPLICATION_CONTEXT_PROVIDER_H_
-#define SDK_ANDROID_NATIVE_UNITTESTS_APPLICATION_CONTEXT_PROVIDER_H_
-
-#include "sdk/android/src/jni/jni_helpers.h"
-
-namespace webrtc {
-namespace test {
-
-ScopedJavaLocalRef<jobject> GetAppContextForTest(JNIEnv* jni);
-
-} // namespace test
-} // namespace webrtc
-
-#endif // SDK_ANDROID_NATIVE_UNITTESTS_APPLICATION_CONTEXT_PROVIDER_H_
diff --git a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
index 7d582d4..201a54e 100644
--- a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
+++ b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
@@ -22,7 +22,7 @@
#include "rtc_base/time_utils.h"
#include "sdk/android/generated_native_unittests_jni/BuildInfo_jni.h"
#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
-#include "sdk/android/native_unittests/application_context_provider.h"
+#include "sdk/android/native_api/jni/application_context_provider.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/opensles_common.h"
@@ -466,7 +466,7 @@
// implementations.
// Creates an audio device using a default audio layer.
jni_ = AttachCurrentThreadIfNeeded();
- context_ = test::GetAppContextForTest(jni_);
+ context_ = GetAppContext(jni_);
audio_device_ = CreateJavaAudioDeviceModule(jni_, context_.obj());
EXPECT_NE(audio_device_.get(), nullptr);
EXPECT_EQ(0, audio_device_->Init());
@@ -491,7 +491,7 @@
}
void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer) {
- audio_device_ = CreateAudioDevice(audio_layer);
+ audio_device_ = CreateAndroidAudioDeviceModule(audio_layer);
EXPECT_NE(audio_device_.get(), nullptr);
EXPECT_EQ(0, audio_device_->Init());
UpdateParameters();
@@ -512,30 +512,6 @@
return audio_device_;
}
- rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
- AudioDeviceModule::AudioLayer audio_layer) {
-#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
- if (audio_layer == AudioDeviceModule::kAndroidAAudioAudio) {
- return rtc::scoped_refptr<AudioDeviceModule>(
- CreateAAudioAudioDeviceModule(jni_, context_.obj()));
- }
-#endif
- if (audio_layer == AudioDeviceModule::kAndroidJavaAudio) {
- return rtc::scoped_refptr<AudioDeviceModule>(
- CreateJavaAudioDeviceModule(jni_, context_.obj()));
- } else if (audio_layer == AudioDeviceModule::kAndroidOpenSLESAudio) {
- return rtc::scoped_refptr<AudioDeviceModule>(
- CreateOpenSLESAudioDeviceModule(jni_, context_.obj()));
- } else if (audio_layer ==
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
- return rtc::scoped_refptr<AudioDeviceModule>(
- CreateJavaInputAndOpenSLESOutputAudioDeviceModule(jni_,
- context_.obj()));
- } else {
- return nullptr;
- }
- }
-
// Returns file name relative to the resource root given a sample rate.
std::string GetFileName(int sample_rate) {
EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100);
@@ -566,7 +542,7 @@
int TestDelayOnAudioLayer(
const AudioDeviceModule::AudioLayer& layer_to_test) {
rtc::scoped_refptr<AudioDeviceModule> audio_device;
- audio_device = CreateAudioDevice(layer_to_test);
+ audio_device = CreateAndroidAudioDeviceModule(layer_to_test);
EXPECT_NE(audio_device.get(), nullptr);
uint16_t playout_delay;
EXPECT_EQ(0, audio_device->PlayoutDelay(&playout_delay));
@@ -576,7 +552,7 @@
AudioDeviceModule::AudioLayer TestActiveAudioLayer(
const AudioDeviceModule::AudioLayer& layer_to_test) {
rtc::scoped_refptr<AudioDeviceModule> audio_device;
- audio_device = CreateAudioDevice(layer_to_test);
+ audio_device = CreateAndroidAudioDeviceModule(layer_to_test);
EXPECT_NE(audio_device.get(), nullptr);
AudioDeviceModule::AudioLayer active;
EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active));
@@ -674,6 +650,22 @@
return volume;
}
+ bool IsLowLatencyPlayoutSupported() {
+ return jni::IsLowLatencyInputSupported(jni_, context_);
+ }
+
+ bool IsLowLatencyRecordSupported() {
+ return jni::IsLowLatencyOutputSupported(jni_, context_);
+ }
+
+ bool IsAAudioSupported() {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+ return true;
+#else
+ return false;
+#endif
+ }
+
JNIEnv* jni_;
ScopedJavaLocalRef<jobject> context_;
rtc::Event test_is_done_;
@@ -687,6 +679,31 @@
// Using the test fixture to create and destruct the audio device module.
}
+// We always ask for a default audio layer when the ADM is constructed. But the
+// ADM will then internally set the best suitable combination of audio layers,
+// for input and output based on if low-latency output and/or input audio in
+// combination with OpenSL ES is supported or not. This test ensures that the
+// correct selection is done.
+TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
+ const AudioDeviceModule::AudioLayer audio_layer =
+ TestActiveAudioLayer(AudioDeviceModule::kPlatformDefaultAudio);
+ bool low_latency_output = IsLowLatencyPlayoutSupported();
+ bool low_latency_input = IsLowLatencyRecordSupported();
+ bool aaudio = IsAAudioSupported();
+ AudioDeviceModule::AudioLayer expected_audio_layer;
+ if (aaudio) {
+ expected_audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
+ } else if (low_latency_output && low_latency_input) {
+ expected_audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
+ } else if (low_latency_output && !low_latency_input) {
+ expected_audio_layer =
+ AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
+ } else {
+ expected_audio_layer = AudioDeviceModule::kAndroidJavaAudio;
+ }
+ EXPECT_EQ(expected_audio_layer, audio_layer);
+}
+
// Verify that it is possible to explicitly create the two types of supported
// ADMs. These two tests overrides the default selection of native audio layer
// by ignoring if the device supports low-latency output or not.
@@ -714,15 +731,18 @@
EXPECT_EQ(expected_layer, active_layer);
}
-// TODO(bugs.webrtc.org/8914)
-// TODO(phensman): Add test for AAudio/Java combination when this combination
-// is supported.
#if !defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
DISABLED_CorrectAudioLayerIsUsedForAAudioInBothDirections
+
+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
+ DISABLED_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
#else
#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
CorrectAudioLayerIsUsedForAAudioInBothDirections
+
+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
+ CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
#endif
TEST_F(AudioDeviceTest,
MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections) {
@@ -733,6 +753,15 @@
EXPECT_EQ(expected_layer, active_layer);
}
+TEST_F(AudioDeviceTest,
+ MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo) {
+ AudioDeviceModule::AudioLayer expected_layer =
+ AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio;
+ AudioDeviceModule::AudioLayer active_layer =
+ TestActiveAudioLayer(expected_layer);
+ EXPECT_EQ(expected_layer, active_layer);
+}
+
// The Android ADM supports two different delay reporting modes. One for the
// low-latency output path (in combination with OpenSL ES), and one for the
// high-latency output path (Java backends in both directions). These two tests
@@ -1129,7 +1158,7 @@
TEST(JavaAudioDeviceTest, TestRunningTwoAdmsSimultaneously) {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
- ScopedJavaLocalRef<jobject> context = test::GetAppContextForTest(jni);
+ ScopedJavaLocalRef<jobject> context = GetAppContext(jni);
// Create and start the first ADM.
rtc::scoped_refptr<AudioDeviceModule> adm_1 =
diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
index 8bb6e33..b751390 100644
--- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
+++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
@@ -24,7 +24,7 @@
#include "sdk/android/generated_native_unittests_jni/PeerConnectionFactoryInitializationHelper_jni.h"
#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
#include "sdk/android/native_api/jni/jvm.h"
-#include "sdk/android/native_unittests/application_context_provider.h"
+#include "sdk/android/native_api/jni/application_context_provider.h"
#include "sdk/android/src/jni/jni_helpers.h"
#include "test/gtest.h"
@@ -57,7 +57,7 @@
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
media_deps.adm =
- CreateJavaAudioDeviceModule(jni, GetAppContextForTest(jni).obj());
+ CreateJavaAudioDeviceModule(jni, GetAppContext(jni).obj());
media_deps.video_encoder_factory =
std::make_unique<webrtc::InternalEncoderFactory>();
media_deps.video_decoder_factory =
diff --git a/sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java b/sdk/android/src/java/org/webrtc/ApplicationContextProvider.java
similarity index 90%
rename from sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java
rename to sdk/android/src/java/org/webrtc/ApplicationContextProvider.java
index e10d347..6400a04 100644
--- a/sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java
+++ b/sdk/android/src/java/org/webrtc/ApplicationContextProvider.java
@@ -14,7 +14,7 @@
public class ApplicationContextProvider {
@CalledByNative
- public static Context getApplicationContextForTest() {
+ public static Context getApplicationContext() {
return ContextUtils.getApplicationContext();
}
}
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
index f398602..506e33f 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
@@ -55,11 +55,13 @@
: getMinInputFrameSize(sampleRate, numberOfInputChannels);
}
- private static boolean isLowLatencyOutputSupported(Context context) {
+ @CalledByNative
+ static boolean isLowLatencyOutputSupported(Context context) {
return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
}
- private static boolean isLowLatencyInputSupported(Context context) {
+ @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
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.cc b/sdk/android/src/jni/audio_device/audio_device_module.cc
index 7c59d3e..3742d89 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.cc
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@@ -633,6 +633,14 @@
RTC_CHECK(output_parameters->is_valid());
}
+bool IsLowLatencyInputSupported(JNIEnv* env, const JavaRef<jobject>& j_context) {
+ return Java_WebRtcAudioManager_isLowLatencyInputSupported(env, j_context);
+}
+
+bool IsLowLatencyOutputSupported(JNIEnv* env, const JavaRef<jobject>& j_context) {
+ return Java_WebRtcAudioManager_isLowLatencyOutputSupported(env, j_context);
+}
+
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::AudioLayer audio_layer,
bool is_stereo_playout_supported,
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.h b/sdk/android/src/jni/audio_device/audio_device_module.h
index 1918336..9ec73de 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.h
+++ b/sdk/android/src/jni/audio_device/audio_device_module.h
@@ -86,6 +86,10 @@
AudioParameters* input_parameters,
AudioParameters* output_parameters);
+bool IsLowLatencyInputSupported(JNIEnv* env, const JavaRef<jobject>& j_context);
+
+bool IsLowLatencyOutputSupported(JNIEnv* env, const JavaRef<jobject>& j_context);
+
// Glue together an audio input and audio output to get an AudioDeviceModule.
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::AudioLayer audio_layer,
