Reland "Remove old audio device implementation."
This reverts commit e9d2b4efdd5dddaa3a476c0ac2a9cf9125b39929.
Reason for revert: Fixing build break of downstream project.
Original change's description:
> Revert "Remove old audio device implementation."
>
> This reverts commit 0cfa4cba5cae5e942f5d8e0e4e93b94982d0bfc3.
>
> Reason for revert: audio_device_ios_objc target is removed, but still referenced by iPhone Meetins:Meeting_build_test, which now fails to build
>
> Original change's description:
> > Remove old audio device implementation.
> >
> > The iOS ADM implementation now lives in sdk/objc/native/api/audio_device_module.{h,mm}.
> >
> > Bug: webrtc:10514
> > Change-Id: Ib0b162027b5680ebc40d621a57f1155f08e7a057
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131326
> > Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27488}
>
> TBR=henrika@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I5be10b3d17403a79ea30afc255cde01171bc9f5b
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10514
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131960
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27492}
TBR=henrika@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org,jeroendb@webrtc.org
Change-Id: Ic05131d902f9623bd5ae7635d3c71880ffd3c6d3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10514
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131944
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27533}diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 266972b..c42f862 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -45,47 +45,6 @@
]
}
-if (rtc_include_internal_audio_device && is_ios) {
- rtc_source_set("audio_device_ios_objc") {
- visibility = [
- ":audio_device_impl",
- ":audio_device_ios_objc_unittests",
- ]
- sources = [
- "ios/audio_device_ios.h",
- "ios/audio_device_ios.mm",
- "ios/audio_device_not_implemented_ios.mm",
- "ios/audio_session_observer.h",
- "ios/objc/RTCAudioSession.h",
- "ios/objc/RTCAudioSessionConfiguration.h",
- "ios/objc/RTCAudioSessionDelegateAdapter.h",
- "ios/objc/RTCAudioSessionDelegateAdapter.mm",
- "ios/voice_processing_audio_unit.h",
- "ios/voice_processing_audio_unit.mm",
- ]
- libs = [
- "AudioToolbox.framework",
- "AVFoundation.framework",
- "Foundation.framework",
- "UIKit.framework",
- ]
- deps = [
- ":audio_device_api",
- ":audio_device_buffer",
- ":audio_device_generic",
- "../../api:array_view",
- "../../rtc_base",
- "../../rtc_base:checks",
- "../../rtc_base:gtest_prod",
- "../../rtc_base/system:fallthrough",
- "../../sdk:audio_device",
- "../../sdk:audio_objc",
- "../../sdk:base_objc",
- "../../system_wrappers:metrics",
- ]
- }
-}
-
rtc_source_set("audio_device_api") {
visibility = [ "*" ]
sources = [
@@ -225,7 +184,7 @@
"//third_party/abseil-cpp/absl/memory",
]
if (rtc_include_internal_audio_device && is_ios) {
- deps += [ ":audio_device_ios_objc" ]
+ deps += [ "../../sdk:audio_device" ]
}
sources = [
@@ -398,32 +357,6 @@
}
if (rtc_include_tests) {
- # TODO(kthelgason): Reenable these tests on simulator.
- # See bugs.webrtc.org/7812
- if (rtc_include_internal_audio_device && is_ios && !use_ios_simulator) {
- rtc_source_set("audio_device_ios_objc_unittests") {
- testonly = true
- visibility = [ ":*" ]
- sources = [
- "ios/audio_device_unittest_ios.mm",
- ]
- deps = [
- ":audio_device",
- ":audio_device_buffer",
- ":audio_device_impl",
- ":audio_device_ios_objc",
- ":mock_audio_device",
- "../../api:scoped_refptr",
- "../../rtc_base:rtc_base_approved",
- "../../sdk:audio_objc",
- "../../system_wrappers",
- "../../test:fileutils",
- "../../test:test_support",
- "//third_party/ocmock",
- ]
- }
- }
-
rtc_source_set("audio_device_unittests") {
testonly = true
diff --git a/modules/audio_device/DEPS b/modules/audio_device/DEPS
index f74767a..fc5eed7 100644
--- a/modules/audio_device/DEPS
+++ b/modules/audio_device/DEPS
@@ -7,28 +7,7 @@
"ensure_initialized\.cc": [
"+base/android",
],
- "audio_device_ios\.h": [
- "+sdk/objc",
- ],
- "audio_device_ios\.mm": [
- "+sdk/objc",
- ],
- "audio_device_unittest_ios\.mm": [
- "+sdk/objc",
- ],
- "RTCAudioSession\.h": [
- "+sdk/objc",
- ],
- "RTCAudioSessionConfiguration\.h": [
- "+sdk/objc",
- ],
- "RTCAudioSessionDelegateAdapter\.h": [
- "+sdk/objc",
- ],
- "RTCAudioSessionDelegateAdapter\.mm": [
- "+sdk/objc",
- ],
- "voice_processing_audio_unit\.mm": [
+ "audio_device_impl\.cc": [
"+sdk/objc",
],
}
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index 7b08a5a..f7efced 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -45,7 +45,7 @@
#include "modules/audio_device/linux/audio_device_pulse_linux.h"
#endif
#elif defined(WEBRTC_IOS)
-#include "modules/audio_device/ios/audio_device_ios.h"
+#include "sdk/objc/native/src/audio/audio_device_ios.h"
#elif defined(WEBRTC_MAC)
#include "modules/audio_device/mac/audio_device_mac.h"
#endif
@@ -287,7 +287,7 @@
// iOS ADM implementation.
#if defined(WEBRTC_IOS)
if (audio_layer == kPlatformDefaultAudio) {
- audio_device_.reset(new AudioDeviceIOS());
+ audio_device_.reset(new ios_adm::AudioDeviceIOS());
RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
}
// END #if defined(WEBRTC_IOS)
diff --git a/modules/audio_device/ios/audio_device_ios.h b/modules/audio_device/ios/audio_device_ios.h
deleted file mode 100644
index e90bb44..0000000
--- a/modules/audio_device/ios/audio_device_ios.h
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 2012 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_IOS_AUDIO_DEVICE_IOS_H_
-#define MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
-
-#include <memory>
-
-#include "modules/audio_device/audio_device_generic.h"
-#include "modules/audio_device/ios/audio_session_observer.h"
-#include "modules/audio_device/ios/voice_processing_audio_unit.h"
-#include "rtc_base/buffer.h"
-#include "rtc_base/gtest_prod_util.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
-#include "rtc_base/thread_checker.h"
-#include "sdk/objc/base/RTCMacros.h"
-
-RTC_FWD_DECL_OBJC_CLASS(RTCAudioSessionDelegateAdapter);
-
-namespace webrtc {
-
-class FineAudioBuffer;
-
-// Implements full duplex 16-bit mono PCM audio support for iOS using a
-// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
-// supports audio echo cancellation. It also adds automatic gain control,
-// adjustment of voice-processing quality and muting.
-//
-// An instance must be created and destroyed on one and the same thread.
-// All supported public methods must also be called on the same thread.
-// A thread checker will RTC_DCHECK if any supported method is called on an
-// invalid thread.
-//
-// Recorded audio will be delivered on a real-time internal I/O thread in the
-// audio unit. The audio unit will also ask for audio data to play out on this
-// same thread.
-class AudioDeviceIOS : public AudioDeviceGeneric,
- public AudioSessionObserver,
- public VoiceProcessingAudioUnitObserver,
- public rtc::MessageHandler {
- public:
- AudioDeviceIOS();
- ~AudioDeviceIOS();
-
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
-
- InitStatus Init() override;
- int32_t Terminate() override;
- bool Initialized() const override;
-
- int32_t InitPlayout() override;
- bool PlayoutIsInitialized() const override;
-
- int32_t InitRecording() override;
- bool RecordingIsInitialized() const override;
-
- int32_t StartPlayout() override;
- int32_t StopPlayout() override;
- bool Playing() const override { return playing_; }
-
- int32_t StartRecording() override;
- int32_t StopRecording() override;
- bool Recording() const override { return recording_; }
-
- // These methods returns hard-coded delay values and not dynamic delay
- // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
- // AEC will always be disabled in the Libjingle layer to avoid running two
- // AEC implementations at the same time. And, it saves resources to avoid
- // updating these delay values continuously.
- // TODO(henrika): it would be possible to mark these two methods as not
- // implemented since they are only called for A/V-sync purposes today and
- // A/V-sync is not supported on iOS. However, we avoid adding error messages
- // the log by using these dummy implementations instead.
- int32_t PlayoutDelay(uint16_t& delayMS) const override;
-
- // Native audio parameters stored during construction.
- // These methods are unique for the iOS implementation.
- int GetPlayoutAudioParameters(AudioParameters* params) const override;
- int GetRecordAudioParameters(AudioParameters* params) const override;
-
- // These methods are currently not fully implemented on iOS:
-
- // See audio_device_not_implemented.cc for trivial implementations.
- int32_t ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const override;
- int32_t PlayoutIsAvailable(bool& available) override;
- int32_t RecordingIsAvailable(bool& available) override;
- int16_t PlayoutDevices() override;
- int16_t RecordingDevices() override;
- int32_t PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override;
- int32_t RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override;
- int32_t SetPlayoutDevice(uint16_t index) override;
- int32_t SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device) override;
- int32_t SetRecordingDevice(uint16_t index) override;
- int32_t SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device) override;
- int32_t InitSpeaker() override;
- bool SpeakerIsInitialized() const override;
- int32_t InitMicrophone() override;
- bool MicrophoneIsInitialized() const override;
- int32_t SpeakerVolumeIsAvailable(bool& available) override;
- int32_t SetSpeakerVolume(uint32_t volume) override;
- int32_t SpeakerVolume(uint32_t& volume) const override;
- int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
- int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
- int32_t MicrophoneVolumeIsAvailable(bool& available) override;
- int32_t SetMicrophoneVolume(uint32_t volume) override;
- int32_t MicrophoneVolume(uint32_t& volume) const override;
- int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
- int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
- int32_t MicrophoneMuteIsAvailable(bool& available) override;
- int32_t SetMicrophoneMute(bool enable) override;
- int32_t MicrophoneMute(bool& enabled) const override;
- int32_t SpeakerMuteIsAvailable(bool& available) override;
- int32_t SetSpeakerMute(bool enable) override;
- int32_t SpeakerMute(bool& enabled) const override;
- int32_t StereoPlayoutIsAvailable(bool& available) override;
- int32_t SetStereoPlayout(bool enable) override;
- int32_t StereoPlayout(bool& enabled) const override;
- int32_t StereoRecordingIsAvailable(bool& available) override;
- int32_t SetStereoRecording(bool enable) override;
- int32_t StereoRecording(bool& enabled) const override;
-
- // AudioSessionObserver methods. May be called from any thread.
- void OnInterruptionBegin() override;
- void OnInterruptionEnd() override;
- void OnValidRouteChange() override;
- void OnCanPlayOrRecordChange(bool can_play_or_record) override;
- void OnChangedOutputVolume() override;
-
- // VoiceProcessingAudioUnitObserver methods.
- OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) override;
- OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) override;
-
- // Handles messages from posts.
- void OnMessage(rtc::Message* msg) override;
-
- private:
- // Called by the relevant AudioSessionObserver methods on |thread_|.
- void HandleInterruptionBegin();
- void HandleInterruptionEnd();
- void HandleValidRouteChange();
- void HandleCanPlayOrRecordChange(bool can_play_or_record);
- void HandleSampleRateChange(float sample_rate);
- void HandlePlayoutGlitchDetected();
- void HandleOutputVolumeChange();
-
- // Uses current |playout_parameters_| and |record_parameters_| to inform the
- // audio device buffer (ADB) about our internal audio parameters.
- void UpdateAudioDeviceBuffer();
-
- // Since the preferred audio parameters are only hints to the OS, the actual
- // values may be different once the AVAudioSession has been activated.
- // This method asks for the current hardware parameters and takes actions
- // if they should differ from what we have asked for initially. It also
- // defines |playout_parameters_| and |record_parameters_|.
- void SetupAudioBuffersForActiveAudioSession();
-
- // Creates the audio unit.
- bool CreateAudioUnit();
-
- // Updates the audio unit state based on current state.
- void UpdateAudioUnit(bool can_play_or_record);
-
- // Configures the audio session for WebRTC.
- bool ConfigureAudioSession();
- // Unconfigures the audio session.
- void UnconfigureAudioSession();
-
- // Activates our audio session, creates and initializes the voice-processing
- // audio unit and verifies that we got the preferred native audio parameters.
- bool InitPlayOrRecord();
-
- // Closes and deletes the voice-processing I/O unit.
- void ShutdownPlayOrRecord();
-
- // Resets thread-checkers before a call is restarted.
- void PrepareForNewStart();
-
- // Ensures that methods are called from the same thread as this object is
- // created on.
- rtc::ThreadChecker thread_checker_;
-
- // Native I/O audio thread checker.
- rtc::ThreadChecker io_thread_checker_;
-
- // Thread that this object is created on.
- rtc::Thread* thread_;
-
- // 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_;
-
- // Contains audio parameters (sample rate, #channels, buffer size etc.) for
- // the playout and recording sides. These structure is set in two steps:
- // first, native sample rate and #channels are defined in Init(). Next, the
- // audio session is activated and we verify that the preferred parameters
- // were granted by the OS. At this stage it is also possible to add a third
- // component to the parameters; the native I/O buffer duration.
- // A RTC_CHECK will be hit if we for some reason fail to open an audio session
- // using the specified parameters.
- AudioParameters playout_parameters_;
- AudioParameters record_parameters_;
-
- // The AudioUnit used to play and record audio.
- std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_;
-
- // 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. Is also supports a similar scheme for the recording
- // side.
- // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
- // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
- // in each callback (one every 8ms). This class can then ask for 128 and the
- // FineAudioBuffer will ask WebRTC for new data only when needed and also
- // cache non-utilized audio between callbacks. On the recording side, iOS
- // can provide audio data frames of size 128 and these are accumulated until
- // enough data to supply one 10ms call exists. This 10ms chunk is then sent
- // to WebRTC and the remaining part is stored.
- std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
-
- // Temporary storage for recorded data. AudioUnitRender() renders into this
- // array as soon as a frame of the desired buffer size has been recorded.
- // On real iOS devices, the size will be fixed and set once. For iOS
- // simulators, the size can vary from callback to callback and the size
- // will be changed dynamically to account for this behavior.
- rtc::BufferT<int16_t> record_audio_buffer_;
-
- // Set to 1 when recording is active and 0 otherwise.
- volatile int recording_;
-
- // Set to 1 when playout is active and 0 otherwise.
- volatile int playing_;
-
- // Set to true after successful call to Init(), false otherwise.
- bool initialized_ RTC_GUARDED_BY(thread_checker_);
-
- // Set to true after successful call to InitRecording() or InitPlayout(),
- // false otherwise.
- bool audio_is_initialized_;
-
- // Set to true if audio session is interrupted, false otherwise.
- bool is_interrupted_;
-
- // Audio interruption observer instance.
- RTCAudioSessionDelegateAdapter* audio_session_observer_
- RTC_GUARDED_BY(thread_checker_);
-
- // Set to true if we've activated the audio session.
- bool has_configured_session_ RTC_GUARDED_BY(thread_checker_);
-
- // Counts number of detected audio glitches on the playout side.
- int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_checker_);
- int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_);
-
- // Counts number of playout callbacks per call.
- // The value isupdated on the native I/O thread and later read on the
- // creating thread (see thread_checker_) but at this stage no audio is
- // active. Hence, it is a "thread safe" design and no lock is needed.
- int64_t num_playout_callbacks_;
-
- // Contains the time for when the last output volume change was detected.
- int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_checker_);
-
- // Exposes private members for testing purposes only.
- FRIEND_TEST_ALL_PREFIXES(AudioDeviceTest, testInterruptedAudioSession);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
diff --git a/modules/audio_device/ios/audio_device_ios.mm b/modules/audio_device/ios/audio_device_ios.mm
deleted file mode 100644
index 2c3140a..0000000
--- a/modules/audio_device/ios/audio_device_ios.mm
+++ /dev/null
@@ -1,908 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-#import <AVFoundation/AVFoundation.h>
-#import <Foundation/Foundation.h>
-
-#include "modules/audio_device/ios/audio_device_ios.h"
-
-#include <cmath>
-
-#include "api/array_view.h"
-#include "modules/audio_device/fine_audio_buffer.h"
-#include "rtc_base/atomic_ops.h"
-#include "rtc_base/bind.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
-#include "rtc_base/time_utils.h"
-#include "sdk/objc/native/src/audio/helpers.h"
-#include "system_wrappers/include/metrics.h"
-
-#import "modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
-#import "sdk/objc/base/RTCLogging.h"
-#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
-#import "sdk/objc/components/audio/RTCAudioSession.h"
-#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
-
-namespace webrtc {
-
-#define LOGI() RTC_LOG(LS_INFO) << "AudioDeviceIOS::"
-
-#define LOG_AND_RETURN_IF_ERROR(error, message) \
- do { \
- OSStatus err = error; \
- if (err) { \
- RTC_LOG(LS_ERROR) << message << ": " << err; \
- return false; \
- } \
- } while (0)
-
-#define LOG_IF_ERROR(error, message) \
- do { \
- OSStatus err = error; \
- if (err) { \
- RTC_LOG(LS_ERROR) << message << ": " << err; \
- } \
- } while (0)
-
-// Hardcoded delay estimates based on real measurements.
-// TODO(henrika): these value is not used in combination with built-in AEC.
-// Can most likely be removed.
-const UInt16 kFixedPlayoutDelayEstimate = 30;
-const UInt16 kFixedRecordDelayEstimate = 30;
-
-enum AudioDeviceMessageType : uint32_t {
- kMessageTypeInterruptionBegin,
- kMessageTypeInterruptionEnd,
- kMessageTypeValidRouteChange,
- kMessageTypeCanPlayOrRecordChange,
- kMessageTypePlayoutGlitchDetected,
- kMessageOutputVolumeChange,
-};
-
-using ios::CheckAndLogError;
-
-#if !defined(NDEBUG)
-// Returns true when the code runs on a device simulator.
-static bool DeviceIsSimulator() {
- return ios::GetDeviceName() == "x86_64";
-}
-
-// Helper method that logs essential device information strings.
-static void LogDeviceInfo() {
- RTC_LOG(LS_INFO) << "LogDeviceInfo";
- @autoreleasepool {
- RTC_LOG(LS_INFO) << " system name: " << ios::GetSystemName();
- RTC_LOG(LS_INFO) << " system version: " << ios::GetSystemVersionAsString();
- RTC_LOG(LS_INFO) << " device type: " << ios::GetDeviceType();
- RTC_LOG(LS_INFO) << " device name: " << ios::GetDeviceName();
- RTC_LOG(LS_INFO) << " process name: " << ios::GetProcessName();
- RTC_LOG(LS_INFO) << " process ID: " << ios::GetProcessID();
- RTC_LOG(LS_INFO) << " OS version: " << ios::GetOSVersionString();
- RTC_LOG(LS_INFO) << " processing cores: " << ios::GetProcessorCount();
- RTC_LOG(LS_INFO) << " low power mode: " << ios::GetLowPowerModeEnabled();
-#if TARGET_IPHONE_SIMULATOR
- RTC_LOG(LS_INFO) << " TARGET_IPHONE_SIMULATOR is defined";
-#endif
- RTC_LOG(LS_INFO) << " DeviceIsSimulator: " << DeviceIsSimulator();
- }
-}
-#endif // !defined(NDEBUG)
-
-AudioDeviceIOS::AudioDeviceIOS()
- : audio_device_buffer_(nullptr),
- audio_unit_(nullptr),
- recording_(0),
- playing_(0),
- initialized_(false),
- audio_is_initialized_(false),
- is_interrupted_(false),
- has_configured_session_(false),
- num_detected_playout_glitches_(0),
- last_playout_time_(0),
- num_playout_callbacks_(0),
- last_output_volume_change_time_(0) {
- LOGI() << "ctor" << ios::GetCurrentThreadDescription();
- io_thread_checker_.Detach();
- thread_ = rtc::Thread::Current();
- audio_session_observer_ = [[RTCAudioSessionDelegateAdapter alloc] initWithObserver:this];
-}
-
-AudioDeviceIOS::~AudioDeviceIOS() {
- LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
- audio_session_observer_ = nil;
- RTC_DCHECK(thread_checker_.IsCurrent());
- Terminate();
-}
-
-void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- LOGI() << "AttachAudioBuffer";
- RTC_DCHECK(audioBuffer);
- RTC_DCHECK(thread_checker_.IsCurrent());
- audio_device_buffer_ = audioBuffer;
-}
-
-AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
- LOGI() << "Init";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- if (initialized_) {
- return InitStatus::OK;
- }
-#if !defined(NDEBUG)
- LogDeviceInfo();
-#endif
- // Store the preferred sample rate and preferred number of channels already
- // here. They have not been set and confirmed yet since configureForWebRTC
- // is not called until audio is about to start. However, it makes sense to
- // store the parameters now and then verify at a later stage.
- RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
- playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
- record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
- // Ensure that the audio device buffer (ADB) knows about the internal audio
- // parameters. Note that, even if we are unable to get a mono audio session,
- // we will always tell the I/O audio unit to do a channel format conversion
- // to guarantee mono on the "input side" of the audio unit.
- UpdateAudioDeviceBuffer();
- initialized_ = true;
- return InitStatus::OK;
-}
-
-int32_t AudioDeviceIOS::Terminate() {
- LOGI() << "Terminate";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- if (!initialized_) {
- return 0;
- }
- StopPlayout();
- StopRecording();
- initialized_ = false;
- return 0;
-}
-
-bool AudioDeviceIOS::Initialized() const {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- return initialized_;
-}
-
-int32_t AudioDeviceIOS::InitPlayout() {
- LOGI() << "InitPlayout";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_DCHECK(initialized_);
- RTC_DCHECK(!audio_is_initialized_);
- RTC_DCHECK(!playing_);
- if (!audio_is_initialized_) {
- if (!InitPlayOrRecord()) {
- RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
- return -1;
- }
- }
- audio_is_initialized_ = true;
- return 0;
-}
-
-bool AudioDeviceIOS::PlayoutIsInitialized() const {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- return audio_is_initialized_;
-}
-
-bool AudioDeviceIOS::RecordingIsInitialized() const {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- return audio_is_initialized_;
-}
-
-int32_t AudioDeviceIOS::InitRecording() {
- LOGI() << "InitRecording";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_DCHECK(initialized_);
- RTC_DCHECK(!audio_is_initialized_);
- RTC_DCHECK(!recording_);
- if (!audio_is_initialized_) {
- if (!InitPlayOrRecord()) {
- RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
- return -1;
- }
- }
- audio_is_initialized_ = true;
- return 0;
-}
-
-int32_t AudioDeviceIOS::StartPlayout() {
- LOGI() << "StartPlayout";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_DCHECK(audio_is_initialized_);
- RTC_DCHECK(!playing_);
- RTC_DCHECK(audio_unit_);
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetPlayout();
- }
- if (!recording_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
- if (!audio_unit_->Start()) {
- RTCLogError(@"StartPlayout failed to start audio unit.");
- return -1;
- }
- RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
- }
- rtc::AtomicOps::ReleaseStore(&playing_, 1);
- num_playout_callbacks_ = 0;
- num_detected_playout_glitches_ = 0;
- return 0;
-}
-
-int32_t AudioDeviceIOS::StopPlayout() {
- LOGI() << "StopPlayout";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- if (!audio_is_initialized_ || !playing_) {
- return 0;
- }
- if (!recording_) {
- ShutdownPlayOrRecord();
- audio_is_initialized_ = false;
- }
- rtc::AtomicOps::ReleaseStore(&playing_, 0);
-
- // Derive average number of calls to OnGetPlayoutData() between detected
- // audio glitches and add the result to a histogram.
- int average_number_of_playout_callbacks_between_glitches = 100000;
- RTC_DCHECK_GE(num_playout_callbacks_, num_detected_playout_glitches_);
- if (num_detected_playout_glitches_ > 0) {
- average_number_of_playout_callbacks_between_glitches =
- num_playout_callbacks_ / num_detected_playout_glitches_;
- }
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
- average_number_of_playout_callbacks_between_glitches);
- RTCLog(@"Average number of playout callbacks between glitches: %d",
- average_number_of_playout_callbacks_between_glitches);
- return 0;
-}
-
-int32_t AudioDeviceIOS::StartRecording() {
- LOGI() << "StartRecording";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_DCHECK(audio_is_initialized_);
- RTC_DCHECK(!recording_);
- RTC_DCHECK(audio_unit_);
- if (fine_audio_buffer_) {
- fine_audio_buffer_->ResetRecord();
- }
- if (!playing_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
- if (!audio_unit_->Start()) {
- RTCLogError(@"StartRecording failed to start audio unit.");
- return -1;
- }
- RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
- }
- rtc::AtomicOps::ReleaseStore(&recording_, 1);
- return 0;
-}
-
-int32_t AudioDeviceIOS::StopRecording() {
- LOGI() << "StopRecording";
- RTC_DCHECK_RUN_ON(&thread_checker_);
- if (!audio_is_initialized_ || !recording_) {
- return 0;
- }
- if (!playing_) {
- ShutdownPlayOrRecord();
- audio_is_initialized_ = false;
- }
- rtc::AtomicOps::ReleaseStore(&recording_, 0);
- return 0;
-}
-
-int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
- delayMS = kFixedPlayoutDelayEstimate;
- return 0;
-}
-
-int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
- LOGI() << "GetPlayoutAudioParameters";
- RTC_DCHECK(playout_parameters_.is_valid());
- RTC_DCHECK(thread_checker_.IsCurrent());
- *params = playout_parameters_;
- return 0;
-}
-
-int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
- LOGI() << "GetRecordAudioParameters";
- RTC_DCHECK(record_parameters_.is_valid());
- RTC_DCHECK(thread_checker_.IsCurrent());
- *params = record_parameters_;
- return 0;
-}
-
-void AudioDeviceIOS::OnInterruptionBegin() {
- RTC_DCHECK(thread_);
- LOGI() << "OnInterruptionBegin";
- thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionBegin);
-}
-
-void AudioDeviceIOS::OnInterruptionEnd() {
- RTC_DCHECK(thread_);
- LOGI() << "OnInterruptionEnd";
- thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionEnd);
-}
-
-void AudioDeviceIOS::OnValidRouteChange() {
- RTC_DCHECK(thread_);
- thread_->Post(RTC_FROM_HERE, this, kMessageTypeValidRouteChange);
-}
-
-void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
- RTC_DCHECK(thread_);
- thread_->Post(RTC_FROM_HERE,
- this,
- kMessageTypeCanPlayOrRecordChange,
- new rtc::TypedMessageData<bool>(can_play_or_record));
-}
-
-void AudioDeviceIOS::OnChangedOutputVolume() {
- RTC_DCHECK(thread_);
- thread_->Post(RTC_FROM_HERE, this, kMessageOutputVolumeChange);
-}
-
-OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* /* io_data */) {
- RTC_DCHECK_RUN_ON(&io_thread_checker_);
- OSStatus result = noErr;
- // Simply return if recording is not enabled.
- if (!rtc::AtomicOps::AcquireLoad(&recording_)) return result;
-
- // Set the size of our own audio buffer and clear it first to avoid copying
- // in combination with potential reallocations.
- // On real iOS devices, the size will only be set once (at first callback).
- record_audio_buffer_.Clear();
- record_audio_buffer_.SetSize(num_frames);
-
- // Allocate AudioBuffers to be used as storage for the received audio.
- // The AudioBufferList structure works as a placeholder for the
- // AudioBuffer structure, which holds a pointer to the actual data buffer
- // in |record_audio_buffer_|. Recorded audio will be rendered into this memory
- // at each input callback when calling AudioUnitRender().
- AudioBufferList audio_buffer_list;
- audio_buffer_list.mNumberBuffers = 1;
- AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0];
- audio_buffer->mNumberChannels = record_parameters_.channels();
- audio_buffer->mDataByteSize =
- record_audio_buffer_.size() * VoiceProcessingAudioUnit::kBytesPerSample;
- audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
-
- // Obtain the recorded audio samples by initiating a rendering cycle.
- // Since it happens on the input bus, the |io_data| parameter is a reference
- // to the preallocated audio buffer list that the audio unit renders into.
- // We can make the audio unit provide a buffer instead in io_data, but we
- // currently just use our own.
- // TODO(henrika): should error handling be improved?
- result = audio_unit_->Render(flags, time_stamp, bus_number, num_frames, &audio_buffer_list);
- if (result != noErr) {
- RTCLogError(@"Failed to render audio.");
- return result;
- }
-
- // Get a pointer to the recorded audio and send it to the WebRTC ADB.
- // Use the FineAudioBuffer instance to convert between native buffer size
- // and the 10ms buffer size used by WebRTC.
- fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_, kFixedRecordDelayEstimate);
- return noErr;
-}
-
-OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- RTC_DCHECK_RUN_ON(&io_thread_checker_);
- // Verify 16-bit, noninterleaved mono PCM signal format.
- RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
- AudioBuffer* audio_buffer = &io_data->mBuffers[0];
- RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
-
- // Produce silence and give audio unit a hint about it if playout is not
- // activated.
- if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
- const size_t size_in_bytes = audio_buffer->mDataByteSize;
- RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
- *flags |= kAudioUnitRenderAction_OutputIsSilence;
- memset(static_cast<int8_t*>(audio_buffer->mData), 0, size_in_bytes);
- return noErr;
- }
-
- // Measure time since last call to OnGetPlayoutData() and see if it is larger
- // than a well defined threshold which depends on the current IO buffer size.
- // If so, we have an indication of a glitch in the output audio since the
- // core audio layer will most likely run dry in this state.
- ++num_playout_callbacks_;
- const int64_t now_time = rtc::TimeMillis();
- if (time_stamp->mSampleTime != num_frames) {
- const int64_t delta_time = now_time - last_playout_time_;
- const int glitch_threshold = 1.6 * playout_parameters_.GetBufferSizeInMilliseconds();
- if (delta_time > glitch_threshold) {
- RTCLogWarning(@"Possible playout audio glitch detected.\n"
- " Time since last OnGetPlayoutData was %lld ms.\n",
- delta_time);
- // Exclude extreme delta values since they do most likely not correspond
- // to a real glitch. Instead, the most probable cause is that a headset
- // has been plugged in or out. There are more direct ways to detect
- // audio device changes (see HandleValidRouteChange()) but experiments
- // show that using it leads to more complex implementations.
- // TODO(henrika): more tests might be needed to come up with an even
- // better upper limit.
- if (glitch_threshold < 120 && delta_time > 120) {
- RTCLog(@"Glitch warning is ignored. Probably caused by device switch.");
- } else {
- thread_->Post(RTC_FROM_HERE, this, kMessageTypePlayoutGlitchDetected);
- }
- }
- }
- last_playout_time_ = now_time;
-
- // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
- // the native I/O audio unit) and copy the result to the audio buffer in the
- // |io_data| destination.
- fine_audio_buffer_->GetPlayoutData(
- rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
- kFixedPlayoutDelayEstimate);
- return noErr;
-}
-
-void AudioDeviceIOS::OnMessage(rtc::Message* msg) {
- switch (msg->message_id) {
- case kMessageTypeInterruptionBegin:
- HandleInterruptionBegin();
- break;
- case kMessageTypeInterruptionEnd:
- HandleInterruptionEnd();
- break;
- case kMessageTypeValidRouteChange:
- HandleValidRouteChange();
- break;
- case kMessageTypeCanPlayOrRecordChange: {
- rtc::TypedMessageData<bool>* data = static_cast<rtc::TypedMessageData<bool>*>(msg->pdata);
- HandleCanPlayOrRecordChange(data->data());
- delete data;
- break;
- }
- case kMessageTypePlayoutGlitchDetected:
- HandlePlayoutGlitchDetected();
- break;
- case kMessageOutputVolumeChange:
- HandleOutputVolumeChange();
- break;
- }
-}
-
-void AudioDeviceIOS::HandleInterruptionBegin() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.", is_interrupted_);
- if (audio_unit_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
- RTCLog(@"Stopping the audio unit due to interruption begin.");
- if (!audio_unit_->Stop()) {
- RTCLogError(@"Failed to stop the audio unit for interruption begin.");
- } else {
- PrepareForNewStart();
- }
- }
- is_interrupted_ = true;
-}
-
-void AudioDeviceIOS::HandleInterruptionEnd() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
- "Updating audio unit state.",
- is_interrupted_);
- is_interrupted_ = false;
- UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
-}
-
-void AudioDeviceIOS::HandleValidRouteChange() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- RTCLog(@"%@", session);
- HandleSampleRateChange(session.sampleRate);
-}
-
-void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) {
- RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record);
- UpdateAudioUnit(can_play_or_record);
-}
-
-void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Handling sample rate change to %f.", sample_rate);
-
- // Don't do anything if we're interrupted.
- if (is_interrupted_) {
- RTCLog(@"Ignoring sample rate change to %f due to interruption.", sample_rate);
- return;
- }
-
- // If we don't have an audio unit yet, or the audio unit is uninitialized,
- // there is no work to do.
- if (!audio_unit_ || audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
- return;
- }
-
- // The audio unit is already initialized or started.
- // Check to see if the sample rate or buffer size has changed.
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- const double session_sample_rate = session.sampleRate;
- const NSTimeInterval session_buffer_duration = session.IOBufferDuration;
- const size_t session_frames_per_buffer =
- static_cast<size_t>(session_sample_rate * session_buffer_duration + .5);
- const double current_sample_rate = playout_parameters_.sample_rate();
- const size_t current_frames_per_buffer = playout_parameters_.frames_per_buffer();
- RTCLog(@"Handling playout sample rate change to: %f\n"
- " Session sample rate: %f frames_per_buffer: %lu\n"
- " ADM sample rate: %f frames_per_buffer: %lu",
- sample_rate,
- session_sample_rate,
- (unsigned long)session_frames_per_buffer,
- current_sample_rate,
- (unsigned long)current_frames_per_buffer);
-
- // Sample rate and buffer size are the same, no work to do.
- if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
- current_frames_per_buffer == session_frames_per_buffer) {
- RTCLog(@"Ignoring sample rate change since audio parameters are intact.");
- return;
- }
-
- // Extra sanity check to ensure that the new sample rate is valid.
- if (session_sample_rate <= 0.0) {
- RTCLogError(@"Sample rate is invalid: %f", session_sample_rate);
- return;
- }
-
- // We need to adjust our format and buffer sizes.
- // The stream format is about to be changed and it requires that we first
- // stop and uninitialize the audio unit to deallocate its resources.
- RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
- bool restart_audio_unit = false;
- if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
- audio_unit_->Stop();
- restart_audio_unit = true;
- PrepareForNewStart();
- }
- if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
- audio_unit_->Uninitialize();
- }
-
- // Allocate new buffers given the new stream format.
- SetupAudioBuffersForActiveAudioSession();
-
- // Initialize the audio unit again with the new sample rate.
- RTC_DCHECK_EQ(playout_parameters_.sample_rate(), session_sample_rate);
- if (!audio_unit_->Initialize(session_sample_rate)) {
- RTCLogError(@"Failed to initialize the audio unit with sample rate: %f", session_sample_rate);
- return;
- }
-
- // Restart the audio unit if it was already running.
- if (restart_audio_unit && !audio_unit_->Start()) {
- RTCLogError(@"Failed to start audio unit with sample rate: %f", session_sample_rate);
- return;
- }
- RTCLog(@"Successfully handled sample rate change.");
-}
-
-void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- // Don't update metrics if we're interrupted since a "glitch" is expected
- // in this state.
- if (is_interrupted_) {
- RTCLog(@"Ignoring audio glitch due to interruption.");
- return;
- }
- // Avoid doing glitch detection for two seconds after a volume change
- // has been detected to reduce the risk of false alarm.
- if (last_output_volume_change_time_ > 0 &&
- rtc::TimeSince(last_output_volume_change_time_) < 2000) {
- RTCLog(@"Ignoring audio glitch due to recent output volume change.");
- return;
- }
- num_detected_playout_glitches_++;
- RTCLog(@"Number of detected playout glitches: %lld", num_detected_playout_glitches_);
-
- int64_t glitch_count = num_detected_playout_glitches_;
- dispatch_async(dispatch_get_main_queue(), ^{
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- [session notifyDidDetectPlayoutGlitch:glitch_count];
- });
-}
-
-void AudioDeviceIOS::HandleOutputVolumeChange() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Output volume change detected.");
- // Store time of this detection so it can be used to defer detection of
- // glitches too close in time to this event.
- last_output_volume_change_time_ = rtc::TimeMillis();
-}
-
-void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
- LOGI() << "UpdateAudioDevicebuffer";
- // AttachAudioBuffer() is called at construction by the main class but check
- // just in case.
- RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
- RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0);
- RTC_DCHECK_GT(record_parameters_.sample_rate(), 0);
- RTC_DCHECK_EQ(playout_parameters_.channels(), 1);
- RTC_DCHECK_EQ(record_parameters_.channels(), 1);
- // Inform the audio device buffer (ADB) about the new audio format.
- audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
- audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
- audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
- audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
-}
-
-void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
- LOGI() << "SetupAudioBuffersForActiveAudioSession";
- // Verify the current values once the audio session has been activated.
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- double sample_rate = session.sampleRate;
- NSTimeInterval io_buffer_duration = session.IOBufferDuration;
- RTCLog(@"%@", session);
-
- // Log a warning message for the case when we are unable to set the preferred
- // hardware sample rate but continue and use the non-ideal sample rate after
- // reinitializing the audio parameters. Most BT headsets only support 8kHz or
- // 16kHz.
- RTCAudioSessionConfiguration* webRTCConfig = [RTCAudioSessionConfiguration webRTCConfiguration];
- if (sample_rate != webRTCConfig.sampleRate) {
- RTC_LOG(LS_WARNING) << "Unable to set the preferred sample rate";
- }
-
- // Crash reports indicates that it can happen in rare cases that the reported
- // sample rate is less than or equal to zero. If that happens and if a valid
- // sample rate has already been set during initialization, the best guess we
- // can do is to reuse the current sample rate.
- if (sample_rate <= DBL_EPSILON && playout_parameters_.sample_rate() > 0) {
- RTCLogError(@"Reported rate is invalid: %f. "
- "Using %d as sample rate instead.",
- sample_rate, playout_parameters_.sample_rate());
- sample_rate = playout_parameters_.sample_rate();
- }
-
- // At this stage, we also know the exact IO buffer duration and can add
- // that info to the existing audio parameters where it is converted into
- // number of audio frames.
- // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
- // Hence, 128 is the size we expect to see in upcoming render callbacks.
- playout_parameters_.reset(sample_rate, playout_parameters_.channels(), io_buffer_duration);
- RTC_DCHECK(playout_parameters_.is_complete());
- record_parameters_.reset(sample_rate, record_parameters_.channels(), io_buffer_duration);
- RTC_DCHECK(record_parameters_.is_complete());
- RTC_LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
- RTC_LOG(LS_INFO) << " bytes per I/O buffer: " << playout_parameters_.GetBytesPerBuffer();
- RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(), record_parameters_.GetBytesPerBuffer());
-
- // Update the ADB parameters since the sample rate might have changed.
- UpdateAudioDeviceBuffer();
-
- // Create a modified audio buffer class which allows us to ask for,
- // or deliver, any number of samples (and not only multiple of 10ms) to match
- // the native audio unit buffer size.
- RTC_DCHECK(audio_device_buffer_);
- fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_));
-}
-
-bool AudioDeviceIOS::CreateAudioUnit() {
- RTC_DCHECK(!audio_unit_);
-
- audio_unit_.reset(new VoiceProcessingAudioUnit(this));
- if (!audio_unit_->Init()) {
- audio_unit_.reset();
- return false;
- }
-
- return true;
-}
-
-void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
- can_play_or_record,
- is_interrupted_);
-
- if (is_interrupted_) {
- RTCLog(@"Ignoring audio unit update due to interruption.");
- return;
- }
-
- // If we're not initialized we don't need to do anything. Audio unit will
- // be initialized on initialization.
- if (!audio_is_initialized_) return;
-
- // If we're initialized, we must have an audio unit.
- RTC_DCHECK(audio_unit_);
-
- bool should_initialize_audio_unit = false;
- bool should_uninitialize_audio_unit = false;
- bool should_start_audio_unit = false;
- bool should_stop_audio_unit = false;
-
- switch (audio_unit_->GetState()) {
- case VoiceProcessingAudioUnit::kInitRequired:
- RTCLog(@"VPAU state: InitRequired");
- RTC_NOTREACHED();
- break;
- case VoiceProcessingAudioUnit::kUninitialized:
- RTCLog(@"VPAU state: Uninitialized");
- should_initialize_audio_unit = can_play_or_record;
- should_start_audio_unit = should_initialize_audio_unit && (playing_ || recording_);
- break;
- case VoiceProcessingAudioUnit::kInitialized:
- RTCLog(@"VPAU state: Initialized");
- should_start_audio_unit = can_play_or_record && (playing_ || recording_);
- should_uninitialize_audio_unit = !can_play_or_record;
- break;
- case VoiceProcessingAudioUnit::kStarted:
- RTCLog(@"VPAU state: Started");
- RTC_DCHECK(playing_ || recording_);
- should_stop_audio_unit = !can_play_or_record;
- should_uninitialize_audio_unit = should_stop_audio_unit;
- break;
- }
-
- if (should_initialize_audio_unit) {
- RTCLog(@"Initializing audio unit for UpdateAudioUnit");
- ConfigureAudioSession();
- SetupAudioBuffersForActiveAudioSession();
- if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
- RTCLogError(@"Failed to initialize audio unit.");
- return;
- }
- }
-
- if (should_start_audio_unit) {
- RTCLog(@"Starting audio unit for UpdateAudioUnit");
- // Log session settings before trying to start audio streaming.
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- RTCLog(@"%@", session);
- if (!audio_unit_->Start()) {
- RTCLogError(@"Failed to start audio unit.");
- return;
- }
- }
-
- if (should_stop_audio_unit) {
- RTCLog(@"Stopping audio unit for UpdateAudioUnit");
- if (!audio_unit_->Stop()) {
- RTCLogError(@"Failed to stop audio unit.");
- return;
- }
- }
-
- if (should_uninitialize_audio_unit) {
- RTCLog(@"Uninitializing audio unit for UpdateAudioUnit");
- audio_unit_->Uninitialize();
- UnconfigureAudioSession();
- }
-}
-
-bool AudioDeviceIOS::ConfigureAudioSession() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Configuring audio session.");
- if (has_configured_session_) {
- RTCLogWarning(@"Audio session already configured.");
- return false;
- }
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- [session lockForConfiguration];
- bool success = [session configureWebRTCSession:nil];
- [session unlockForConfiguration];
- if (success) {
- has_configured_session_ = true;
- RTCLog(@"Configured audio session.");
- } else {
- RTCLog(@"Failed to configure audio session.");
- }
- return success;
-}
-
-void AudioDeviceIOS::UnconfigureAudioSession() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Unconfiguring audio session.");
- if (!has_configured_session_) {
- RTCLogWarning(@"Audio session already unconfigured.");
- return;
- }
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- [session lockForConfiguration];
- [session unconfigureWebRTCSession:nil];
- [session unlockForConfiguration];
- has_configured_session_ = false;
- RTCLog(@"Unconfigured audio session.");
-}
-
-bool AudioDeviceIOS::InitPlayOrRecord() {
- LOGI() << "InitPlayOrRecord";
- RTC_DCHECK_RUN_ON(&thread_checker_);
-
- // There should be no audio unit at this point.
- if (!CreateAudioUnit()) {
- return false;
- }
-
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- // Subscribe to audio session events.
- [session pushDelegate:audio_session_observer_];
- is_interrupted_ = session.isInterrupted ? true : false;
-
- // Lock the session to make configuration changes.
- [session lockForConfiguration];
- NSError* error = nil;
- if (![session beginWebRTCSession:&error]) {
- [session unlockForConfiguration];
- RTCLogError(@"Failed to begin WebRTC session: %@", error.localizedDescription);
- audio_unit_.reset();
- return false;
- }
-
- // If we are ready to play or record, and if the audio session can be
- // configured, then initialize the audio unit.
- if (session.canPlayOrRecord) {
- if (!ConfigureAudioSession()) {
- // One possible reason for failure is if an attempt was made to use the
- // audio session during or after a Media Services failure.
- // See AVAudioSessionErrorCodeMediaServicesFailed for details.
- [session unlockForConfiguration];
- audio_unit_.reset();
- return false;
- }
- SetupAudioBuffersForActiveAudioSession();
- audio_unit_->Initialize(playout_parameters_.sample_rate());
- }
-
- // Release the lock.
- [session unlockForConfiguration];
- return true;
-}
-
-void AudioDeviceIOS::ShutdownPlayOrRecord() {
- LOGI() << "ShutdownPlayOrRecord";
- RTC_DCHECK_RUN_ON(&thread_checker_);
-
- // Stop the audio unit to prevent any additional audio callbacks.
- audio_unit_->Stop();
-
- // Close and delete the voice-processing I/O unit.
- audio_unit_.reset();
-
- // Detach thread checker for the AURemoteIO::IOThread to ensure that the
- // next session uses a fresh thread id.
- io_thread_checker_.Detach();
-
- // Remove audio session notification observers.
- RTCAudioSession* session = [RTCAudioSession sharedInstance];
- [session removeDelegate:audio_session_observer_];
-
- // All I/O should be stopped or paused prior to deactivating the audio
- // session, hence we deactivate as last action.
- [session lockForConfiguration];
- UnconfigureAudioSession();
- [session endWebRTCSession:nil];
- [session unlockForConfiguration];
-}
-
-void AudioDeviceIOS::PrepareForNewStart() {
- LOGI() << "PrepareForNewStart";
- // The audio unit has been stopped and preparations are needed for an upcoming
- // restart. It will result in audio callbacks from a new native I/O thread
- // which means that we must detach thread checkers here to be prepared for an
- // upcoming new audio stream.
- io_thread_checker_.Detach();
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/ios/audio_device_not_implemented_ios.mm b/modules/audio_device/ios/audio_device_not_implemented_ios.mm
deleted file mode 100644
index 2e99aea..0000000
--- a/modules/audio_device/ios/audio_device_not_implemented_ios.mm
+++ /dev/null
@@ -1,205 +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/ios/audio_device_ios.h"
-
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-
-namespace webrtc {
-
-int32_t AudioDeviceIOS::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const {
- audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
- return 0;
-}
-
-int16_t AudioDeviceIOS::PlayoutDevices() {
- // TODO(henrika): improve.
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return (int16_t)1;
-}
-
-int16_t AudioDeviceIOS::RecordingDevices() {
- // TODO(henrika): improve.
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return (int16_t)1;
-}
-
-int32_t AudioDeviceIOS::InitSpeaker() {
- return 0;
-}
-
-bool AudioDeviceIOS::SpeakerIsInitialized() const {
- return true;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MinSpeakerVolume(uint32_t& minVolume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::InitMicrophone() {
- return 0;
-}
-
-bool AudioDeviceIOS::MicrophoneIsInitialized() const {
- return true;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
- enabled = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
- enabled = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
- available = false;
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
- RTC_LOG_F(LS_WARNING) << "Not implemented";
- return 0;
-}
-
-int32_t AudioDeviceIOS::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType) {
- RTC_NOTREACHED() << "Not implemented";
- return -1;
-}
-
-int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
- available = true;
- return 0;
-}
-
-int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
- available = true;
- return 0;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/ios/audio_device_unittest_ios.mm b/modules/audio_device/ios/audio_device_unittest_ios.mm
deleted file mode 100644
index 74cfcc2..0000000
--- a/modules/audio_device/ios/audio_device_unittest_ios.mm
+++ /dev/null
@@ -1,877 +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 <algorithm>
-#include <limits>
-#include <list>
-#include <memory>
-#include <numeric>
-#include <string>
-#include <vector>
-
-#include "api/scoped_refptr.h"
-#include "modules/audio_device/audio_device_impl.h"
-#include "modules/audio_device/include/audio_device.h"
-#include "modules/audio_device/include/mock_audio_transport.h"
-#include "modules/audio_device/ios/audio_device_ios.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/critical_section.h"
-#include "rtc_base/event.h"
-#include "rtc_base/format_macros.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/time_utils.h"
-#include "test/gmock.h"
-#include "test/gtest.h"
-#include "test/testsupport/file_utils.h"
-
-#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
-#import "sdk/objc/components/audio/RTCAudioSession.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;
-// Number of bits per PCM audio sample.
-static const size_t kBitsPerSample = 16;
-// Number of bytes per PCM audio sample.
-static const size_t kBytesPerSample = kBitsPerSample / 8;
-// 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 = 15;
-// Run the full-duplex test during this time (unit is in seconds).
-// Note that first |kNumIgnoreFirstCallbacks| are ignored.
-static const int kFullDuplexTimeInSec = 10;
-// 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.
-// TODO(henrika): fine tune this setting for iOS.
-static const int kImpulseFrequencyInHz = 1;
-// Length of round-trip latency measurements. Number of transmitted impulses
-// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
-// TODO(henrika): fine tune this setting for iOS.
-static const int kMeasureLatencyTimeInSec = 5;
-// Utilized in round-trip latency measurements to avoid capturing noise samples.
-// TODO(henrika): fine tune this setting for iOS.
-static const int kImpulseThreshold = 50;
-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_);
- rtc::CritScope lock(&lock_);
- fifo_->push_back(memory);
- const size_t size = fifo_->size();
- if (size > largest_size_) {
- largest_size_ = size;
- PRINTD("(%" PRIuS ")", 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("-");
- rtc::CritScope 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*>;
- rtc::CriticalSection 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 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 MockAudioTransportIOS : public test::MockAudioTransport {
- public:
- explicit MockAudioTransportIOS(int type)
- : num_callbacks_(0),
- type_(type),
- play_count_(0),
- rec_count_(0),
- audio_stream_(nullptr) {}
-
- virtual ~MockAudioTransportIOS() {}
-
- // 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,
- size_t 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, &MockAudioTransportIOS::RealNeedMorePlayData));
- }
- if (rec_mode()) {
- ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
- .WillByDefault(Invoke(
- this, &MockAudioTransportIOS::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) {
- 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()) {
- if (test_is_done_) {
- 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,
- 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);
- } else {
- memset(audioSamples, 0, nSamples * nBytesPerSample);
- }
- if (ReceivedEnoughCallbacks()) {
- if (test_is_done_) {
- 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_;
-};
-
-// AudioDeviceTest test fixture.
-class AudioDeviceTest : public ::testing::Test {
- protected:
- AudioDeviceTest() {
- old_sev_ = rtc::LogMessage::GetLogToDebug();
- // Set suitable logging level here. Change to rtc::LS_INFO for more verbose
- // output. See webrtc/rtc_base/logging.h for complete list of options.
- rtc::LogMessage::LogToDebug(rtc::LS_INFO);
- // Add extra logging fields here (timestamps and thread id).
- // rtc::LogMessage::LogTimestamps();
- rtc::LogMessage::LogThreads();
- // 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());
- EXPECT_EQ(0,
- audio_device()->GetPlayoutAudioParameters(&playout_parameters_));
- EXPECT_EQ(0, audio_device()->GetRecordAudioParameters(&record_parameters_));
- }
- virtual ~AudioDeviceTest() {
- EXPECT_EQ(0, audio_device_->Terminate());
- rtc::LogMessage::LogToDebug(old_sev_);
- }
-
- int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
- int record_sample_rate() const { return record_parameters_.sample_rate(); }
- int playout_channels() const { return playout_parameters_.channels(); }
- int 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();
- }
-
- rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
- return audio_device_;
- }
-
- AudioDeviceModuleImpl* audio_device_impl() const {
- return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
- }
-
- 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));
- 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 ||
- sample_rate == 16000);
- 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_DEBUG_PRINTF
- PRINTD("file name: %s\n", file_name.c_str());
- const size_t bytes = test::GetFileSize(file_name);
- PRINTD("file size: %" PRIuS " [bytes]\n", bytes);
- PRINTD("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
- const int seconds =
- static_cast<int>(bytes / (sample_rate * kBytesPerSample));
- PRINTD("file size: %d [secs]\n", seconds);
- PRINTD("file size: %" PRIuS " [callbacks]\n",
- seconds * kNumCallbacksPerSecond);
-#endif
- return file_name;
- }
-
- void StartPlayout() {
- 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());
- }
-
- void StartRecording() {
- 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());
- }
-
- rtc::Event test_is_done_;
- rtc::scoped_refptr<AudioDeviceModule> audio_device_;
- AudioParameters playout_parameters_;
- AudioParameters record_parameters_;
- rtc::LoggingSeverity old_sev_;
-};
-
-TEST_F(AudioDeviceTest, ConstructDestruct) {
- // Using the test fixture to create and destruct the audio device module.
-}
-
-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());
-}
-
-// Tests that playout can be initiated, started and stopped. No audio callback
-// is registered in this test.
-// Failing when running on real iOS devices: bugs.webrtc.org/6889.
-TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) {
- StartPlayout();
- StopPlayout();
- StartPlayout();
- StopPlayout();
-}
-
-// Tests that recording can be initiated, started and stopped. No audio callback
-// is registered in this test.
-// Can sometimes fail when running on real devices: bugs.webrtc.org/7888.
-TEST_F(AudioDeviceTest, DISABLED_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.
-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 we can create two ADMs and start playing on the second ADM.
-// Only the first active instance shall activate an audio session and the
-// last active instance shall deactivate the audio session. The test does not
-// explicitly verify correct audio session calls but instead focuses on
-// ensuring that audio starts for both ADMs.
-
-// Failing when running on real iOS devices: bugs.webrtc.org/6889.
-TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
- // Create and initialize a second/extra ADM instance. The default ADM is
- // created by the test harness.
- rtc::scoped_refptr<AudioDeviceModule> second_audio_device =
- CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
- EXPECT_NE(second_audio_device.get(), nullptr);
- EXPECT_EQ(0, second_audio_device->Init());
-
- // Start playout for the default ADM but don't wait here. Instead use the
- // upcoming second stream for that. We set the same expectation on number
- // of callbacks as for the second stream.
- NiceMock<MockAudioTransportIOS> mock(kPlayout);
- mock.HandleCallbacks(nullptr, nullptr, 0);
- 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();
-
- // Initialize playout for the second ADM. If all is OK, the second ADM shall
- // reuse the audio session activated when the first ADM started playing.
- // This call will also ensure that we avoid a problem related to initializing
- // two different audio unit instances back to back (see webrtc:5166 for
- // details).
- EXPECT_EQ(0, second_audio_device->InitPlayout());
- EXPECT_TRUE(second_audio_device->PlayoutIsInitialized());
-
- // Start playout for the second ADM and verify that it starts as intended.
- // Passing this test ensures that initialization of the second audio unit
- // has been done successfully and that there is no conflict with the already
- // playing first ADM.
- MockAudioTransportIOS mock2(kPlayout);
- mock2.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
- EXPECT_CALL(
- mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
- playout_channels(), playout_sample_rate(),
- NotNull(), _, _, _))
- .Times(AtLeast(kNumCallbacks));
- EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
- EXPECT_EQ(0, second_audio_device->StartPlayout());
- EXPECT_TRUE(second_audio_device->Playing());
- test_is_done_.Wait(kTestTimeOutInMilliseconds);
- EXPECT_EQ(0, second_audio_device->StopPlayout());
- EXPECT_FALSE(second_audio_device->Playing());
- EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
-
- EXPECT_EQ(0, second_audio_device->Terminate());
-}
-
-// 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) {
- MockAudioTransportIOS 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.
-TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
- MockAudioTransportIOS 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(),
- _, // TODO(henrika): fix delay
- 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) {
- MockAudioTransportIOS 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(),
- _, // TODO(henrika): fix delay
- 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(1, playout_channels());
- NiceMock<MockAudioTransportIOS> 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();
-}
-
-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());
-}
-
-// 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.
-TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
- EXPECT_EQ(record_channels(), playout_channels());
- EXPECT_EQ(record_sample_rate(), playout_sample_rate());
- NiceMock<MockAudioTransportIOS> 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();
- EXPECT_LE(fifo_audio_stream->average_size(), 10u);
- EXPECT_LE(fifo_audio_stream->largest_size(), 20u);
-}
-
-// 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<MockAudioTransportIOS> 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();
-}
-
-// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
-// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
-// AudioDeviceIOS listens to RTCAudioSession interrupted notifications by:
-// - In AudioDeviceIOS.InitPlayOrRecord registers its audio_session_observer_
-// callback with RTCAudioSession's delegate list.
-// - When RTCAudioSession receives an iOS audio interrupted notification, it
-// passes the notification to callbacks in its delegate list which sets
-// AudioDeviceIOS's is_interrupted_ flag to true.
-// - When AudioDeviceIOS.ShutdownPlayOrRecord is called, its
-// audio_session_observer_ callback is removed from RTCAudioSessions's
-// delegate list.
-// So if RTCAudioSession receives an iOS end audio interruption notification,
-// AudioDeviceIOS is not notified as its callback is not in RTCAudioSession's
-// delegate list. This causes AudioDeviceIOS's is_interrupted_ flag to be in
-// the wrong (true) state and the audio session will ignore audio changes.
-// As RTCAudioSession keeps its own interrupted state, the fix is to initialize
-// AudioDeviceIOS's is_interrupted_ flag to RTCAudioSession's isInterrupted
-// flag in AudioDeviceIOS.InitPlayOrRecord.
-TEST_F(AudioDeviceTest, testInterruptedAudioSession) {
- RTCAudioSession *session = [RTCAudioSession sharedInstance];
- std::unique_ptr<webrtc::AudioDeviceIOS> audio_device;
- audio_device.reset(new webrtc::AudioDeviceIOS());
- std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
- audio_buffer.reset(new webrtc::AudioDeviceBuffer());
- audio_device->AttachAudioBuffer(audio_buffer.get());
- audio_device->Init();
- audio_device->InitPlayout();
- // Force interruption.
- [session notifyDidBeginInterruption];
-
- // Wait for notification to propagate.
- rtc::MessageQueueManager::ProcessAllMessageQueuesForTesting();
- EXPECT_TRUE(audio_device->is_interrupted_);
-
- // Force it for testing.
- audio_device->playing_ = false;
- audio_device->ShutdownPlayOrRecord();
- // Force it for testing.
- audio_device->audio_is_initialized_ = false;
-
- [session notifyDidEndInterruptionWithShouldResumeSession:YES];
- // Wait for notification to propagate.
- rtc::MessageQueueManager::ProcessAllMessageQueuesForTesting();
- EXPECT_TRUE(audio_device->is_interrupted_);
-
- audio_device->Init();
- audio_device->InitPlayout();
- EXPECT_FALSE(audio_device->is_interrupted_);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_device/ios/audio_session_observer.h b/modules/audio_device/ios/audio_session_observer.h
deleted file mode 100644
index c79cdd14..0000000
--- a/modules/audio_device/ios/audio_session_observer.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright 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_IOS_AUDIO_SESSION_OBSERVER_H_
-#define MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
-
-#include "rtc_base/async_invoker.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-
-// Observer interface for listening to AVAudioSession events.
-class AudioSessionObserver {
- public:
- // Called when audio session interruption begins.
- virtual void OnInterruptionBegin() = 0;
-
- // Called when audio session interruption ends.
- virtual void OnInterruptionEnd() = 0;
-
- // Called when audio route changes.
- virtual void OnValidRouteChange() = 0;
-
- // Called when the ability to play or record changes.
- virtual void OnCanPlayOrRecordChange(bool can_play_or_record) = 0;
-
- virtual void OnChangedOutputVolume() = 0;
-
- protected:
- virtual ~AudioSessionObserver() {}
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
diff --git a/modules/audio_device/ios/objc/RTCAudioSession.h b/modules/audio_device/ios/objc/RTCAudioSession.h
deleted file mode 100644
index 23abc3d..0000000
--- a/modules/audio_device/ios/objc/RTCAudioSession.h
+++ /dev/null
@@ -1,11 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import "sdk/objc/components/audio/RTCAudioSession.h"
diff --git a/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h b/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
deleted file mode 100644
index 2584053..0000000
--- a/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
+++ /dev/null
@@ -1,11 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
diff --git a/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h b/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
deleted file mode 100644
index 54f4c26..0000000
--- a/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright 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.
- */
-
-#import "sdk/objc/components/audio/RTCAudioSession.h"
-
-namespace webrtc {
-class AudioSessionObserver;
-}
-
-/** Adapter that forwards RTCAudioSessionDelegate calls to the appropriate
- * methods on the AudioSessionObserver.
- */
-@interface RTCAudioSessionDelegateAdapter : NSObject <RTCAudioSessionDelegate>
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** |observer| is a raw pointer and should be kept alive
- * for this object's lifetime.
- */
-- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
-
-@end
diff --git a/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm b/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm
deleted file mode 100644
index 818e077..0000000
--- a/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright 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.
- */
-
-#import "modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
-
-#include "modules/audio_device/ios/audio_session_observer.h"
-
-#import "sdk/objc/base/RTCLogging.h"
-
-@implementation RTCAudioSessionDelegateAdapter {
- webrtc::AudioSessionObserver *_observer;
-}
-
-- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
- NSParameterAssert(observer);
- if (self = [super init]) {
- _observer = observer;
- }
- return self;
-}
-
-#pragma mark - RTCAudioSessionDelegate
-
-- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session {
- _observer->OnInterruptionBegin();
-}
-
-- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
- shouldResumeSession:(BOOL)shouldResumeSession {
- _observer->OnInterruptionEnd();
-}
-
-- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
- reason:(AVAudioSessionRouteChangeReason)reason
- previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
- switch (reason) {
- case AVAudioSessionRouteChangeReasonUnknown:
- case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
- case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
- case AVAudioSessionRouteChangeReasonCategoryChange:
- // It turns out that we see a category change (at least in iOS 9.2)
- // when making a switch from a BT device to e.g. Speaker using the
- // iOS Control Center and that we therefore must check if the sample
- // rate has changed. And if so is the case, restart the audio unit.
- case AVAudioSessionRouteChangeReasonOverride:
- case AVAudioSessionRouteChangeReasonWakeFromSleep:
- case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
- _observer->OnValidRouteChange();
- break;
- case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
- // The set of input and output ports has not changed, but their
- // configuration has, e.g., a port’s selected data source has
- // changed. Ignore this type of route change since we are focusing
- // on detecting headset changes.
- RTCLog(@"Ignoring RouteConfigurationChange");
- break;
- }
-}
-
-- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session {
-}
-
-- (void)audioSessionMediaServerReset:(RTCAudioSession *)session {
-}
-
-- (void)audioSession:(RTCAudioSession *)session
- didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
- _observer->OnCanPlayOrRecordChange(canPlayOrRecord);
-}
-
-- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session {
-}
-
-- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session {
-}
-
-- (void)audioSession:(RTCAudioSession *)audioSession
- didChangeOutputVolume:(float)outputVolume {
- _observer->OnChangedOutputVolume();
-}
-
-@end
diff --git a/modules/audio_device/ios/voice_processing_audio_unit.h b/modules/audio_device/ios/voice_processing_audio_unit.h
deleted file mode 100644
index 3105c12..0000000
--- a/modules/audio_device/ios/voice_processing_audio_unit.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * Copyright 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_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
-#define MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
-
-#include <AudioUnit/AudioUnit.h>
-
-namespace webrtc {
-
-class VoiceProcessingAudioUnitObserver {
- public:
- // Callback function called on a real-time priority I/O thread from the audio
- // unit. This method is used to signal that recorded audio is available.
- virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) = 0;
-
- // Callback function called on a real-time priority I/O thread from the audio
- // unit. This method is used to provide audio samples to the audio unit.
- virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) = 0;
-
- protected:
- ~VoiceProcessingAudioUnitObserver() {}
-};
-
-// Convenience class to abstract away the management of a Voice Processing
-// I/O Audio Unit. The Voice Processing I/O unit has the same characteristics
-// as the Remote I/O unit (supports full duplex low-latency audio input and
-// output) and adds AEC for for two-way duplex communication. It also adds AGC,
-// adjustment of voice-processing quality, and muting. Hence, ideal for
-// VoIP applications.
-class VoiceProcessingAudioUnit {
- public:
- explicit VoiceProcessingAudioUnit(VoiceProcessingAudioUnitObserver* observer);
- ~VoiceProcessingAudioUnit();
-
- // TODO(tkchin): enum for state and state checking.
- enum State : int32_t {
- // Init() should be called.
- kInitRequired,
- // Audio unit created but not initialized.
- kUninitialized,
- // Initialized but not started. Equivalent to stopped.
- kInitialized,
- // Initialized and started.
- kStarted,
- };
-
- // Number of bytes per audio sample for 16-bit signed integer representation.
- static const UInt32 kBytesPerSample;
-
- // Initializes this class by creating the underlying audio unit instance.
- // Creates a Voice-Processing I/O unit and configures it for full-duplex
- // audio. The selected stream format is selected to avoid internal resampling
- // and to match the 10ms callback rate for WebRTC as well as possible.
- // Does not intialize the audio unit.
- bool Init();
-
- VoiceProcessingAudioUnit::State GetState() const;
-
- // Initializes the underlying audio unit with the given sample rate.
- bool Initialize(Float64 sample_rate);
-
- // Starts the underlying audio unit.
- bool Start();
-
- // Stops the underlying audio unit.
- bool Stop();
-
- // Uninitializes the underlying audio unit.
- bool Uninitialize();
-
- // Calls render on the underlying audio unit.
- OSStatus Render(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 output_bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data);
-
- private:
- // The C API used to set callbacks requires static functions. When these are
- // called, they will invoke the relevant instance method by casting
- // in_ref_con to VoiceProcessingAudioUnit*.
- static OSStatus OnGetPlayoutData(void* in_ref_con,
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data);
- static OSStatus OnDeliverRecordedData(void* in_ref_con,
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data);
-
- // Notifies observer that samples are needed for playback.
- OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data);
- // Notifies observer that recorded samples are available for render.
- OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data);
-
- // Returns the predetermined format with a specific sample rate. See
- // implementation file for details on format.
- AudioStreamBasicDescription GetFormat(Float64 sample_rate) const;
-
- // Deletes the underlying audio unit.
- void DisposeAudioUnit();
-
- VoiceProcessingAudioUnitObserver* observer_;
- AudioUnit vpio_unit_;
- VoiceProcessingAudioUnit::State state_;
-};
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
diff --git a/modules/audio_device/ios/voice_processing_audio_unit.mm b/modules/audio_device/ios/voice_processing_audio_unit.mm
deleted file mode 100644
index 41477d1..0000000
--- a/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ /dev/null
@@ -1,468 +0,0 @@
-/*
- * Copyright 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.
- */
-
-#import "modules/audio_device/ios/voice_processing_audio_unit.h"
-
-#include "rtc_base/checks.h"
-#include "rtc_base/system/fallthrough.h"
-#include "system_wrappers/include/metrics.h"
-
-#import "sdk/objc/base//RTCLogging.h"
-#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
-
-#if !defined(NDEBUG)
-static void LogStreamDescription(AudioStreamBasicDescription description) {
- char formatIdString[5];
- UInt32 formatId = CFSwapInt32HostToBig(description.mFormatID);
- bcopy(&formatId, formatIdString, 4);
- formatIdString[4] = '\0';
- RTCLog(@"AudioStreamBasicDescription: {\n"
- " mSampleRate: %.2f\n"
- " formatIDString: %s\n"
- " mFormatFlags: 0x%X\n"
- " mBytesPerPacket: %u\n"
- " mFramesPerPacket: %u\n"
- " mBytesPerFrame: %u\n"
- " mChannelsPerFrame: %u\n"
- " mBitsPerChannel: %u\n"
- " mReserved: %u\n}",
- description.mSampleRate, formatIdString,
- static_cast<unsigned int>(description.mFormatFlags),
- static_cast<unsigned int>(description.mBytesPerPacket),
- static_cast<unsigned int>(description.mFramesPerPacket),
- static_cast<unsigned int>(description.mBytesPerFrame),
- static_cast<unsigned int>(description.mChannelsPerFrame),
- static_cast<unsigned int>(description.mBitsPerChannel),
- static_cast<unsigned int>(description.mReserved));
-}
-#endif
-
-namespace webrtc {
-
-// Calls to AudioUnitInitialize() can fail if called back-to-back on different
-// ADM instances. A fall-back solution is to allow multiple sequential calls
-// with as small delay between each. This factor sets the max number of allowed
-// initialization attempts.
-static const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
-// A VP I/O unit's bus 1 connects to input hardware (microphone).
-static const AudioUnitElement kInputBus = 1;
-// A VP I/O unit's bus 0 connects to output hardware (speaker).
-static const AudioUnitElement kOutputBus = 0;
-
-// Returns the automatic gain control (AGC) state on the processed microphone
-// signal. Should be on by default for Voice Processing audio units.
-static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
- RTC_DCHECK(audio_unit);
- UInt32 size = sizeof(*enabled);
- OSStatus result = AudioUnitGetProperty(audio_unit,
- kAUVoiceIOProperty_VoiceProcessingEnableAGC,
- kAudioUnitScope_Global,
- kInputBus,
- enabled,
- &size);
- RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));
- return result;
-}
-
-VoiceProcessingAudioUnit::VoiceProcessingAudioUnit(
- VoiceProcessingAudioUnitObserver* observer)
- : observer_(observer), vpio_unit_(nullptr), state_(kInitRequired) {
- RTC_DCHECK(observer);
-}
-
-VoiceProcessingAudioUnit::~VoiceProcessingAudioUnit() {
- DisposeAudioUnit();
-}
-
-const UInt32 VoiceProcessingAudioUnit::kBytesPerSample = 2;
-
-bool VoiceProcessingAudioUnit::Init() {
- RTC_DCHECK_EQ(state_, kInitRequired);
-
- // Create an audio component description to identify the Voice Processing
- // I/O audio unit.
- AudioComponentDescription vpio_unit_description;
- vpio_unit_description.componentType = kAudioUnitType_Output;
- vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
- vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
- vpio_unit_description.componentFlags = 0;
- vpio_unit_description.componentFlagsMask = 0;
-
- // Obtain an audio unit instance given the description.
- AudioComponent found_vpio_unit_ref =
- AudioComponentFindNext(nullptr, &vpio_unit_description);
-
- // Create a Voice Processing IO audio unit.
- OSStatus result = noErr;
- result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
- if (result != noErr) {
- vpio_unit_ = nullptr;
- RTCLogError(@"AudioComponentInstanceNew failed. Error=%ld.", (long)result);
- return false;
- }
-
- // Enable input on the input scope of the input element.
- UInt32 enable_input = 1;
- result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
- kAudioUnitScope_Input, kInputBus, &enable_input,
- sizeof(enable_input));
- if (result != noErr) {
- DisposeAudioUnit();
- RTCLogError(@"Failed to enable input on input scope of input element. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Enable output on the output scope of the output element.
- UInt32 enable_output = 1;
- result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
- kAudioUnitScope_Output, kOutputBus,
- &enable_output, sizeof(enable_output));
- if (result != noErr) {
- DisposeAudioUnit();
- RTCLogError(@"Failed to enable output on output scope of output element. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Specify the callback function that provides audio samples to the audio
- // unit.
- AURenderCallbackStruct render_callback;
- render_callback.inputProc = OnGetPlayoutData;
- render_callback.inputProcRefCon = this;
- result = AudioUnitSetProperty(
- vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
- kOutputBus, &render_callback, sizeof(render_callback));
- if (result != noErr) {
- DisposeAudioUnit();
- RTCLogError(@"Failed to specify the render callback on the output bus. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Disable AU buffer allocation for the recorder, we allocate our own.
- // TODO(henrika): not sure that it actually saves resource to make this call.
- UInt32 flag = 0;
- result = AudioUnitSetProperty(
- vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
- kAudioUnitScope_Output, kInputBus, &flag, sizeof(flag));
- if (result != noErr) {
- DisposeAudioUnit();
- RTCLogError(@"Failed to disable buffer allocation on the input bus. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Specify the callback to be called by the I/O thread to us when input audio
- // is available. The recorded samples can then be obtained by calling the
- // AudioUnitRender() method.
- AURenderCallbackStruct input_callback;
- input_callback.inputProc = OnDeliverRecordedData;
- input_callback.inputProcRefCon = this;
- result = AudioUnitSetProperty(vpio_unit_,
- kAudioOutputUnitProperty_SetInputCallback,
- kAudioUnitScope_Global, kInputBus,
- &input_callback, sizeof(input_callback));
- if (result != noErr) {
- DisposeAudioUnit();
- RTCLogError(@"Failed to specify the input callback on the input bus. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- state_ = kUninitialized;
- return true;
-}
-
-VoiceProcessingAudioUnit::State VoiceProcessingAudioUnit::GetState() const {
- return state_;
-}
-
-bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
- RTC_DCHECK_GE(state_, kUninitialized);
- RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
-
- OSStatus result = noErr;
- AudioStreamBasicDescription format = GetFormat(sample_rate);
- UInt32 size = sizeof(format);
-#if !defined(NDEBUG)
- LogStreamDescription(format);
-#endif
-
- // Set the format on the output scope of the input element/bus.
- result =
- AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Output, kInputBus, &format, size);
- if (result != noErr) {
- RTCLogError(@"Failed to set format on output scope of input bus. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Set the format on the input scope of the output element/bus.
- result =
- AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Input, kOutputBus, &format, size);
- if (result != noErr) {
- RTCLogError(@"Failed to set format on input scope of output bus. "
- "Error=%ld.",
- (long)result);
- return false;
- }
-
- // Initialize the Voice Processing I/O unit instance.
- // Calls to AudioUnitInitialize() can fail if called back-to-back on
- // different ADM instances. The error message in this case is -66635 which is
- // undocumented. Tests have shown that calling AudioUnitInitialize a second
- // time, after a short sleep, avoids this issue.
- // See webrtc:5166 for details.
- int failed_initalize_attempts = 0;
- result = AudioUnitInitialize(vpio_unit_);
- while (result != noErr) {
- RTCLogError(@"Failed to initialize the Voice Processing I/O unit. "
- "Error=%ld.",
- (long)result);
- ++failed_initalize_attempts;
- if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
- // Max number of initialization attempts exceeded, hence abort.
- RTCLogError(@"Too many initialization attempts.");
- return false;
- }
- RTCLog(@"Pause 100ms and try audio unit initialization again...");
- [NSThread sleepForTimeInterval:0.1f];
- result = AudioUnitInitialize(vpio_unit_);
- }
- if (result == noErr) {
- RTCLog(@"Voice Processing I/O unit is now initialized.");
- }
-
- // AGC should be enabled by default for Voice Processing I/O units but it is
- // checked below and enabled explicitly if needed. This scheme is used
- // to be absolutely sure that the AGC is enabled since we have seen cases
- // where only zeros are recorded and a disabled AGC could be one of the
- // reasons why it happens.
- int agc_was_enabled_by_default = 0;
- UInt32 agc_is_enabled = 0;
- result = GetAGCState(vpio_unit_, &agc_is_enabled);
- if (result != noErr) {
- RTCLogError(@"Failed to get AGC state (1st attempt). "
- "Error=%ld.",
- (long)result);
- // Example of error code: kAudioUnitErr_NoConnection (-10876).
- // All error codes related to audio units are negative and are therefore
- // converted into a postive value to match the UMA APIs.
- RTC_HISTOGRAM_COUNTS_SPARSE_100000(
- "WebRTC.Audio.GetAGCStateErrorCode1", (-1) * result);
- } else if (agc_is_enabled) {
- // Remember that the AGC was enabled by default. Will be used in UMA.
- agc_was_enabled_by_default = 1;
- } else {
- // AGC was initially disabled => try to enable it explicitly.
- UInt32 enable_agc = 1;
- result =
- AudioUnitSetProperty(vpio_unit_,
- kAUVoiceIOProperty_VoiceProcessingEnableAGC,
- kAudioUnitScope_Global, kInputBus, &enable_agc,
- sizeof(enable_agc));
- if (result != noErr) {
- RTCLogError(@"Failed to enable the built-in AGC. "
- "Error=%ld.",
- (long)result);
- RTC_HISTOGRAM_COUNTS_SPARSE_100000(
- "WebRTC.Audio.SetAGCStateErrorCode", (-1) * result);
- }
- result = GetAGCState(vpio_unit_, &agc_is_enabled);
- if (result != noErr) {
- RTCLogError(@"Failed to get AGC state (2nd attempt). "
- "Error=%ld.",
- (long)result);
- RTC_HISTOGRAM_COUNTS_SPARSE_100000(
- "WebRTC.Audio.GetAGCStateErrorCode2", (-1) * result);
- }
- }
-
- // Track if the built-in AGC was enabled by default (as it should) or not.
- RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCWasEnabledByDefault",
- agc_was_enabled_by_default);
- RTCLog(@"WebRTC.Audio.BuiltInAGCWasEnabledByDefault: %d",
- agc_was_enabled_by_default);
- // As a final step, add an UMA histogram for tracking the AGC state.
- // At this stage, the AGC should be enabled, and if it is not, more work is
- // needed to find out the root cause.
- RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCIsEnabled", agc_is_enabled);
- RTCLog(@"WebRTC.Audio.BuiltInAGCIsEnabled: %u",
- static_cast<unsigned int>(agc_is_enabled));
-
- state_ = kInitialized;
- return true;
-}
-
-bool VoiceProcessingAudioUnit::Start() {
- RTC_DCHECK_GE(state_, kUninitialized);
- RTCLog(@"Starting audio unit.");
-
- OSStatus result = AudioOutputUnitStart(vpio_unit_);
- if (result != noErr) {
- RTCLogError(@"Failed to start audio unit. Error=%ld", (long)result);
- return false;
- } else {
- RTCLog(@"Started audio unit");
- }
- state_ = kStarted;
- return true;
-}
-
-bool VoiceProcessingAudioUnit::Stop() {
- RTC_DCHECK_GE(state_, kUninitialized);
- RTCLog(@"Stopping audio unit.");
-
- OSStatus result = AudioOutputUnitStop(vpio_unit_);
- if (result != noErr) {
- RTCLogError(@"Failed to stop audio unit. Error=%ld", (long)result);
- return false;
- } else {
- RTCLog(@"Stopped audio unit");
- }
-
- state_ = kInitialized;
- return true;
-}
-
-bool VoiceProcessingAudioUnit::Uninitialize() {
- RTC_DCHECK_GE(state_, kUninitialized);
- RTCLog(@"Unintializing audio unit.");
-
- OSStatus result = AudioUnitUninitialize(vpio_unit_);
- if (result != noErr) {
- RTCLogError(@"Failed to uninitialize audio unit. Error=%ld", (long)result);
- return false;
- } else {
- RTCLog(@"Uninitialized audio unit.");
- }
-
- state_ = kUninitialized;
- return true;
-}
-
-OSStatus VoiceProcessingAudioUnit::Render(AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 output_bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- RTC_DCHECK(vpio_unit_) << "Init() not called.";
-
- OSStatus result = AudioUnitRender(vpio_unit_, flags, time_stamp,
- output_bus_number, num_frames, io_data);
- if (result != noErr) {
- RTCLogError(@"Failed to render audio unit. Error=%ld", (long)result);
- }
- return result;
-}
-
-OSStatus VoiceProcessingAudioUnit::OnGetPlayoutData(
- void* in_ref_con,
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- VoiceProcessingAudioUnit* audio_unit =
- static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
- return audio_unit->NotifyGetPlayoutData(flags, time_stamp, bus_number,
- num_frames, io_data);
-}
-
-OSStatus VoiceProcessingAudioUnit::OnDeliverRecordedData(
- void* in_ref_con,
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- VoiceProcessingAudioUnit* audio_unit =
- static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
- return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
- num_frames, io_data);
-}
-
-OSStatus VoiceProcessingAudioUnit::NotifyGetPlayoutData(
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- return observer_->OnGetPlayoutData(flags, time_stamp, bus_number, num_frames,
- io_data);
-}
-
-OSStatus VoiceProcessingAudioUnit::NotifyDeliverRecordedData(
- AudioUnitRenderActionFlags* flags,
- const AudioTimeStamp* time_stamp,
- UInt32 bus_number,
- UInt32 num_frames,
- AudioBufferList* io_data) {
- return observer_->OnDeliverRecordedData(flags, time_stamp, bus_number,
- num_frames, io_data);
-}
-
-AudioStreamBasicDescription VoiceProcessingAudioUnit::GetFormat(
- Float64 sample_rate) const {
- // Set the application formats for input and output:
- // - use same format in both directions
- // - avoid resampling in the I/O unit by using the hardware sample rate
- // - linear PCM => noncompressed audio data format with one frame per packet
- // - no need to specify interleaving since only mono is supported
- AudioStreamBasicDescription format;
- RTC_DCHECK_EQ(1, kRTCAudioSessionPreferredNumberOfChannels);
- format.mSampleRate = sample_rate;
- format.mFormatID = kAudioFormatLinearPCM;
- format.mFormatFlags =
- kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
- format.mBytesPerPacket = kBytesPerSample;
- format.mFramesPerPacket = 1; // uncompressed.
- format.mBytesPerFrame = kBytesPerSample;
- format.mChannelsPerFrame = kRTCAudioSessionPreferredNumberOfChannels;
- format.mBitsPerChannel = 8 * kBytesPerSample;
- return format;
-}
-
-void VoiceProcessingAudioUnit::DisposeAudioUnit() {
- if (vpio_unit_) {
- switch (state_) {
- case kStarted:
- Stop();
- // Fall through.
- RTC_FALLTHROUGH();
- case kInitialized:
- Uninitialize();
- break;
- case kUninitialized:
- RTC_FALLTHROUGH();
- case kInitRequired:
- break;
- }
-
- RTCLog(@"Disposing audio unit.");
- OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
- if (result != noErr) {
- RTCLogError(@"AudioComponentInstanceDispose failed. Error=%ld.",
- (long)result);
- }
- vpio_unit_ = nullptr;
- }
-}
-
-} // namespace webrtc
