zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Moving iOS Audio Device to sdk.

Browse Source 
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
This commit is contained in:
Peter Hanspers
2018-05-15 10:22:36 +02:00
committed by Commit Bot
parent ef00310a4a
commit 8d95e3b211
19 changed files with 3863 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
modules/audio_device/BUILD.gn
View File

@ -94,13 +94,7 @@ if (rtc_include_internal_audio_device && is_ios) {
}
rtc_source_set("audio_device_api") {
visibility = [
":audio_device",
":audio_device_buffer",
":audio_device_impl",
":audio_device_ios_objc",
":audio_device_generic",
]
visibility = [ "*" ]
sources = [
"include/audio_device.h",
"include/audio_device_defines.h",

104
sdk/BUILD.gn
View File

@ -94,12 +94,85 @@ if (is_ios || is_mac) {
if (!build_with_chromium) {
if (is_ios) {
rtc_static_library("native_api_audio_device_module") {
visibility = [ "*" ]
sources = [
"objc/Framework/Native/api/audio_device_module.h",
"objc/Framework/Native/api/audio_device_module.mm",
]
deps = [
":audio_device",
"../modules/audio_device:audio_device_api",
"../modules/audio_device:audio_device_generic",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_api",
]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin
# (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_source_set("audio_session_observer") {
visibility = [ ":*" ]
sources = [
"objc/Framework/Native/src/audio/audio_session_observer.h",
]
deps = [
"../rtc_base:rtc_base",
]
}
rtc_static_library("audio_device") {
visibility = [ "*" ]
sources = [
"objc/Framework/Native/src/audio/audio_device_ios.h",
"objc/Framework/Native/src/audio/audio_device_ios.mm",
"objc/Framework/Native/src/audio/audio_device_module_ios.h",
"objc/Framework/Native/src/audio/audio_device_module_ios.mm",
"objc/Framework/Native/src/audio/voice_processing_audio_unit.h",
"objc/Framework/Native/src/audio/voice_processing_audio_unit.mm",
]
deps = [
":audio_objc",
":audio_session_observer",
":common_objc",
"../api:array_view",
"../modules/audio_device:audio_device_api",
"../modules/audio_device:audio_device_buffer",
"../modules/audio_device:audio_device_generic",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base/system:fallthrough",
"../sdk:common_objc",
"../system_wrappers:metrics_api",
]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin
# (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("audio_objc") {
sources = [
"objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm",
"objc/Framework/Classes/Audio/RTCAudioSession+Private.h",
"objc/Framework/Classes/Audio/RTCAudioSession.mm",
"objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m",
"objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h",
"objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm",
"objc/Framework/Headers/WebRTC/RTCAudioSession.h",
"objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
]
@ -108,8 +181,10 @@ if (is_ios || is_mac) {
public_configs = [ ":common_config_objc" ]
deps = [
":audio_session_observer",
":common_objc",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
@ -689,6 +764,10 @@ if (is_ios || is_mac) {
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
if (is_ios) {
deps += [ ":native_api_audio_device_module" ]
}
}
if (rtc_include_tests) {
@ -710,15 +789,28 @@ if (is_ios || is_mac) {
"objc/Framework/UnitTests/frame_buffer_helpers.mm",
]
# TODO(peterhanspers): Reenable these tests on simulator.
# See bugs.webrtc.org/7812
if (!use_ios_simulator) {
sources += [
"objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm",
"objc/Framework/UnitTests/RTCAudioDevice_xctest.mm",
]
}
deps = [
":audio_device",
":audio_objc",
":common_objc",
":framework_objc",
":native_api",
":native_api_audio_device_module",
":native_video",
":videocapture_objc",
":videoframebuffer_objc",
":videosource_objc",
":videotoolbox_objc",
"../../system_wrappers:system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../api/video:video_frame_i420",
"../common_video:common_video",
@ -750,8 +842,12 @@ if (is_ios || is_mac) {
}
bundle_data("sdk_unittests_bundle_data") {
# Sample video taken from https://media.xiph.org/video/derf/
sources = [
"objc/Framework/UnitTests/audio_short16.pcm",
"objc/Framework/UnitTests/audio_short44.pcm",
"objc/Framework/UnitTests/audio_short48.pcm",
# Sample video taken from https://media.xiph.org/video/derf/
"objc/Framework/UnitTests/foreman.mp4",
]
outputs = [
@ -772,6 +868,12 @@ if (is_ios || is_mac) {
":sdk_unittests_sources",
]
ldflags = [ "-all_load" ]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin
# (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}

34
sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h Normal file
View File

@ -0,0 +1,34 @@
/*
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
NS_ASSUME_NONNULL_BEGIN
namespace webrtc {
class AudioSessionObserver;
}
/** Adapter that forwards RTCAudioSessionDelegate calls to the appropriate
* methods on the AudioSessionObserver.
*/
@interface RTCNativeAudioSessionDelegateAdapter : 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
NS_ASSUME_NONNULL_END

89
sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm Normal file
View File

@ -0,0 +1,89 @@
/*
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "RTCNativeAudioSessionDelegateAdapter.h"
#include "sdk/objc/Framework/Native/src/audio/audio_session_observer.h"
#import "WebRTC/RTCLogging.h"
@implementation RTCNativeAudioSessionDelegateAdapter {
webrtc::AudioSessionObserver *_observer;
}
- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
RTC_DCHECK(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

16
sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm
View File

@ -38,6 +38,10 @@
#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
#endif
#if defined(WEBRTC_IOS)
#import "sdk/objc/Framework/Native/api/audio_device_module.h"
#endif
// Adding the nogncheck to disable the including header check.
// The no-media version PeerConnectionFactory doesn't depend on media related
// C++ target.
@ -55,6 +59,14 @@
@synthesize nativeFactory = _nativeFactory;
- (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
#if defined(WEBRTC_IOS)
return webrtc::CreateAudioDeviceModule();
#else
return nullptr;
#endif
}
- (instancetype)init {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
@ -65,7 +77,7 @@
[[RTCVideoEncoderFactoryH264 alloc] init])
nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory(
[[RTCVideoDecoderFactoryH264 alloc] init])
audioDeviceModule:nullptr
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr];
#endif
}
@ -87,7 +99,7 @@
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
nativeVideoEncoderFactory:std::move(native_encoder_factory)
nativeVideoDecoderFactory:std::move(native_decoder_factory)
audioDeviceModule:nullptr
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr];
#endif
}

24
sdk/objc/Framework/Native/api/audio_device_module.h Normal file
View File

@ -0,0 +1,24 @@
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
#include <memory>
#include "modules/audio_device/include/audio_device.h"
namespace webrtc {
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule();
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_

30
sdk/objc/Framework/Native/api/audio_device_module.mm Normal file
View File

@ -0,0 +1,30 @@
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "audio_device_module.h"
#include "rtc_base/logging.h"
#include "rtc_base/refcountedobject.h"
#include "sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h"
namespace webrtc {
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule() {
RTC_LOG(INFO) << __FUNCTION__;
#if defined(WEBRTC_IOS)
return new rtc::RefCountedObject<ios_adm::AudioDeviceModuleIOS>();
#else
RTC_LOG(LERROR)
<< "current platform is not supported => this module will self destruct!";
return nullptr;
#endif
}
}

293
sdk/objc/Framework/Native/src/audio/audio_device_ios.h Normal file
View File

@ -0,0 +1,293 @@
/*
* 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 "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
#include "modules/audio_device/audio_device_generic.h"
#include "audio_session_observer.h"
#include "voice_processing_audio_unit.h"
#include "rtc_base/buffer.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter);
namespace webrtc {
class FineAudioBuffer;
namespace ios_adm {
// 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;
bool IsInterrupted();
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();
// 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.
RTCNativeAudioSessionDelegateAdapter* 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_);
};
} // namespace ios_adm
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_

1104
sdk/objc/Framework/Native/src/audio/audio_device_ios.mm Normal file
View File

File diff suppressed because it is too large Load Diff

143
sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h Normal file
View File

@ -0,0 +1,143 @@
/*
* 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 SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_
#define SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_
#include <memory>
#include "audio_device_ios.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/criticalsection.h"
namespace webrtc {
class AudioDeviceGeneric;
namespace ios_adm {
class AudioDeviceModuleIOS : public AudioDeviceModule {
public:
int32_t AttachAudioBuffer();
AudioDeviceModuleIOS();
~AudioDeviceModuleIOS() override;
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
// Full-duplex transportation of PCM audio
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
// Main initializaton and termination
int32_t Init() override;
int32_t Terminate() override;
bool Initialized() const override;
// Device enumeration
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;
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(WindowsDeviceType device) override;
// Audio transport initialization
int32_t PlayoutIsAvailable(bool* available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool* available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Recording() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
// Speaker volume controls
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;
// Microphone volume controls
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;
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool* available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool* enabled) const override;
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool* available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool* enabled) const override;
// Stereo support
int32_t StereoPlayoutIsAvailable(bool* available) const override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool* enabled) const override;
int32_t StereoRecordingIsAvailable(bool* available) const override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool* enabled) const override;
// Delay information and control
int32_t PlayoutDelay(uint16_t* delayMS) const override;
bool BuiltInAECIsAvailable() const override;
int32_t EnableBuiltInAEC(bool enable) override;
bool BuiltInAGCIsAvailable() const override;
int32_t EnableBuiltInAGC(bool enable) override;
bool BuiltInNSIsAvailable() const override;
int32_t EnableBuiltInNS(bool enable) override;
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override;
int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
private:
bool initialized_ = false;
std::unique_ptr<AudioDeviceIOS> audio_device_;
std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
};
} // namespace ios_adm
} // namespace webrtc
#endif // SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_

673
sdk/objc/Framework/Native/src/audio/audio_device_module_ios.mm Normal file
View File

@ -0,0 +1,673 @@
/*
* 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.
*/
#include "audio_device_module_ios.h"
#include "modules/audio_device/audio_device_config.h"
#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/refcount.h"
#include "rtc_base/refcountedobject.h"
#include "system_wrappers/include/metrics.h"
#if defined(WEBRTC_IOS)
#include "audio_device_ios.h"
#endif
#define CHECKinitialized_() \
{ \
if (!initialized_) { \
return -1; \
}; \
}
#define CHECKinitialized__BOOL() \
{ \
if (!initialized_) { \
return false; \
}; \
}
namespace webrtc {
namespace ios_adm {
AudioDeviceModuleIOS::AudioDeviceModuleIOS() {
RTC_LOG(INFO) << "current platform is IOS";
RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
}
int32_t AudioDeviceModuleIOS::AttachAudioBuffer() {
RTC_LOG(INFO) << __FUNCTION__;
audio_device_->AttachAudioBuffer(audio_device_buffer_.get());
return 0;
}
AudioDeviceModuleIOS::~AudioDeviceModuleIOS() {
RTC_LOG(INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleIOS::ActiveAudioLayer(AudioLayer* audioLayer) const {
RTC_LOG(INFO) << __FUNCTION__;
AudioLayer activeAudio;
if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
return -1;
}
*audioLayer = activeAudio;
return 0;
}
int32_t AudioDeviceModuleIOS::Init() {
RTC_LOG(INFO) << __FUNCTION__;
if (initialized_)
return 0;
audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer());
audio_device_.reset(new ios_adm::AudioDeviceIOS());
RTC_CHECK(audio_device_);
this->AttachAudioBuffer();
AudioDeviceGeneric::InitStatus status = audio_device_->Init();
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.InitializationResult", static_cast<int>(status),
static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
if (status != AudioDeviceGeneric::InitStatus::OK) {
RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
return -1;
}
initialized_ = true;
return 0;
}
int32_t AudioDeviceModuleIOS::Terminate() {
RTC_LOG(INFO) << __FUNCTION__;
if (!initialized_)
return 0;
if (audio_device_->Terminate() == -1) {
return -1;
}
initialized_ = false;
return 0;
}
bool AudioDeviceModuleIOS::Initialized() const {
RTC_LOG(INFO) << __FUNCTION__ << ": " << initialized_;
return initialized_;
}
int32_t AudioDeviceModuleIOS::InitSpeaker() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitSpeaker();
}
int32_t AudioDeviceModuleIOS::InitMicrophone() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitMicrophone();
}
int32_t AudioDeviceModuleIOS::SpeakerVolumeIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetSpeakerVolume(uint32_t volume) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerVolume(volume);
}
int32_t AudioDeviceModuleIOS::SpeakerVolume(uint32_t* volume) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->SpeakerVolume(level) == -1) {
return -1;
}
*volume = level;
RTC_LOG(INFO) << "output: " << *volume;
return 0;
}
bool AudioDeviceModuleIOS::SpeakerIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->SpeakerIsInitialized();
RTC_LOG(INFO) << "output: " << isInitialized;
return isInitialized;
}
bool AudioDeviceModuleIOS::MicrophoneIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->MicrophoneIsInitialized();
RTC_LOG(INFO) << "output: " << isInitialized;
return isInitialized;
}
int32_t AudioDeviceModuleIOS::MaxSpeakerVolume(uint32_t* maxVolume) const {
CHECKinitialized_();
uint32_t maxVol = 0;
if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
return -1;
}
*maxVolume = maxVol;
return 0;
}
int32_t AudioDeviceModuleIOS::MinSpeakerVolume(uint32_t* minVolume) const {
CHECKinitialized_();
uint32_t minVol = 0;
if (audio_device_->MinSpeakerVolume(minVol) == -1) {
return -1;
}
*minVolume = minVol;
return 0;
}
int32_t AudioDeviceModuleIOS::SpeakerMuteIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetSpeakerMute(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerMute(enable);
}
int32_t AudioDeviceModuleIOS::SpeakerMute(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->SpeakerMute(muted) == -1) {
return -1;
}
*enabled = muted;
RTC_LOG(INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleIOS::MicrophoneMuteIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetMicrophoneMute(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneMute(enable));
}
int32_t AudioDeviceModuleIOS::MicrophoneMute(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->MicrophoneMute(muted) == -1) {
return -1;
}
*enabled = muted;
RTC_LOG(INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleIOS::MicrophoneVolumeIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetMicrophoneVolume(uint32_t volume) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneVolume(volume));
}
int32_t AudioDeviceModuleIOS::MicrophoneVolume(uint32_t* volume) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->MicrophoneVolume(level) == -1) {
return -1;
}
*volume = level;
RTC_LOG(INFO) << "output: " << *volume;
return 0;
}
int32_t AudioDeviceModuleIOS::StereoRecordingIsAvailable(
bool* available) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetStereoRecording(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->RecordingIsInitialized()) {
RTC_LOG(WARNING) << "recording in stereo is not supported";
return -1;
}
if (audio_device_->SetStereoRecording(enable) == -1) {
RTC_LOG(WARNING) << "failed to change stereo recording";
return -1;
}
int8_t nChannels(1);
if (enable) {
nChannels = 2;
}
audio_device_buffer_.get()->SetRecordingChannels(nChannels);
return 0;
}
int32_t AudioDeviceModuleIOS::StereoRecording(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoRecording(stereo) == -1) {
return -1;
}
*enabled = stereo;
RTC_LOG(INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleIOS::StereoPlayoutIsAvailable(bool* available) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetStereoPlayout(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
<< "unable to set stereo mode while playing side is initialized";
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
RTC_LOG(WARNING) << "stereo playout is not supported";
return -1;
}
int8_t nChannels(1);
if (enable) {
nChannels = 2;
}
audio_device_buffer_.get()->SetPlayoutChannels(nChannels);
return 0;
}
int32_t AudioDeviceModuleIOS::StereoPlayout(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoPlayout(stereo) == -1) {
return -1;
}
*enabled = stereo;
RTC_LOG(INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleIOS::PlayoutIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::RecordingIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::MaxMicrophoneVolume(uint32_t* maxVolume) const {
CHECKinitialized_();
uint32_t maxVol(0);
if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
return -1;
}
*maxVolume = maxVol;
return 0;
}
int32_t AudioDeviceModuleIOS::MinMicrophoneVolume(uint32_t* minVolume) const {
CHECKinitialized_();
uint32_t minVol(0);
if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
return -1;
}
*minVolume = minVol;
return 0;
}
int16_t AudioDeviceModuleIOS::PlayoutDevices() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
RTC_LOG(INFO) << "output: " << nPlayoutDevices;
return (int16_t)(nPlayoutDevices);
}
int32_t AudioDeviceModuleIOS::SetPlayoutDevice(uint16_t index) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(index);
}
int32_t AudioDeviceModuleIOS::SetPlayoutDevice(WindowsDeviceType device) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(device);
}
int32_t AudioDeviceModuleIOS::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
}
if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
return -1;
}
if (name != NULL) {
RTC_LOG(INFO) << "output: name = " << name;
}
if (guid != NULL) {
RTC_LOG(INFO) << "output: guid = " << guid;
}
return 0;
}
int32_t AudioDeviceModuleIOS::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
}
if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
return -1;
}
if (name != NULL) {
RTC_LOG(INFO) << "output: name = " << name;
}
if (guid != NULL) {
RTC_LOG(INFO) << "output: guid = " << guid;
}
return 0;
}
int16_t AudioDeviceModuleIOS::RecordingDevices() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nRecordingDevices = audio_device_->RecordingDevices();
RTC_LOG(INFO) << "output: " << nRecordingDevices;
return (int16_t)nRecordingDevices;
}
int32_t AudioDeviceModuleIOS::SetRecordingDevice(uint16_t index) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetRecordingDevice(index);
}
int32_t AudioDeviceModuleIOS::SetRecordingDevice(WindowsDeviceType device) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetRecordingDevice(device);
}
int32_t AudioDeviceModuleIOS::InitPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (PlayoutIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::InitRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (RecordingIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::PlayoutIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->PlayoutIsInitialized();
}
bool AudioDeviceModuleIOS::RecordingIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->RecordingIsInitialized();
}
int32_t AudioDeviceModuleIOS::StartPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (Playing()) {
return 0;
}
audio_device_buffer_.get()->StartPlayout();
int32_t result = audio_device_->StartPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::StopPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopPlayout();
audio_device_buffer_.get()->StopPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::Playing() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Playing();
}
int32_t AudioDeviceModuleIOS::StartRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (Recording()) {
return 0;
}
audio_device_buffer_.get()->StartRecording();
int32_t result = audio_device_->StartRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::StopRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopRecording();
audio_device_buffer_.get()->StopRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::Recording() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Recording();
}
int32_t AudioDeviceModuleIOS::RegisterAudioCallback(
AudioTransport* audioCallback) {
RTC_LOG(INFO) << __FUNCTION__;
return audio_device_buffer_.get()->RegisterAudioCallback(audioCallback);
}
int32_t AudioDeviceModuleIOS::PlayoutDelay(uint16_t* delayMS) const {
CHECKinitialized_();
uint16_t delay = 0;
if (audio_device_->PlayoutDelay(delay) == -1) {
RTC_LOG(LERROR) << "failed to retrieve the playout delay";
return -1;
}
*delayMS = delay;
return 0;
}
bool AudioDeviceModuleIOS::BuiltInAECIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAECIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInAEC(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAEC(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleIOS::BuiltInAGCIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInAGC(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAGC(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleIOS::BuiltInNSIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInNSIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInNS(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInNS(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
#if defined(WEBRTC_IOS)
int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
AudioParameters* params) const {
RTC_LOG(INFO) << __FUNCTION__;
int r = audio_device_->GetPlayoutAudioParameters(params);
RTC_LOG(INFO) << "output: " << r;
return r;
}
int AudioDeviceModuleIOS::GetRecordAudioParameters(
AudioParameters* params) const {
RTC_LOG(INFO) << __FUNCTION__;
int r = audio_device_->GetRecordAudioParameters(params);
RTC_LOG(INFO) << "output: " << r;
return r;
}
#endif // WEBRTC_IOS
}
}

42
sdk/objc/Framework/Native/src/audio/audio_session_observer.h Normal file
View File

@ -0,0 +1,42 @@
/*
* 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/asyncinvoker.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_

139
sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.h Normal file
View File

@ -0,0 +1,139 @@
/*
* 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 {
namespace ios_adm {
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 ios_adm
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_

470
sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.mm Normal file
View File

@ -0,0 +1,470 @@
/*
* 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 "voice_processing_audio_unit.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/fallthrough.h"
#include "system_wrappers/include/metrics.h"
#import "WebRTC/RTCLogging.h"
#import "sdk/objc/Framework/Headers/WebRTC/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 {
namespace ios_adm {
// 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 ios_adm
} // namespace webrtc

592
sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm Normal file
View File

@ -0,0 +1,592 @@
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import <XCTest/XCTest.h>
#if defined(WEBRTC_IOS)
#import "sdk/objc/Framework/Native/api/audio_device_module.h"
#endif
#include "system_wrappers/include/event_wrapper.h"
#include "rtc_base/scoped_ref_ptr.h"
typedef int32_t(^NeedMorePlayDataBlock)(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);
typedef int32_t(^RecordedDataIsAvailableBlock)(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);
// This class implements the AudioTransport API and forwards all methods to the appropriate blocks.
class MockAudioTransport : public webrtc::AudioTransport {
public:
MockAudioTransport() {}
~MockAudioTransport() {}
void expectNeedMorePlayData(NeedMorePlayDataBlock block) {
needMorePlayDataBlock = block;
}
void expectRecordedDataIsAvailable(RecordedDataIsAvailableBlock block) {
recordedDataIsAvailableBlock = block;
}
int32_t NeedMorePlayData(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) {
return needMorePlayDataBlock(nSamples,
nBytesPerSample,
nChannels,
samplesPerSec,
audioSamples,
nSamplesOut,
elapsed_time_ms,
ntp_time_ms);
}
int32_t RecordedDataIsAvailable(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) {
return recordedDataIsAvailableBlock(audioSamples,
nSamples,
nBytesPerSample,
nChannels,
samplesPerSec,
totalDelayMS,
clockDrift,
currentMicLevel,
keyPressed,
newMicLevel);
}
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
}
private:
NeedMorePlayDataBlock needMorePlayDataBlock;
RecordedDataIsAvailableBlock recordedDataIsAvailableBlock;
};
// Number of callbacks (input or output) the tests waits for before we set
// an event indicating that the test was OK.
static const NSUInteger kNumCallbacks = 10;
// Max amount of time we wait for an event to be set while counting callbacks.
static const NSTimeInterval kTestTimeOutInSec = 20.0;
// Number of bits per PCM audio sample.
static const NSUInteger kBitsPerSample = 16;
// Number of bytes per PCM audio sample.
static const NSUInteger kBytesPerSample = kBitsPerSample / 8;
// Average number of audio callbacks per second assuming 10ms packet size.
static const NSUInteger kNumCallbacksPerSecond = 100;
// Play out a test file during this time (unit is in seconds).
static const NSUInteger kFilePlayTimeInSec = 15;
// Run the full-duplex test during this time (unit is in seconds).
// Note that first |kNumIgnoreFirstCallbacks| are ignored.
static const NSUInteger 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 NSUInteger kNumIgnoreFirstCallbacks = 50;
@interface RTCAudioDeviceModuleTests : XCTestCase {
rtc::scoped_refptr<webrtc::AudioDeviceModule> audioDeviceModule;
webrtc::AudioParameters playoutParameters;
webrtc::AudioParameters recordParameters;
MockAudioTransport mock;
}
@end
@implementation RTCAudioDeviceModuleTests
- (void)setUp {
[super setUp];
audioDeviceModule = webrtc::CreateAudioDeviceModule();
XCTAssertEqual(0, audioDeviceModule->Init());
XCTAssertEqual(0, audioDeviceModule->GetPlayoutAudioParameters(&playoutParameters));
XCTAssertEqual(0, audioDeviceModule->GetRecordAudioParameters(&recordParameters));
}
- (void)tearDown {
XCTAssertEqual(0, audioDeviceModule->Terminate());
audioDeviceModule = nullptr;
[super tearDown];
}
- (void)startPlayout {
XCTAssertFalse(audioDeviceModule->Playing());
XCTAssertEqual(0, audioDeviceModule->InitPlayout());
XCTAssertTrue(audioDeviceModule->PlayoutIsInitialized());
XCTAssertEqual(0, audioDeviceModule->StartPlayout());
XCTAssertTrue(audioDeviceModule->Playing());
}
- (void)stopPlayout {
XCTAssertEqual(0, audioDeviceModule->StopPlayout());
XCTAssertFalse(audioDeviceModule->Playing());
}
- (void)startRecording{
XCTAssertFalse(audioDeviceModule->Recording());
XCTAssertEqual(0, audioDeviceModule->InitRecording());
XCTAssertTrue(audioDeviceModule->RecordingIsInitialized());
XCTAssertEqual(0, audioDeviceModule->StartRecording());
XCTAssertTrue(audioDeviceModule->Recording());
}
- (void)stopRecording{
XCTAssertEqual(0, audioDeviceModule->StopRecording());
XCTAssertFalse(audioDeviceModule->Recording());
}
- (NSURL*)fileURLForSampleRate:(int)sampleRate {
XCTAssertTrue(sampleRate == 48000 || sampleRate == 44100 || sampleRate == 16000);
NSString *filename = [NSString stringWithFormat:@"audio_short%d", sampleRate / 1000];
NSURL *url = [[NSBundle mainBundle] URLForResource:filename withExtension:@"pcm"];
XCTAssertNotNil(url);
return url;
}
#pragma mark - Tests
- (void)testConstructDestruct {
// Using the test fixture to create and destruct the audio device module.
}
- (void)testInitTerminate {
// Initialization is part of the test fixture.
XCTAssertTrue(audioDeviceModule->Initialized());
XCTAssertEqual(0, audioDeviceModule->Terminate());
XCTAssertFalse(audioDeviceModule->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.
- (void)DISABLED_testStartStopPlayout {
[self startPlayout];
[self stopPlayout];
[self startPlayout];
[self 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.
- (void)DISABLED_testStartStopRecording {
[self startRecording];
[self stopRecording];
[self startRecording];
[self 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.
- (void)testStopPlayoutRequiresInitToRestart {
XCTAssertEqual(0, audioDeviceModule->InitPlayout());
XCTAssertEqual(0, audioDeviceModule->StartPlayout());
XCTAssertEqual(0, audioDeviceModule->StopPlayout());
XCTAssertFalse(audioDeviceModule->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.
- (void)DISABLED_testStartPlayoutOnTwoInstances {
// Create and initialize a second/extra ADM instance. The default ADM is
// created by the test harness.
rtc::scoped_refptr<webrtc::AudioDeviceModule> secondAudioDeviceModule =
webrtc::CreateAudioDeviceModule();
XCTAssertNotEqual(secondAudioDeviceModule.get(), nullptr);
XCTAssertEqual(0, secondAudioDeviceModule->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.
mock.expectNeedMorePlayData(^int32_t(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) {
nSamplesOut = nSamples;
XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, playoutParameters.channels());
XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
XCTAssertNotEqual((void*)NULL, audioSamples);
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self 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).
XCTAssertEqual(0, secondAudioDeviceModule->InitPlayout());
XCTAssertTrue(secondAudioDeviceModule->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.
XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
playoutExpectation.expectedFulfillmentCount = kNumCallbacks;
MockAudioTransport mock2;
mock2.expectNeedMorePlayData(^int32_t(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) {
nSamplesOut = nSamples;
XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, playoutParameters.channels());
XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
XCTAssertNotEqual((void*)NULL, audioSamples);
[playoutExpectation fulfill];
return 0;
});
XCTAssertEqual(0, secondAudioDeviceModule->RegisterAudioCallback(&mock2));
XCTAssertEqual(0, secondAudioDeviceModule->StartPlayout());
XCTAssertTrue(secondAudioDeviceModule->Playing());
[self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
XCTAssertEqual(0, secondAudioDeviceModule->StopPlayout());
XCTAssertFalse(secondAudioDeviceModule->Playing());
XCTAssertFalse(secondAudioDeviceModule->PlayoutIsInitialized());
XCTAssertEqual(0, secondAudioDeviceModule->Terminate());
}
// Start playout and verify that the native audio layer starts asking for real
// audio samples to play out using the NeedMorePlayData callback.
- (void)testStartPlayoutVerifyCallbacks {
XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
playoutExpectation.expectedFulfillmentCount = kNumCallbacks;
mock.expectNeedMorePlayData(^int32_t(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) {
nSamplesOut = nSamples;
XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, playoutParameters.channels());
XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
XCTAssertNotEqual((void*)NULL, audioSamples);
[playoutExpectation fulfill];
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self startPlayout];
[self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
[self stopPlayout];
}
// Start recording and verify that the native audio layer starts feeding real
// audio samples via the RecordedDataIsAvailable callback.
- (void)testStartRecordingVerifyCallbacks {
XCTestExpectation *recordExpectation =
[self expectationWithDescription:@"RecordedDataIsAvailable"];
recordExpectation.expectedFulfillmentCount = kNumCallbacks;
mock.expectRecordedDataIsAvailable(^(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) {
XCTAssertNotEqual((void*)NULL, audioSamples);
XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, recordParameters.channels());
XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
XCTAssertEqual(0, clockDrift);
XCTAssertEqual(0u, currentMicLevel);
XCTAssertFalse(keyPressed);
[recordExpectation fulfill];
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self startRecording];
[self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
[self stopRecording];
}
// Start playout and recording (full-duplex audio) and verify that audio is
// active in both directions.
- (void)testStartPlayoutAndRecordingVerifyCallbacks {
XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
__block NSUInteger callbackCount = 0;
XCTestExpectation *recordExpectation =
[self expectationWithDescription:@"RecordedDataIsAvailable"];
recordExpectation.expectedFulfillmentCount = kNumCallbacks;
mock.expectNeedMorePlayData(^int32_t(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) {
nSamplesOut = nSamples;
XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, playoutParameters.channels());
XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
XCTAssertNotEqual((void*)NULL, audioSamples);
if (callbackCount++ >= kNumCallbacks) {
[playoutExpectation fulfill];
}
return 0;
});
mock.expectRecordedDataIsAvailable(^(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) {
XCTAssertNotEqual((void*)NULL, audioSamples);
XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
XCTAssertEqual(nBytesPerSample, kBytesPerSample);
XCTAssertEqual(nChannels, recordParameters.channels());
XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
XCTAssertEqual(0, clockDrift);
XCTAssertEqual(0u, currentMicLevel);
XCTAssertFalse(keyPressed);
[recordExpectation fulfill];
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self startPlayout];
[self startRecording];
[self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
[self stopRecording];
[self 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.
- (void)testRunPlayoutWithFileAsSource {
XCTAssertEqual(1u, playoutParameters.channels());
// Using XCTestExpectation to count callbacks is very slow.
XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
const int expectedCallbackCount = kFilePlayTimeInSec * kNumCallbacksPerSecond;
__block int callbackCount = 0;
NSURL *fileURL = [self fileURLForSampleRate:playoutParameters.sample_rate()];
NSInputStream *inputStream = [[NSInputStream alloc] initWithURL:fileURL];
mock.expectNeedMorePlayData(^int32_t(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) {
[inputStream read:(uint8_t *)audioSamples maxLength:nSamples*nBytesPerSample*nChannels];
nSamplesOut = nSamples;
if (callbackCount++ == expectedCallbackCount) {
[playoutExpectation fulfill];
}
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self startPlayout];
NSTimeInterval waitTimeout = kFilePlayTimeInSec * 2.0;
[self waitForExpectationsWithTimeout:waitTimeout handler:nil];
[self stopPlayout];
}
- (void)testDevices {
// Device enumeration is not supported. Verify fixed values only.
XCTAssertEqual(1, audioDeviceModule->PlayoutDevices());
XCTAssertEqual(1, audioDeviceModule->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.
- (void)testRunPlayoutAndRecordingInFullDuplex {
XCTAssertEqual(recordParameters.channels(), playoutParameters.channels());
XCTAssertEqual(recordParameters.sample_rate(), playoutParameters.sample_rate());
XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
__block NSUInteger playoutCallbacks = 0;
NSUInteger expectedPlayoutCallbacks = kFullDuplexTimeInSec * kNumCallbacksPerSecond;
// FIFO queue and measurements
NSMutableArray *fifoBuffer = [NSMutableArray arrayWithCapacity:20];
__block NSUInteger fifoMaxSize = 0;
__block NSUInteger fifoTotalWrittenElements = 0;
__block NSUInteger fifoWriteCount = 0;
mock.expectRecordedDataIsAvailable(^(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) {
if (fifoWriteCount++ < kNumIgnoreFirstCallbacks) {
return 0;
}
NSData *data = [NSData dataWithBytes:audioSamples length:nSamples*nBytesPerSample*nChannels];
@synchronized(fifoBuffer) {
[fifoBuffer addObject:data];
fifoMaxSize = MAX(fifoMaxSize, fifoBuffer.count);
fifoTotalWrittenElements += fifoBuffer.count;
}
return 0;
});
mock.expectNeedMorePlayData(^int32_t(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) {
nSamplesOut = nSamples;
NSData *data;
@synchronized(fifoBuffer) {
data = fifoBuffer.firstObject;
if (data) {
[fifoBuffer removeObjectAtIndex:0];
}
}
if (data) {
memcpy(audioSamples, (char*) data.bytes, data.length);
} else {
memset(audioSamples, 0, nSamples*nBytesPerSample*nChannels);
}
if (playoutCallbacks++ == expectedPlayoutCallbacks) {
[playoutExpectation fulfill];
}
return 0;
});
XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
[self startRecording];
[self startPlayout];
NSTimeInterval waitTimeout = kFullDuplexTimeInSec * 2.0;
[self waitForExpectationsWithTimeout:waitTimeout handler:nil];
size_t fifoAverageSize =
(fifoTotalWrittenElements == 0)
? 0.0
: 0.5 + (double)fifoTotalWrittenElements / (fifoWriteCount - kNumIgnoreFirstCallbacks);
[self stopPlayout];
[self stopRecording];
XCTAssertLessThan(fifoAverageSize, 10u);
XCTAssertLessThan(fifoMaxSize, 20u);
}
@end

112
sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm Normal file
View File

@ -0,0 +1,112 @@
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import <XCTest/XCTest.h>
#import "sdk/objc/Framework/Native/src/audio/audio_device_ios.h"
#import "sdk/objc/Framework/Native/api/audio_device_module.h"
#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
@interface RTCAudioDeviceTests: XCTestCase {
rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
std::unique_ptr<webrtc::ios_adm::AudioDeviceIOS> _audio_device;
}
@property(nonatomic) RTCAudioSession *audioSession;
@end
@implementation RTCAudioDeviceTests
@synthesize audioSession = _audioSession;
- (void)setUp {
[super setUp];
_audioDeviceModule = webrtc::CreateAudioDeviceModule();
_audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS());
self.audioSession = [RTCAudioSession sharedInstance];
NSError *error = nil;
[self.audioSession lockForConfiguration];
[self.audioSession setCategory:AVAudioSessionCategoryPlayAndRecord
withOptions:0
error:&error];
XCTAssertNil(error);
[self.audioSession setMode:AVAudioSessionModeVoiceChat error:&error];
XCTAssertNil(error);
[self.audioSession setActive:YES error:&error];
XCTAssertNil(error);
[self.audioSession unlockForConfiguration];
}
- (void)tearDown {
_audio_device->Terminate();
_audio_device.reset(nullptr);
_audioDeviceModule = nullptr;
[self.audioSession notifyDidEndInterruptionWithShouldResumeSession:NO];
[super tearDown];
}
// 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.
- (void)testInterruptedAudioSession {
XCTAssertTrue(self.audioSession.isActive);
XCTAssertTrue([self.audioSession.category isEqual:AVAudioSessionCategoryPlayAndRecord] ||
[self.audioSession.category isEqual:AVAudioSessionCategoryPlayback]);
XCTAssertEqual(AVAudioSessionModeVoiceChat, self.audioSession.mode);
std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
audio_buffer.reset(new webrtc::AudioDeviceBuffer());
_audio_device->AttachAudioBuffer(audio_buffer.get());
XCTAssertEqual(webrtc::AudioDeviceGeneric::InitStatus::OK, _audio_device->Init());
XCTAssertEqual(0, _audio_device->InitPlayout());
XCTAssertEqual(0, _audio_device->StartPlayout());
// Force interruption.
[self.audioSession notifyDidBeginInterruption];
// Wait for notification to propagate.
rtc::MessageQueueManager::ProcessAllMessageQueues();
XCTAssertTrue(_audio_device->IsInterrupted());
// Force it for testing.
_audio_device->StopPlayout();
[self.audioSession notifyDidEndInterruptionWithShouldResumeSession:YES];
// Wait for notification to propagate.
rtc::MessageQueueManager::ProcessAllMessageQueues();
XCTAssertTrue(_audio_device->IsInterrupted());
_audio_device->Init();
_audio_device->InitPlayout();
XCTAssertFalse(_audio_device->IsInterrupted());
}
@end

BIN
sdk/objc/Framework/UnitTests/audio_short16.pcm Normal file
View File

Binary file not shown.

BIN
sdk/objc/Framework/UnitTests/audio_short44.pcm Normal file
View File

Binary file not shown.

BIN
sdk/objc/Framework/UnitTests/audio_short48.pcm Normal file
View File

Binary file not shown.