Moving iOS Audio Device to sdk.
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}diff --git a/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm b/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm
new file mode 100644
index 0000000..fa99d90
--- /dev/null
+++ b/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm
@@ -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