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8d95e3b211
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}
112 lines
4.3 KiB
Text
112 lines
4.3 KiB
Text
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import <XCTest/XCTest.h>
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#import "sdk/objc/Framework/Native/src/audio/audio_device_ios.h"
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#import "sdk/objc/Framework/Native/api/audio_device_module.h"
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#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
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@interface RTCAudioDeviceTests: XCTestCase {
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rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
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std::unique_ptr<webrtc::ios_adm::AudioDeviceIOS> _audio_device;
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}
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@property(nonatomic) RTCAudioSession *audioSession;
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@end
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@implementation RTCAudioDeviceTests
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@synthesize audioSession = _audioSession;
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- (void)setUp {
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[super setUp];
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_audioDeviceModule = webrtc::CreateAudioDeviceModule();
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_audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS());
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self.audioSession = [RTCAudioSession sharedInstance];
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NSError *error = nil;
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[self.audioSession lockForConfiguration];
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[self.audioSession setCategory:AVAudioSessionCategoryPlayAndRecord
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withOptions:0
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error:&error];
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XCTAssertNil(error);
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[self.audioSession setMode:AVAudioSessionModeVoiceChat error:&error];
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XCTAssertNil(error);
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[self.audioSession setActive:YES error:&error];
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XCTAssertNil(error);
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[self.audioSession unlockForConfiguration];
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}
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- (void)tearDown {
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_audio_device->Terminate();
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_audio_device.reset(nullptr);
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_audioDeviceModule = nullptr;
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[self.audioSession notifyDidEndInterruptionWithShouldResumeSession:NO];
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[super tearDown];
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}
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// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
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// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
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// AudioDeviceIOS listens to RTCAudioSession interrupted notifications by:
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// - In AudioDeviceIOS.InitPlayOrRecord registers its audio_session_observer_
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// callback with RTCAudioSession's delegate list.
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// - When RTCAudioSession receives an iOS audio interrupted notification, it
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// passes the notification to callbacks in its delegate list which sets
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// AudioDeviceIOS's is_interrupted_ flag to true.
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// - When AudioDeviceIOS.ShutdownPlayOrRecord is called, its
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// audio_session_observer_ callback is removed from RTCAudioSessions's
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// delegate list.
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// So if RTCAudioSession receives an iOS end audio interruption notification,
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// AudioDeviceIOS is not notified as its callback is not in RTCAudioSession's
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// delegate list. This causes AudioDeviceIOS's is_interrupted_ flag to be in
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// the wrong (true) state and the audio session will ignore audio changes.
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// As RTCAudioSession keeps its own interrupted state, the fix is to initialize
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// AudioDeviceIOS's is_interrupted_ flag to RTCAudioSession's isInterrupted
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// flag in AudioDeviceIOS.InitPlayOrRecord.
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- (void)testInterruptedAudioSession {
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XCTAssertTrue(self.audioSession.isActive);
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XCTAssertTrue([self.audioSession.category isEqual:AVAudioSessionCategoryPlayAndRecord] ||
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[self.audioSession.category isEqual:AVAudioSessionCategoryPlayback]);
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XCTAssertEqual(AVAudioSessionModeVoiceChat, self.audioSession.mode);
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std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
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audio_buffer.reset(new webrtc::AudioDeviceBuffer());
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_audio_device->AttachAudioBuffer(audio_buffer.get());
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XCTAssertEqual(webrtc::AudioDeviceGeneric::InitStatus::OK, _audio_device->Init());
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XCTAssertEqual(0, _audio_device->InitPlayout());
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XCTAssertEqual(0, _audio_device->StartPlayout());
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// Force interruption.
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[self.audioSession notifyDidBeginInterruption];
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// Wait for notification to propagate.
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rtc::MessageQueueManager::ProcessAllMessageQueues();
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XCTAssertTrue(_audio_device->IsInterrupted());
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// Force it for testing.
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_audio_device->StopPlayout();
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[self.audioSession notifyDidEndInterruptionWithShouldResumeSession:YES];
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// Wait for notification to propagate.
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rtc::MessageQueueManager::ProcessAllMessageQueues();
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XCTAssertTrue(_audio_device->IsInterrupted());
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_audio_device->Init();
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_audio_device->InitPlayout();
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XCTAssertFalse(_audio_device->IsInterrupted());
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}
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@end
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