Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2855023003
Cr-Commit-Position: refs/heads/master@{#18443}
This commit is contained in:
denicija
2017-06-05 05:48:47 -07:00
committed by Commit Bot
parent 90d9e10330
commit 59ee91b68a
19 changed files with 378 additions and 316 deletions

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/*
* 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 "WebRTC/RTCAudioSession.h"
#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCLogging.h"
#import "RTCAudioSession+Private.h"
@implementation RTCAudioSession (Configuration)
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
error:(NSError **)outError {
return [self setConfiguration:configuration
active:NO
shouldSetActive:NO
error:outError];
}
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
active:(BOOL)active
error:(NSError **)outError {
return [self setConfiguration:configuration
active:active
shouldSetActive:YES
error:outError];
}
#pragma mark - Private
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
active:(BOOL)active
shouldSetActive:(BOOL)shouldSetActive
error:(NSError **)outError {
NSParameterAssert(configuration);
if (outError) {
*outError = nil;
}
if (![self checkLock:outError]) {
return NO;
}
// Provide an error even if there isn't one so we can log it. We will not
// return immediately on error in this function and instead try to set
// everything we can.
NSError *error = nil;
if (self.category != configuration.category ||
self.categoryOptions != configuration.categoryOptions) {
NSError *categoryError = nil;
if (![self setCategory:configuration.category
withOptions:configuration.categoryOptions
error:&categoryError]) {
RTCLogError(@"Failed to set category: %@",
categoryError.localizedDescription);
error = categoryError;
} else {
RTCLog(@"Set category to: %@", configuration.category);
}
}
if (self.mode != configuration.mode) {
NSError *modeError = nil;
if (![self setMode:configuration.mode error:&modeError]) {
RTCLogError(@"Failed to set mode: %@",
modeError.localizedDescription);
error = modeError;
} else {
RTCLog(@"Set mode to: %@", configuration.mode);
}
}
// Sometimes category options don't stick after setting mode.
if (self.categoryOptions != configuration.categoryOptions) {
NSError *categoryError = nil;
if (![self setCategory:configuration.category
withOptions:configuration.categoryOptions
error:&categoryError]) {
RTCLogError(@"Failed to set category options: %@",
categoryError.localizedDescription);
error = categoryError;
} else {
RTCLog(@"Set category options to: %ld",
(long)configuration.categoryOptions);
}
}
if (self.preferredSampleRate != configuration.sampleRate) {
NSError *sampleRateError = nil;
if (![self setPreferredSampleRate:configuration.sampleRate
error:&sampleRateError]) {
RTCLogError(@"Failed to set preferred sample rate: %@",
sampleRateError.localizedDescription);
error = sampleRateError;
} else {
RTCLog(@"Set preferred sample rate to: %.2f",
configuration.sampleRate);
}
}
if (self.preferredIOBufferDuration != configuration.ioBufferDuration) {
NSError *bufferDurationError = nil;
if (![self setPreferredIOBufferDuration:configuration.ioBufferDuration
error:&bufferDurationError]) {
RTCLogError(@"Failed to set preferred IO buffer duration: %@",
bufferDurationError.localizedDescription);
error = bufferDurationError;
} else {
RTCLog(@"Set preferred IO buffer duration to: %f",
configuration.ioBufferDuration);
}
}
if (shouldSetActive) {
NSError *activeError = nil;
if (![self setActive:active error:&activeError]) {
RTCLogError(@"Failed to setActive to %d: %@",
active, activeError.localizedDescription);
error = activeError;
}
}
if (self.isActive &&
// TODO(tkchin): Figure out which category/mode numChannels is valid for.
[self.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
// Try to set the preferred number of hardware audio channels. These calls
// must be done after setting the audio session’s category and mode and
// activating the session.
NSInteger inputNumberOfChannels = configuration.inputNumberOfChannels;
if (self.inputNumberOfChannels != inputNumberOfChannels) {
NSError *inputChannelsError = nil;
if (![self setPreferredInputNumberOfChannels:inputNumberOfChannels
error:&inputChannelsError]) {
RTCLogError(@"Failed to set preferred input number of channels: %@",
inputChannelsError.localizedDescription);
error = inputChannelsError;
} else {
RTCLog(@"Set input number of channels to: %ld",
(long)inputNumberOfChannels);
}
}
NSInteger outputNumberOfChannels = configuration.outputNumberOfChannels;
if (self.outputNumberOfChannels != outputNumberOfChannels) {
NSError *outputChannelsError = nil;
if (![self setPreferredOutputNumberOfChannels:outputNumberOfChannels
error:&outputChannelsError]) {
RTCLogError(@"Failed to set preferred output number of channels: %@",
outputChannelsError.localizedDescription);
error = outputChannelsError;
} else {
RTCLog(@"Set output number of channels to: %ld",
(long)outputNumberOfChannels);
}
}
}
if (outError) {
*outError = error;
}
return error == nil;
}
@end

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/*
* 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 "WebRTC/RTCAudioSession.h"
#include <vector>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSessionConfiguration;
@interface RTCAudioSession ()
/** Number of times setActive:YES has succeeded without a balanced call to
* setActive:NO.
*/
@property(nonatomic, readonly) int activationCount;
/** The number of times |beginWebRTCSession| was called without a balanced call
* to |endWebRTCSession|.
*/
@property(nonatomic, readonly) int webRTCSessionCount;
/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
@property(readonly) BOOL canPlayOrRecord;
/** Tracks whether we have been sent an interruption event that hasn't been matched by either an
* interrupted end event or a foreground event.
*/
@property(nonatomic, assign) BOOL isInterrupted;
- (BOOL)checkLock:(NSError **)outError;
/** Adds the delegate to the list of delegates, and places it at the front of
* the list. This delegate will be notified before other delegates of
* audio events.
*/
- (void)pushDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Signals RTCAudioSession that a WebRTC session is about to begin and
* audio configuration is needed. Will configure the audio session for WebRTC
* if not already configured and if configuration is not delayed.
* Successful calls must be balanced by a call to endWebRTCSession.
*/
- (BOOL)beginWebRTCSession:(NSError **)outError;
/** Signals RTCAudioSession that a WebRTC session is about to end and audio
* unconfiguration is needed. Will unconfigure the audio session for WebRTC
* if this is the last unmatched call and if configuration is not delayed.
*/
- (BOOL)endWebRTCSession:(NSError **)outError;
/** Configure the audio session for WebRTC. This call will fail if the session
* is already configured. On other failures, we will attempt to restore the
* previously used audio session configuration.
* |lockForConfiguration| must be called first.
* Successful calls to configureWebRTCSession must be matched by calls to
* |unconfigureWebRTCSession|.
*/
- (BOOL)configureWebRTCSession:(NSError **)outError;
/** Unconfigures the session for WebRTC. This will attempt to restore the
* audio session to the settings used before |configureWebRTCSession| was
* called.
* |lockForConfiguration| must be called first.
*/
- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
/** Returns a configuration error with the given description. */
- (NSError *)configurationErrorWithDescription:(NSString *)description;
// Properties and methods for tests.
@property(nonatomic, readonly)
std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
- (void)notifyDidBeginInterruption;
- (void)notifyDidEndInterruptionWithShouldResumeSession:
(BOOL)shouldResumeSession;
- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
- (void)notifyMediaServicesWereLost;
- (void)notifyMediaServicesWereReset;
- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
- (void)notifyDidStartPlayOrRecord;
- (void)notifyDidStopPlayOrRecord;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 "WebRTC/RTCAudioSession.h"
#import <UIKit/UIKit.h>
#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCLogging.h"
#import "RTCAudioSession+Private.h"
NSString * const kRTCAudioSessionErrorDomain = @"org.webrtc.RTCAudioSession";
NSInteger const kRTCAudioSessionErrorLockRequired = -1;
NSInteger const kRTCAudioSessionErrorConfiguration = -2;
NSString * const kRTCAudioSessionOutputVolumeSelector = @"outputVolume";
// This class needs to be thread-safe because it is accessed from many threads.
// TODO(tkchin): Consider more granular locking. We're not expecting a lot of
// lock contention so coarse locks should be fine for now.
@implementation RTCAudioSession {
rtc::CriticalSection _crit;
AVAudioSession *_session;
volatile int _activationCount;
volatile int _lockRecursionCount;
volatile int _webRTCSessionCount;
BOOL _isActive;
BOOL _useManualAudio;
BOOL _isAudioEnabled;
BOOL _canPlayOrRecord;
BOOL _isInterrupted;
}
@synthesize session = _session;
@synthesize delegates = _delegates;
+ (instancetype)sharedInstance {
static dispatch_once_t onceToken;
static RTCAudioSession *sharedInstance = nil;
dispatch_once(&onceToken, ^{
sharedInstance = [[self alloc] init];
});
return sharedInstance;
}
- (instancetype)init {
if (self = [super init]) {
_session = [AVAudioSession sharedInstance];
NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
[center addObserver:self
selector:@selector(handleInterruptionNotification:)
name:AVAudioSessionInterruptionNotification
object:nil];
[center addObserver:self
selector:@selector(handleRouteChangeNotification:)
name:AVAudioSessionRouteChangeNotification
object:nil];
[center addObserver:self
selector:@selector(handleMediaServicesWereLost:)
name:AVAudioSessionMediaServicesWereLostNotification
object:nil];
[center addObserver:self
selector:@selector(handleMediaServicesWereReset:)
name:AVAudioSessionMediaServicesWereResetNotification
object:nil];
// Posted on the main thread when the primary audio from other applications
// starts and stops. Foreground applications may use this notification as a
// hint to enable or disable audio that is secondary.
[center addObserver:self
selector:@selector(handleSilenceSecondaryAudioHintNotification:)
name:AVAudioSessionSilenceSecondaryAudioHintNotification
object:nil];
// Also track foreground event in order to deal with interruption ended situation.
[center addObserver:self
selector:@selector(handleApplicationDidBecomeActive:)
name:UIApplicationDidBecomeActiveNotification
object:nil];
[_session addObserver:self
forKeyPath:kRTCAudioSessionOutputVolumeSelector
options:NSKeyValueObservingOptionNew | NSKeyValueObservingOptionOld
context:nil];
RTCLog(@"RTCAudioSession (%p): init.", self);
}
return self;
}
- (void)dealloc {
[[NSNotificationCenter defaultCenter] removeObserver:self];
[_session removeObserver:self forKeyPath:kRTCAudioSessionOutputVolumeSelector context:nil];
RTCLog(@"RTCAudioSession (%p): dealloc.", self);
}
- (NSString *)description {
NSString *format =
@"RTCAudioSession: {\n"
" category: %@\n"
" categoryOptions: %ld\n"
" mode: %@\n"
" isActive: %d\n"
" sampleRate: %.2f\n"
" IOBufferDuration: %f\n"
" outputNumberOfChannels: %ld\n"
" inputNumberOfChannels: %ld\n"
" outputLatency: %f\n"
" inputLatency: %f\n"
" outputVolume: %f\n"
"}";
NSString *description = [NSString stringWithFormat:format,
self.category, (long)self.categoryOptions, self.mode,
self.isActive, self.sampleRate, self.IOBufferDuration,
self.outputNumberOfChannels, self.inputNumberOfChannels,
self.outputLatency, self.inputLatency, self.outputVolume];
return description;
}
- (void)setIsActive:(BOOL)isActive {
@synchronized(self) {
_isActive = isActive;
}
}
- (BOOL)isActive {
@synchronized(self) {
return _isActive;
}
}
- (BOOL)isLocked {
return _lockRecursionCount > 0;
}
- (void)setUseManualAudio:(BOOL)useManualAudio {
@synchronized(self) {
if (_useManualAudio == useManualAudio) {
return;
}
_useManualAudio = useManualAudio;
}
[self updateCanPlayOrRecord];
}
- (BOOL)useManualAudio {
@synchronized(self) {
return _useManualAudio;
}
}
- (void)setIsAudioEnabled:(BOOL)isAudioEnabled {
@synchronized(self) {
if (_isAudioEnabled == isAudioEnabled) {
return;
}
_isAudioEnabled = isAudioEnabled;
}
[self updateCanPlayOrRecord];
}
- (BOOL)isAudioEnabled {
@synchronized(self) {
return _isAudioEnabled;
}
}
// TODO(tkchin): Check for duplicates.
- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate {
RTCLog(@"Adding delegate: (%p)", delegate);
if (!delegate) {
return;
}
@synchronized(self) {
_delegates.push_back(delegate);
[self removeZeroedDelegates];
}
}
- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate {
RTCLog(@"Removing delegate: (%p)", delegate);
if (!delegate) {
return;
}
@synchronized(self) {
_delegates.erase(std::remove(_delegates.begin(),
_delegates.end(),
delegate),
_delegates.end());
[self removeZeroedDelegates];
}
}
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wthread-safety-analysis"
- (void)lockForConfiguration {
_crit.Enter();
rtc::AtomicOps::Increment(&_lockRecursionCount);
}
- (void)unlockForConfiguration {
// Don't let threads other than the one that called lockForConfiguration
// unlock.
if (_crit.TryEnter()) {
rtc::AtomicOps::Decrement(&_lockRecursionCount);
// One unlock for the tryLock, and another one to actually unlock. If this
// was called without anyone calling lock, we will hit an assertion.
_crit.Leave();
_crit.Leave();
}
}
#pragma clang diagnostic pop
#pragma mark - AVAudioSession proxy methods
- (NSString *)category {
return self.session.category;
}
- (AVAudioSessionCategoryOptions)categoryOptions {
return self.session.categoryOptions;
}
- (NSString *)mode {
return self.session.mode;
}
- (BOOL)secondaryAudioShouldBeSilencedHint {
return self.session.secondaryAudioShouldBeSilencedHint;
}
- (AVAudioSessionRouteDescription *)currentRoute {
return self.session.currentRoute;
}
- (NSInteger)maximumInputNumberOfChannels {
return self.session.maximumInputNumberOfChannels;
}
- (NSInteger)maximumOutputNumberOfChannels {
return self.session.maximumOutputNumberOfChannels;
}
- (float)inputGain {
return self.session.inputGain;
}
- (BOOL)inputGainSettable {
return self.session.inputGainSettable;
}
- (BOOL)inputAvailable {
return self.session.inputAvailable;
}
- (NSArray<AVAudioSessionDataSourceDescription *> *)inputDataSources {
return self.session.inputDataSources;
}
- (AVAudioSessionDataSourceDescription *)inputDataSource {
return self.session.inputDataSource;
}
- (NSArray<AVAudioSessionDataSourceDescription *> *)outputDataSources {
return self.session.outputDataSources;
}
- (AVAudioSessionDataSourceDescription *)outputDataSource {
return self.session.outputDataSource;
}
- (double)sampleRate {
return self.session.sampleRate;
}
- (double)preferredSampleRate {
return self.session.preferredSampleRate;
}
- (NSInteger)inputNumberOfChannels {
return self.session.inputNumberOfChannels;
}
- (NSInteger)outputNumberOfChannels {
return self.session.outputNumberOfChannels;
}
- (float)outputVolume {
return self.session.outputVolume;
}
- (NSTimeInterval)inputLatency {
return self.session.inputLatency;
}
- (NSTimeInterval)outputLatency {
return self.session.outputLatency;
}
- (NSTimeInterval)IOBufferDuration {
return self.session.IOBufferDuration;
}
- (NSTimeInterval)preferredIOBufferDuration {
return self.session.preferredIOBufferDuration;
}
// TODO(tkchin): Simplify the amount of locking happening here. Likely that we
// can just do atomic increments / decrements.
- (BOOL)setActive:(BOOL)active
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
int activationCount = _activationCount;
if (!active && activationCount == 0) {
RTCLogWarning(@"Attempting to deactivate without prior activation.");
}
BOOL success = YES;
BOOL isActive = self.isActive;
// Keep a local error so we can log it.
NSError *error = nil;
BOOL shouldSetActive =
(active && !isActive) || (!active && isActive && activationCount == 1);
// Attempt to activate if we're not active.
// Attempt to deactivate if we're active and it's the last unbalanced call.
if (shouldSetActive) {
AVAudioSession *session = self.session;
// AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation is used to ensure
// that other audio sessions that were interrupted by our session can return
// to their active state. It is recommended for VoIP apps to use this
// option.
AVAudioSessionSetActiveOptions options =
active ? 0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
success = [session setActive:active
withOptions:options
error:&error];
if (outError) {
*outError = error;
}
}
if (success) {
if (shouldSetActive) {
self.isActive = active;
}
if (active) {
[self incrementActivationCount];
}
} else {
RTCLogError(@"Failed to setActive:%d. Error: %@",
active, error.localizedDescription);
}
// Decrement activation count on deactivation whether or not it succeeded.
if (!active) {
[self decrementActivationCount];
}
RTCLog(@"Number of current activations: %d", _activationCount);
return success;
}
- (BOOL)setCategory:(NSString *)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setCategory:category withOptions:options error:outError];
}
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setMode:mode error:outError];
}
- (BOOL)setInputGain:(float)gain error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setInputGain:gain error:outError];
}
- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setPreferredSampleRate:sampleRate error:outError];
}
- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setPreferredIOBufferDuration:duration error:outError];
}
- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setPreferredInputNumberOfChannels:count error:outError];
}
- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setPreferredOutputNumberOfChannels:count error:outError];
}
- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session overrideOutputAudioPort:portOverride error:outError];
}
- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setPreferredInput:inPort error:outError];
}
- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setInputDataSource:dataSource error:outError];
}
- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError {
if (![self checkLock:outError]) {
return NO;
}
return [self.session setOutputDataSource:dataSource error:outError];
}
#pragma mark - Notifications
- (void)handleInterruptionNotification:(NSNotification *)notification {
NSNumber* typeNumber =
notification.userInfo[AVAudioSessionInterruptionTypeKey];
AVAudioSessionInterruptionType type =
(AVAudioSessionInterruptionType)typeNumber.unsignedIntegerValue;
switch (type) {
case AVAudioSessionInterruptionTypeBegan:
RTCLog(@"Audio session interruption began.");
self.isActive = NO;
self.isInterrupted = YES;
[self notifyDidBeginInterruption];
break;
case AVAudioSessionInterruptionTypeEnded: {
RTCLog(@"Audio session interruption ended.");
self.isInterrupted = NO;
[self updateAudioSessionAfterEvent];
NSNumber *optionsNumber =
notification.userInfo[AVAudioSessionInterruptionOptionKey];
AVAudioSessionInterruptionOptions options =
optionsNumber.unsignedIntegerValue;
BOOL shouldResume =
options & AVAudioSessionInterruptionOptionShouldResume;
[self notifyDidEndInterruptionWithShouldResumeSession:shouldResume];
break;
}
}
}
- (void)handleRouteChangeNotification:(NSNotification *)notification {
// Get reason for current route change.
NSNumber* reasonNumber =
notification.userInfo[AVAudioSessionRouteChangeReasonKey];
AVAudioSessionRouteChangeReason reason =
(AVAudioSessionRouteChangeReason)reasonNumber.unsignedIntegerValue;
RTCLog(@"Audio route changed:");
switch (reason) {
case AVAudioSessionRouteChangeReasonUnknown:
RTCLog(@"Audio route changed: ReasonUnknown");
break;
case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
RTCLog(@"Audio route changed: NewDeviceAvailable");
break;
case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
RTCLog(@"Audio route changed: OldDeviceUnavailable");
break;
case AVAudioSessionRouteChangeReasonCategoryChange:
RTCLog(@"Audio route changed: CategoryChange to :%@",
self.session.category);
break;
case AVAudioSessionRouteChangeReasonOverride:
RTCLog(@"Audio route changed: Override");
break;
case AVAudioSessionRouteChangeReasonWakeFromSleep:
RTCLog(@"Audio route changed: WakeFromSleep");
break;
case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
RTCLog(@"Audio route changed: NoSuitableRouteForCategory");
break;
case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
RTCLog(@"Audio route changed: RouteConfigurationChange");
break;
}
AVAudioSessionRouteDescription* previousRoute =
notification.userInfo[AVAudioSessionRouteChangePreviousRouteKey];
// Log previous route configuration.
RTCLog(@"Previous route: %@\nCurrent route:%@",
previousRoute, self.session.currentRoute);
[self notifyDidChangeRouteWithReason:reason previousRoute:previousRoute];
}
- (void)handleMediaServicesWereLost:(NSNotification *)notification {
RTCLog(@"Media services were lost.");
[self updateAudioSessionAfterEvent];
[self notifyMediaServicesWereLost];
}
- (void)handleMediaServicesWereReset:(NSNotification *)notification {
RTCLog(@"Media services were reset.");
[self updateAudioSessionAfterEvent];
[self notifyMediaServicesWereReset];
}
- (void)handleSilenceSecondaryAudioHintNotification:(NSNotification *)notification {
// TODO(henrika): just adding logs here for now until we know if we are ever
// see this notification and might be affected by it or if further actions
// are required.
NSNumber *typeNumber =
notification.userInfo[AVAudioSessionSilenceSecondaryAudioHintTypeKey];
AVAudioSessionSilenceSecondaryAudioHintType type =
(AVAudioSessionSilenceSecondaryAudioHintType)typeNumber.unsignedIntegerValue;
switch (type) {
case AVAudioSessionSilenceSecondaryAudioHintTypeBegin:
RTCLog(@"Another application's primary audio has started.");
break;
case AVAudioSessionSilenceSecondaryAudioHintTypeEnd:
RTCLog(@"Another application's primary audio has stopped.");
break;
}
}
- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
RTCLog(@"Application became active after an interruption. Treating as interruption "
" end. isInterrupted changed from %d to 0.", self.isInterrupted);
if (self.isInterrupted) {
self.isInterrupted = NO;
[self updateAudioSessionAfterEvent];
}
// Always treat application becoming active as an interruption end event.
[self notifyDidEndInterruptionWithShouldResumeSession:YES];
}
#pragma mark - Private
+ (NSError *)lockError {
NSDictionary *userInfo = @{
NSLocalizedDescriptionKey:
@"Must call lockForConfiguration before calling this method."
};
NSError *error =
[[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
code:kRTCAudioSessionErrorLockRequired
userInfo:userInfo];
return error;
}
- (std::vector<__weak id<RTCAudioSessionDelegate> >)delegates {
@synchronized(self) {
// Note: this returns a copy.
return _delegates;
}
}
// TODO(tkchin): check for duplicates.
- (void)pushDelegate:(id<RTCAudioSessionDelegate>)delegate {
@synchronized(self) {
_delegates.insert(_delegates.begin(), delegate);
}
}
- (void)removeZeroedDelegates {
@synchronized(self) {
_delegates.erase(
std::remove_if(_delegates.begin(),
_delegates.end(),
[](id delegate) -> bool { return delegate == nil; }),
_delegates.end());
}
}
- (int)activationCount {
return _activationCount;
}
- (int)incrementActivationCount {
RTCLog(@"Incrementing activation count.");
return rtc::AtomicOps::Increment(&_activationCount);
}
- (NSInteger)decrementActivationCount {
RTCLog(@"Decrementing activation count.");
return rtc::AtomicOps::Decrement(&_activationCount);
}
- (int)webRTCSessionCount {
return _webRTCSessionCount;
}
- (BOOL)canPlayOrRecord {
return !self.useManualAudio || self.isAudioEnabled;
}
- (BOOL)isInterrupted {
@synchronized(self) {
return _isInterrupted;
}
}
- (void)setIsInterrupted:(BOOL)isInterrupted {
@synchronized(self) {
if (_isInterrupted == isInterrupted) {
return;
}
_isInterrupted = isInterrupted;
}
}
- (BOOL)checkLock:(NSError **)outError {
// Check ivar instead of trying to acquire lock so that we won't accidentally
// acquire lock if it hasn't already been called.
if (!self.isLocked) {
if (outError) {
*outError = [RTCAudioSession lockError];
}
return NO;
}
return YES;
}
- (BOOL)beginWebRTCSession:(NSError **)outError {
if (outError) {
*outError = nil;
}
if (![self checkLock:outError]) {
return NO;
}
rtc::AtomicOps::Increment(&_webRTCSessionCount);
[self notifyDidStartPlayOrRecord];
return YES;
}
- (BOOL)endWebRTCSession:(NSError **)outError {
if (outError) {
*outError = nil;
}
if (![self checkLock:outError]) {
return NO;
}
rtc::AtomicOps::Decrement(&_webRTCSessionCount);
[self notifyDidStopPlayOrRecord];
return YES;
}
- (BOOL)configureWebRTCSession:(NSError **)outError {
if (outError) {
*outError = nil;
}
if (![self checkLock:outError]) {
return NO;
}
RTCLog(@"Configuring audio session for WebRTC.");
// Configure the AVAudioSession and activate it.
// Provide an error even if there isn't one so we can log it.
NSError *error = nil;
RTCAudioSessionConfiguration *webRTCConfig =
[RTCAudioSessionConfiguration webRTCConfiguration];
if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
RTCLogError(@"Failed to set WebRTC audio configuration: %@",
error.localizedDescription);
// Do not call setActive:NO if setActive:YES failed.
if (outError) {
*outError = error;
}
return NO;
}
// Ensure that the device currently supports audio input.
// TODO(tkchin): Figure out if this is really necessary.
if (!self.inputAvailable) {
RTCLogError(@"No audio input path is available!");
[self unconfigureWebRTCSession:nil];
if (outError) {
*outError = [self configurationErrorWithDescription:@"No input path."];
}
return NO;
}
// It can happen (e.g. in combination with BT devices) that the attempt to set
// the preferred sample rate for WebRTC (48kHz) fails. If so, make a new
// configuration attempt using the sample rate that worked using the active
// audio session. A typical case is that only 8 or 16kHz can be set, e.g. in
// combination with BT headsets. Using this "trick" seems to avoid a state
// where Core Audio asks for a different number of audio frames than what the
// session's I/O buffer duration corresponds to.
// TODO(henrika): this fix resolves bugs.webrtc.org/6004 but it has only been
// tested on a limited set of iOS devices and BT devices.
double sessionSampleRate = self.sampleRate;
double preferredSampleRate = webRTCConfig.sampleRate;
if (sessionSampleRate != preferredSampleRate) {
RTCLogWarning(
@"Current sample rate (%.2f) is not the preferred rate (%.2f)",
sessionSampleRate, preferredSampleRate);
if (![self setPreferredSampleRate:sessionSampleRate
error:&error]) {
RTCLogError(@"Failed to set preferred sample rate: %@",
error.localizedDescription);
if (outError) {
*outError = error;
}
}
}
return YES;
}
- (BOOL)unconfigureWebRTCSession:(NSError **)outError {
if (outError) {
*outError = nil;
}
if (![self checkLock:outError]) {
return NO;
}
RTCLog(@"Unconfiguring audio session for WebRTC.");
[self setActive:NO error:outError];
return YES;
}
- (NSError *)configurationErrorWithDescription:(NSString *)description {
NSDictionary* userInfo = @{
NSLocalizedDescriptionKey: description,
};
return [[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
code:kRTCAudioSessionErrorConfiguration
userInfo:userInfo];
}
- (void)updateAudioSessionAfterEvent {
BOOL shouldActivate = self.activationCount > 0;
AVAudioSessionSetActiveOptions options = shouldActivate ?
0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
NSError *error = nil;
if ([self.session setActive:shouldActivate
withOptions:options
error:&error]) {
self.isActive = shouldActivate;
} else {
RTCLogError(@"Failed to set session active to %d. Error:%@",
shouldActivate, error.localizedDescription);
}
}
- (void)updateCanPlayOrRecord {
BOOL canPlayOrRecord = NO;
BOOL shouldNotify = NO;
@synchronized(self) {
canPlayOrRecord = !self.useManualAudio || self.isAudioEnabled;
if (_canPlayOrRecord == canPlayOrRecord) {
return;
}
_canPlayOrRecord = canPlayOrRecord;
shouldNotify = YES;
}
if (shouldNotify) {
[self notifyDidChangeCanPlayOrRecord:canPlayOrRecord];
}
}
- (void)audioSessionDidActivate:(AVAudioSession *)session {
if (_session != session) {
RTCLogError(@"audioSessionDidActivate called on different AVAudioSession");
}
[self incrementActivationCount];
self.isActive = YES;
}
- (void)audioSessionDidDeactivate:(AVAudioSession *)session {
if (_session != session) {
RTCLogError(@"audioSessionDidDeactivate called on different AVAudioSession");
}
self.isActive = NO;
[self decrementActivationCount];
}
- (void)observeValueForKeyPath:(NSString *)keyPath
ofObject:(id)object
change:(NSDictionary *)change
context:(void *)context {
if (object == _session) {
NSNumber *newVolume = change[NSKeyValueChangeNewKey];
RTCLog(@"OutputVolumeDidChange to %f", newVolume.floatValue);
[self notifyDidChangeOutputVolume:newVolume.floatValue];
} else {
[super observeValueForKeyPath:keyPath
ofObject:object
change:change
context:context];
}
}
- (void)notifyDidBeginInterruption {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionDidBeginInterruption:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionDidBeginInterruption:self];
}
}
}
- (void)notifyDidEndInterruptionWithShouldResumeSession:
(BOOL)shouldResumeSession {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionDidEndInterruption:shouldResumeSession:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionDidEndInterruption:self
shouldResumeSession:shouldResumeSession];
}
}
}
- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionDidChangeRoute:reason:previousRoute:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionDidChangeRoute:self
reason:reason
previousRoute:previousRoute];
}
}
}
- (void)notifyMediaServicesWereLost {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionMediaServerTerminated:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionMediaServerTerminated:self];
}
}
}
- (void)notifyMediaServicesWereReset {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionMediaServerReset:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionMediaServerReset:self];
}
}
}
- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSession:didChangeCanPlayOrRecord:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSession:self didChangeCanPlayOrRecord:canPlayOrRecord];
}
}
}
- (void)notifyDidStartPlayOrRecord {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionDidStartPlayOrRecord:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionDidStartPlayOrRecord:self];
}
}
}
- (void)notifyDidStopPlayOrRecord {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSessionDidStopPlayOrRecord:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSessionDidStopPlayOrRecord:self];
}
}
}
- (void)notifyDidChangeOutputVolume:(float)volume {
for (auto delegate : self.delegates) {
SEL sel = @selector(audioSession:didChangeOutputVolume:);
if ([delegate respondsToSelector:sel]) {
[delegate audioSession:self didChangeOutputVolume:volume];
}
}
}
@end

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@ -0,0 +1,134 @@
/*
* 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 "WebRTC/RTCAudioSession.h"
#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCDispatcher.h"
#import "WebRTC/UIDevice+RTCDevice.h"
// Try to use mono to save resources. Also avoids channel format conversion
// in the I/O audio unit. Initial tests have shown that it is possible to use
// mono natively for built-in microphones and for BT headsets but not for
// wired headsets. Wired headsets only support stereo as native channel format
// but it is a low cost operation to do a format conversion to mono in the
// audio unit. Hence, we will not hit a RTC_CHECK in
// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
// preferred number of channels and the actual number of channels.
const int kRTCAudioSessionPreferredNumberOfChannels = 1;
// Preferred hardware sample rate (unit is in Hertz). The client sample rate
// will be set to this value as well to avoid resampling the the audio unit's
// format converter. Note that, some devices, e.g. BT headsets, only supports
// 8000Hz as native sample rate.
const double kRTCAudioSessionHighPerformanceSampleRate = 48000.0;
// A lower sample rate will be used for devices with only one core
// (e.g. iPhone 4). The goal is to reduce the CPU load of the application.
const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
// size used by WebRTC. The exact actual size will differ between devices.
// Example: using 48kHz on iPhone 6 results in a native buffer size of
// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
// take care of any buffering required to convert between native buffers and
// buffers used by WebRTC. It is beneficial for the performance if the native
// size is as close to 10ms as possible since it results in "clean" callback
// sequence without bursts of callbacks back to back.
const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
// Use a larger buffer size on devices with only one core (e.g. iPhone 4).
// It will result in a lower CPU consumption at the cost of a larger latency.
// The size of 60ms is based on instrumentation that shows a significant
// reduction in CPU load compared with 10ms on low-end devices.
// TODO(henrika): monitor this size and determine if it should be modified.
const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
static RTCAudioSessionConfiguration *gWebRTCConfiguration = nil;
@implementation RTCAudioSessionConfiguration
@synthesize category = _category;
@synthesize categoryOptions = _categoryOptions;
@synthesize mode = _mode;
@synthesize sampleRate = _sampleRate;
@synthesize ioBufferDuration = _ioBufferDuration;
@synthesize inputNumberOfChannels = _inputNumberOfChannels;
@synthesize outputNumberOfChannels = _outputNumberOfChannels;
- (instancetype)init {
if (self = [super init]) {
// Use a category which supports simultaneous recording and playback.
// By default, using this category implies that our apps audio is
// nonmixable, hence activating the session will interrupt any other
// audio sessions which are also nonmixable.
_category = AVAudioSessionCategoryPlayAndRecord;
_categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
// Specify mode for two-way voice communication (e.g. VoIP).
_mode = AVAudioSessionModeVoiceChat;
// Set the session's sample rate or the hardware sample rate.
// It is essential that we use the same sample rate as stream format
// to ensure that the I/O unit does not have to do sample rate conversion.
// Set the preferred audio I/O buffer duration, in seconds.
NSUInteger processorCount = [NSProcessInfo processInfo].processorCount;
// Use best sample rate and buffer duration if the CPU has more than one
// core.
if (processorCount > 1 && [UIDevice deviceType] != RTCDeviceTypeIPhone4S) {
_sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
_ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
} else {
_sampleRate = kRTCAudioSessionLowComplexitySampleRate;
_ioBufferDuration = kRTCAudioSessionLowComplexityIOBufferDuration;
}
// We try to use mono in both directions to save resources and format
// conversions in the audio unit. Some devices does only support stereo;
// e.g. wired headset on iPhone 6.
// TODO(henrika): add support for stereo if needed.
_inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
_outputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
}
return self;
}
+ (void)initialize {
gWebRTCConfiguration = [[self alloc] init];
}
+ (instancetype)currentConfiguration {
RTCAudioSession *session = [RTCAudioSession sharedInstance];
RTCAudioSessionConfiguration *config =
[[RTCAudioSessionConfiguration alloc] init];
config.category = session.category;
config.categoryOptions = session.categoryOptions;
config.mode = session.mode;
config.sampleRate = session.sampleRate;
config.ioBufferDuration = session.IOBufferDuration;
config.inputNumberOfChannels = session.inputNumberOfChannels;
config.outputNumberOfChannels = session.outputNumberOfChannels;
return config;
}
+ (instancetype)webRTCConfiguration {
@synchronized(self) {
return (RTCAudioSessionConfiguration *)gWebRTCConfiguration;
}
}
+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
@synchronized(self) {
gWebRTCConfiguration = configuration;
}
}
@end