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Merge remote tracking branch 'upstream-master'

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Bug: 153469641
Test: run cuttlefish locally
Change-Id: Ida3bfe62ef5c6549278f4c155a1f690b008e9b9d
This commit is contained in:
Jorge E. Moreira
2020-07-23 13:07:40 -07:00
parent 889b21a2e1 708b22cd03
commit 206ccd0b36
1236 changed files with 50564 additions and 32463 deletions
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12
sdk/objc/api/peerconnection/RTCConfiguration.h
View File

@ -197,18 +197,6 @@ RTC_OBJC_EXPORT
@property(nonatomic, assign) BOOL allowCodecSwitching;
/**
* If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
* that it should use the MediaTransportInterface.
*/
@property(nonatomic, assign) BOOL useMediaTransport;
/**
* If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
* that it should use the MediaTransportInterface for data channels.
*/
@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any

9
sdk/objc/api/peerconnection/RTCConfiguration.mm
View File

@ -52,8 +52,6 @@
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize useMediaTransport = _useMediaTransport;
@synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
@ -106,8 +104,6 @@
_iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
_iceBackupCandidatePairPingInterval =
config.ice_backup_candidate_pair_ping_interval;
_useMediaTransport = config.use_media_transport;
_useMediaTransportForDataChannels = config.use_media_transport_for_data_channels;
_keyType = RTCEncryptionKeyTypeECDSA;
_iceCandidatePoolSize = config.ice_candidate_pool_size;
_shouldPruneTurnPorts = config.prune_turn_ports;
@ -143,7 +139,7 @@
- (NSString *)description {
static NSString *formatString = @"RTC_OBJC_TYPE(RTCConfiguration): "
@"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
@"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n%d\n}\n";
@"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n}\n";
return [NSString
stringWithFormat:formatString,
@ -169,7 +165,6 @@
_disableIPV6OnWiFi,
_maxIPv6Networks,
_activeResetSrtpParams,
_useMediaTransport,
_enableDscp];
}
@ -208,8 +203,6 @@
_iceConnectionReceivingTimeout;
nativeConfig->ice_backup_candidate_pair_ping_interval =
_iceBackupCandidatePairPingInterval;
nativeConfig->use_media_transport = _useMediaTransport;
nativeConfig->use_media_transport_for_data_channels = _useMediaTransportForDataChannels;
rtc::KeyType keyType =
[[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
if (_certificate != nullptr) {

3
sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm
View File

@ -28,7 +28,8 @@ class StatsCollectorCallbackAdapter : public RTCStatsCollectorCallback {
void OnStatsDelivered(const rtc::scoped_refptr<const RTCStatsReport> &report) override {
RTC_DCHECK(completion_handler_);
RTCStatisticsReport *statisticsReport = [[RTCStatisticsReport alloc] initWithReport:*report];
RTC_OBJC_TYPE(RTCStatisticsReport) *statisticsReport =
[[RTC_OBJC_TYPE(RTCStatisticsReport) alloc] initWithReport:*report];
completion_handler_(statisticsReport);
completion_handler_ = nil;
}

4
sdk/objc/api/peerconnection/RTCPeerConnection.h
View File

@ -25,7 +25,7 @@
@class RTC_OBJC_TYPE(RTCRtpTransceiver);
@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
@class RTC_OBJC_TYPE(RTCSessionDescription);
@class RTCStatisticsReport;
@class RTC_OBJC_TYPE(RTCStatisticsReport);
@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
typedef NS_ENUM(NSInteger, RTCRtpMediaType);
@ -341,7 +341,7 @@ RTC_OBJC_EXPORT
@end
typedef void (^RTCStatisticsCompletionHandler)(RTCStatisticsReport *);
typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
@interface RTC_OBJC_TYPE (RTCPeerConnection)
(Stats)

5
sdk/objc/api/peerconnection/RTCPeerConnection.mm
View File

@ -29,7 +29,6 @@
#include "api/jsep_ice_candidate.h"
#include "api/rtc_event_log_output_file.h"
#include "api/transport/media/media_transport_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@ -559,12 +558,12 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
webrtc::PeerConnectionInterface::BitrateParameters params;
webrtc::BitrateSettings params;
if (minBitrateBps != nil) {
params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
}
if (currentBitrateBps != nil) {
params.current_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
params.start_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
}
if (maxBitrateBps != nil) {
params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);

23
sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
View File

@ -17,7 +17,6 @@ namespace webrtc {
class AudioDeviceModule;
class AudioEncoderFactory;
class AudioDecoderFactory;
class MediaTransportFactory;
class NetworkControllerFactoryInterface;
class VideoEncoderFactory;
class VideoDecoderFactory;
@ -53,21 +52,6 @@ NS_ASSUME_NONNULL_BEGIN
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
- (instancetype)
initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
nativeAudioDecoderFactory:
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
nativeVideoEncoderFactory:
(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
nativeVideoDecoderFactory:
(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
mediaTransportFactory:
(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
- (instancetype)
initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
@ -81,14 +65,11 @@ NS_ASSUME_NONNULL_BEGIN
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
networkControllerFactory:(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
networkControllerFactory
mediaTransportFactory:
(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
networkControllerFactory;
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* dependencies.

48
sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
View File

@ -52,7 +52,6 @@
// C++ target.
// TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
// API layer.
#include "api/transport/media/media_transport_interface.h"
#include "media/engine/webrtc_media_engine.h" // nogncheck
@implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
@ -84,15 +83,13 @@
nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
RTCVideoDecoderFactoryH264) alloc] init])
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr
mediaTransportFactory:nullptr];
audioProcessingModule:nullptr];
#endif
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
@ -109,18 +106,9 @@
nativeVideoEncoderFactory:std::move(native_encoder_factory)
nativeVideoDecoderFactory:std::move(native_decoder_factory)
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr
mediaTransportFactory:std::move(mediaTransportFactory)];
audioProcessingModule:nullptr];
#endif
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
return [self initWithEncoderFactory:encoderFactory
decoderFactory:decoderFactory
mediaTransportFactory:nullptr];
}
- (instancetype)initNative {
if (self = [super init]) {
_networkThread = rtc::Thread::CreateWithSocketServer();
@ -170,30 +158,7 @@
nativeVideoDecoderFactory:std::move(videoDecoderFactory)
audioDeviceModule:audioDeviceModule
audioProcessingModule:audioProcessingModule
mediaTransportFactory:nullptr];
}
- (instancetype)initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
nativeAudioDecoderFactory:
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
nativeVideoEncoderFactory:
(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
nativeVideoDecoderFactory:
(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
mediaTransportFactory {
return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
nativeAudioDecoderFactory:audioDecoderFactory
nativeVideoEncoderFactory:std::move(videoEncoderFactory)
nativeVideoDecoderFactory:std::move(videoDecoderFactory)
audioDeviceModule:audioDeviceModule
audioProcessingModule:audioProcessingModule
networkControllerFactory:nullptr
mediaTransportFactory:std::move(mediaTransportFactory)];
networkControllerFactory:nullptr];
}
- (instancetype)initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
@ -208,9 +173,7 @@
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
networkControllerFactory:
(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
networkControllerFactory
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
mediaTransportFactory {
networkControllerFactory {
if (self = [self initNative]) {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = _networkThread.get();
@ -235,7 +198,6 @@
dependencies.event_log_factory =
std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
dependencies.network_controller_factory = std::move(networkControllerFactory);
dependencies.media_transport_factory = std::move(mediaTransportFactory);
#endif
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");

5
sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
View File

@ -13,7 +13,6 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/transport/media/media_transport_interface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
@ -26,7 +25,6 @@
rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
std::unique_ptr<webrtc::MediaTransportFactory> _mediaTransportFactory;
}
+ (RTCPeerConnectionFactoryBuilder *)builder {
@ -41,8 +39,7 @@
nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
audioDeviceModule:_audioDeviceModule
audioProcessingModule:_audioProcessingModule
mediaTransportFactory:std::move(_mediaTransportFactory)];
audioProcessingModule:_audioProcessingModule];
}
- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {

4
sdk/objc/api/peerconnection/RTCRtpSender.h
View File

@ -21,8 +21,8 @@ RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCRtpSender)<NSObject>
/** A unique identifier for this sender. */
@property(nonatomic, readonly) NSString *senderId;
/** A unique identifier for this sender. */
@property(nonatomic, readonly) NSString *senderId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.

4
sdk/objc/api/peerconnection/RTCStatisticsReport+Private.h
View File

@ -12,8 +12,8 @@
#include "api/stats/rtc_stats_report.h"
@interface RTCStatisticsReport (Private)
@interface RTC_OBJC_TYPE (RTCStatisticsReport) (Private)
- (instancetype)initWithReport:(const webrtc::RTCStatsReport &)report;
- (instancetype)initWithReport : (const webrtc::RTCStatsReport &)report;
@end

12
sdk/objc/api/peerconnection/RTCStatisticsReport.h
View File

@ -10,25 +10,29 @@
#import <Foundation/Foundation.h>
@class RTCStatistics;
#import "RTCMacros.h"
@class RTC_OBJC_TYPE(RTCStatistics);
NS_ASSUME_NONNULL_BEGIN
/** A statistics report. Encapsulates a number of RTCStatistics objects. */
@interface RTCStatisticsReport : NSObject
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCStatisticsReport) : NSObject
/** The timestamp of the report in microseconds since 1970-01-01T00:00:00Z. */
@property(nonatomic, readonly) CFTimeInterval timestamp_us;
/** RTCStatistics objects by id. */
@property(nonatomic, readonly) NSDictionary<NSString *, RTCStatistics *> *statistics;
@property(nonatomic, readonly) NSDictionary<NSString *, RTC_OBJC_TYPE(RTCStatistics) *> *statistics;
- (instancetype)init NS_UNAVAILABLE;
@end
/** A part of a report (a subreport) covering a certain area. */
@interface RTCStatistics : NSObject
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCStatistics) : NSObject
/** The id of this subreport, e.g. "RTCMediaStreamTrack_receiver_2". */
@property(nonatomic, readonly) NSString *id;

11
sdk/objc/api/peerconnection/RTCStatisticsReport.mm
View File

@ -100,7 +100,7 @@ NSObject *ValueFromStatsMember(const RTCStatsMemberInterface *member) {
}
} // namespace webrtc
@implementation RTCStatistics
@implementation RTC_OBJC_TYPE (RTCStatistics)
@synthesize id = _id;
@synthesize timestamp_us = _timestamp_us;
@ -139,7 +139,7 @@ NSObject *ValueFromStatsMember(const RTCStatsMemberInterface *member) {
@end
@implementation RTCStatisticsReport
@implementation RTC_OBJC_TYPE (RTCStatisticsReport)
@synthesize timestamp_us = _timestamp_us;
@synthesize statistics = _statistics;
@ -151,16 +151,17 @@ NSObject *ValueFromStatsMember(const RTCStatsMemberInterface *member) {
@end
@implementation RTCStatisticsReport (Private)
@implementation RTC_OBJC_TYPE (RTCStatisticsReport) (Private)
- (instancetype)initWithReport:(const webrtc::RTCStatsReport &)report {
- (instancetype)initWithReport : (const webrtc::RTCStatsReport &)report {
if (self = [super init]) {
_timestamp_us = report.timestamp_us();
NSMutableDictionary *statisticsById =
[NSMutableDictionary dictionaryWithCapacity:report.size()];
for (const auto &stat : report) {
RTCStatistics *statistics = [[RTCStatistics alloc] initWithStatistics:stat];
RTC_OBJC_TYPE(RTCStatistics) *statistics =
[[RTC_OBJC_TYPE(RTCStatistics) alloc] initWithStatistics:stat];
statisticsById[statistics.id] = statistics;
}
_statistics = [statisticsById copy];

4
sdk/objc/components/audio/RTCAudioSession.mm
View File

@ -16,7 +16,7 @@
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/deprecated/recursive_critical_section.h"
#import "RTCAudioSessionConfiguration.h"
#import "base/RTCLogging.h"
@ -35,7 +35,7 @@ NSString * const kRTCAudioSessionOutputVolumeSelector = @"outputVolume";
// TODO(tkchin): Consider more granular locking. We're not expecting a lot of
// lock contention so coarse locks should be fine for now.
@implementation RTC_OBJC_TYPE (RTCAudioSession) {
rtc::CriticalSection _crit;
rtc::RecursiveCriticalSection _crit;
AVAudioSession *_session;
volatile int _activationCount;
volatile int _lockRecursionCount;

1
sdk/objc/native/src/audio/audio_device_ios.mm
View File

@ -21,7 +21,6 @@
#include "rtc_base/atomic_ops.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"

1
sdk/objc/native/src/audio/audio_device_module_ios.h
View File

@ -19,7 +19,6 @@
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/critical_section.h"
namespace webrtc {

7
sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
View File

@ -22,7 +22,6 @@ extern "C" {
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/transport/media/media_transport_interface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
@ -50,8 +49,7 @@ extern "C" {
nativeVideoEncoderFactory:nullptr
nativeVideoDecoderFactory:nullptr
audioDeviceModule:nullptr
audioProcessingModule:nullptr
mediaTransportFactory:nullptr]);
audioProcessingModule:nullptr]);
#endif
RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =
@ -72,8 +70,7 @@ extern "C" {
nativeVideoEncoderFactory:nullptr
nativeVideoDecoderFactory:nullptr
audioDeviceModule:nullptr
audioProcessingModule:nullptr
mediaTransportFactory:nullptr]);
audioProcessingModule:nullptr]);
#endif
RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =