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Obj-C SDK Cleanup

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This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
This commit is contained in:
Anders Carlsson
2018-08-30 09:30:29 +02:00
committed by Commit Bot
parent 9ea5765f78
commit 7bca8ca4e2
470 changed files with 7255 additions and 5258 deletions
Download Patch File Download Diff File Expand all files Collapse all files

13
sdk/objc/DEPS
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@ -1,12 +1,8 @@
include_rules = [
"+WebRTC",
"+Common",
"+Video",
"+Audio",
"+UI",
"+PeerConnection",
"+VideoToolbox",
"+Metal",
"+base",
"+components",
"+helpers",
"+sdk",
"+api",
"+common_video/h264",
"+common_video/include",
@ -17,5 +13,6 @@ include_rules = [
"+system_wrappers",
"+modules/audio_device",
"+modules/audio_processing",
"+native",
"+third_party/libyuv",
]

17
sdk/objc/Framework/Classes/Common/NSString+StdString.h
View File

@ -8,19 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#include <string>
NS_ASSUME_NONNULL_BEGIN
@interface NSString (StdString)
@property(nonatomic, readonly) std::string stdString;
+ (std::string)stdStringForString:(NSString *)nsString;
+ (NSString *)stringForStdString:(const std::string &)stdString;
@end
NS_ASSUME_NONNULL_END
#import "helpers/NSString+StdString.h"

16
sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#if defined(WEBRTC_IOS)
#import <Foundation/Foundation.h>
NS_EXTENSION_UNAVAILABLE_IOS("Application status not available in app extensions.")
@interface RTCUIApplicationStatusObserver : NSObject
+ (instancetype)sharedInstance;
+ (void)prepareForUse;
- (BOOL)isApplicationActive;
@end
#endif // WEBRTC_IOS
#import "helpers/RTCUIApplicationStatusObserver.h"

106
sdk/objc/Framework/Classes/Common/scoped_cftyperef.h
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@ -9,108 +9,4 @@
*
*/
#ifndef WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_
#define WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_
#include <CoreFoundation/CoreFoundation.h>
namespace rtc {
// RETAIN: ScopedTypeRef should retain the object when it takes
// ownership.
// ASSUME: Assume the object already has already been retained.
// ScopedTypeRef takes over ownership.
enum class RetainPolicy { RETAIN, ASSUME };
namespace internal {
template <typename T>
struct CFTypeRefTraits {
static T InvalidValue() { return nullptr; }
static void Release(T ref) { CFRelease(ref); }
static T Retain(T ref) {
CFRetain(ref);
return ref;
}
};
template <typename T, typename Traits>
class ScopedTypeRef {
public:
ScopedTypeRef() : ptr_(Traits::InvalidValue()) {}
explicit ScopedTypeRef(T ptr) : ptr_(ptr) {}
ScopedTypeRef(T ptr, RetainPolicy policy) : ScopedTypeRef(ptr) {
if (ptr_ && policy == RetainPolicy::RETAIN)
Traits::Retain(ptr_);
}
ScopedTypeRef(const ScopedTypeRef<T, Traits>& rhs) : ptr_(rhs.ptr_) {
if (ptr_)
ptr_ = Traits::Retain(ptr_);
}
~ScopedTypeRef() {
if (ptr_) {
Traits::Release(ptr_);
}
}
T get() const { return ptr_; }
T operator->() const { return ptr_; }
explicit operator bool() const { return ptr_; }
bool operator!() const { return !ptr_; }
ScopedTypeRef& operator=(const T& rhs) {
if (ptr_)
Traits::Release(ptr_);
ptr_ = rhs;
return *this;
}
ScopedTypeRef& operator=(const ScopedTypeRef<T, Traits>& rhs) {
reset(rhs.get(), RetainPolicy::RETAIN);
return *this;
}
// This is intended to take ownership of objects that are
// created by pass-by-pointer initializers.
T* InitializeInto() {
RTC_DCHECK(!ptr_);
return &ptr_;
}
void reset(T ptr, RetainPolicy policy = RetainPolicy::ASSUME) {
if (ptr && policy == RetainPolicy::RETAIN)
Traits::Retain(ptr);
if (ptr_)
Traits::Release(ptr_);
ptr_ = ptr;
}
T release() {
T temp = ptr_;
ptr_ = Traits::InvalidValue();
return temp;
}
private:
T ptr_;
};
} // namespace internal
template <typename T>
using ScopedCFTypeRef =
internal::ScopedTypeRef<T, internal::CFTypeRefTraits<T>>;
template <typename T>
static ScopedCFTypeRef<T> AdoptCF(T cftype) {
return ScopedCFTypeRef<T>(cftype, RetainPolicy::RETAIN);
}
template <typename T>
static ScopedCFTypeRef<T> ScopedCF(T cftype) {
return ScopedCFTypeRef<T>(cftype);
}
} // namespace rtc
#endif // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_
#import "helpers/scoped_cftyperef.h"

19
sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCConfiguration.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCConfiguration ()
/** Optional TurnCustomizer.
* With this class one can modify outgoing TURN messages.
* The object passed in must remain valid until PeerConnection::Close() is
* called.
*/
@property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCConfiguration+Native.h"

83
sdk/objc/Framework/Classes/PeerConnection/RTCEncodedImage.mm
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@ -1,83 +0,0 @@
/*
* Copyright 2017 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/RTCVideoCodec.h"
#import "RTCVideoCodec+Private.h"
#include "rtc_base/numerics/safe_conversions.h"
@implementation RTCEncodedImage
@synthesize buffer = _buffer;
@synthesize encodedWidth = _encodedWidth;
@synthesize encodedHeight = _encodedHeight;
@synthesize timeStamp = _timeStamp;
@synthesize captureTimeMs = _captureTimeMs;
@synthesize ntpTimeMs = _ntpTimeMs;
@synthesize flags = _flags;
@synthesize encodeStartMs = _encodeStartMs;
@synthesize encodeFinishMs = _encodeFinishMs;
@synthesize frameType = _frameType;
@synthesize rotation = _rotation;
@synthesize completeFrame = _completeFrame;
@synthesize qp = _qp;
@synthesize contentType = _contentType;
- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage {
if (self = [super init]) {
// Wrap the buffer in NSData without copying, do not take ownership.
_buffer = [NSData dataWithBytesNoCopy:encodedImage._buffer
length:encodedImage._length
freeWhenDone:NO];
_encodedWidth = rtc::dchecked_cast<int32_t>(encodedImage._encodedWidth);
_encodedHeight = rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
_timeStamp = encodedImage.Timestamp();
_captureTimeMs = encodedImage.capture_time_ms_;
_ntpTimeMs = encodedImage.ntp_time_ms_;
_flags = encodedImage.timing_.flags;
_encodeStartMs = encodedImage.timing_.encode_start_ms;
_encodeFinishMs = encodedImage.timing_.encode_finish_ms;
_frameType = static_cast<RTCFrameType>(encodedImage._frameType);
_rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
_completeFrame = encodedImage._completeFrame;
_qp = @(encodedImage.qp_);
_contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
RTCVideoContentTypeScreenshare :
RTCVideoContentTypeUnspecified;
}
return self;
}
- (webrtc::EncodedImage)nativeEncodedImage {
// Return the pointer without copying.
webrtc::EncodedImage encodedImage(
(uint8_t *)_buffer.bytes, (size_t)_buffer.length, (size_t)_buffer.length);
encodedImage._encodedWidth = rtc::dchecked_cast<uint32_t>(_encodedWidth);
encodedImage._encodedHeight = rtc::dchecked_cast<uint32_t>(_encodedHeight);
encodedImage.SetTimestamp(_timeStamp);
encodedImage.capture_time_ms_ = _captureTimeMs;
encodedImage.ntp_time_ms_ = _ntpTimeMs;
encodedImage.timing_.flags = _flags;
encodedImage.timing_.encode_start_ms = _encodeStartMs;
encodedImage.timing_.encode_finish_ms = _encodeFinishMs;
encodedImage._frameType = webrtc::FrameType(_frameType);
encodedImage.rotation_ = webrtc::VideoRotation(_rotation);
encodedImage._completeFrame = _completeFrame;
encodedImage.qp_ = _qp ? _qp.intValue : -1;
encodedImage.content_type_ = (_contentType == RTCVideoContentTypeScreenshare) ?
webrtc::VideoContentType::SCREENSHARE :
webrtc::VideoContentType::UNSPECIFIED;
return encodedImage;
}
@end

25
sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Native.h
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@ -8,27 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCPeerConnection.h"
#include <memory>
namespace rtc {
class BitrateAllocationStrategy;
} // namespace rtc
NS_ASSUME_NONNULL_BEGIN
/**
* This class extension exposes methods that work directly with injectable C++ components.
*/
@interface RTCPeerConnection ()
/** Sets current strategy. If not set default WebRTC allocator will be used. May be changed during
* an active session.
*/
- (void)setBitrateAllocationStrategy:
(std::unique_ptr<rtc::BitrateAllocationStrategy>)bitrateAllocationStrategy;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnection+Native.h"

45
sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
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@ -8,47 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCPeerConnectionFactory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModule;
class AudioEncoderFactory;
class AudioDecoderFactory;
class VideoEncoderFactory;
class VideoDecoderFactory;
class AudioProcessing;
} // namespace webrtc
NS_ASSUME_NONNULL_BEGIN
/**
* This class extension exposes methods that work directly with injectable C++ components.
*/
@interface RTCPeerConnectionFactory ()
- (instancetype)initNative NS_DESIGNATED_INITIALIZER;
/* Initializer used when WebRTC is compiled with no media support */
- (instancetype)initWithNoMedia;
/* Initialize object with injectable native audio/video encoder/decoder factories */
- (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;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"

53
sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
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@ -8,51 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCVideoCodec.h"
#import "WebRTC/RTCVideoCodecH264.h"
#include "api/video_codecs/sdp_video_format.h"
#include "common_video/include/video_frame.h"
#include "media/base/codec.h"
#include "modules/video_coding/include/video_codec_interface.h"
NS_ASSUME_NONNULL_BEGIN
/* Interfaces for converting to/from internal C++ formats. */
@interface RTCEncodedImage ()
- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage;
- (webrtc::EncodedImage)nativeEncodedImage;
@end
@interface RTCVideoEncoderSettings ()
- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *__nullable)videoCodec;
- (webrtc::VideoCodec)nativeVideoCodec;
@end
@interface RTCCodecSpecificInfoH264 ()
- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo;
@end
@interface RTCRtpFragmentationHeader ()
- (instancetype)initWithNativeFragmentationHeader:
(const webrtc::RTPFragmentationHeader *__nullable)fragmentationHeader;
- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader;
@end
@interface RTCVideoCodecInfo ()
- (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format;
- (webrtc::SdpVideoFormat)nativeSdpVideoFormat;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCEncodedImage+Private.h"
#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
#import "api/peerconnection/RTCVideoEncoderSettings+Private.h"
#import "components/video_codec/RTCCodecSpecificInfoH264+Private.h"

167
sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
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@ -1,167 +0,0 @@
/*
* Copyright 2017 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/RTCVideoCodec.h"
#import "NSString+StdString.h"
#import "RTCVideoCodec+Private.h"
#if defined(WEBRTC_IOS)
#import "UIDevice+H264Profile.h"
#endif
#import "WebRTC/RTCVideoCodecFactory.h"
#include "media/base/mediaconstants.h"
namespace {
NSString *MaxSupportedProfileLevelConstrainedHigh();
NSString *MaxSupportedProfileLevelConstrainedBaseline();
} // namespace
NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
MaxSupportedProfileLevelConstrainedHigh();
NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
MaxSupportedProfileLevelConstrainedBaseline();
namespace {
#if defined(WEBRTC_IOS)
using namespace webrtc::H264;
NSString *MaxSupportedLevelForProfile(Profile profile) {
const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
if (profileLevelId && profileLevelId->profile >= profile) {
const absl::optional<std::string> profileString =
ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
if (profileString) {
return [NSString stringForStdString:*profileString];
}
}
return nil;
}
#endif
NSString *MaxSupportedProfileLevelConstrainedBaseline() {
#if defined(WEBRTC_IOS)
NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedBaseline);
if (profile != nil) {
return profile;
}
#endif
return kRTCLevel31ConstrainedBaseline;
}
NSString *MaxSupportedProfileLevelConstrainedHigh() {
#if defined(WEBRTC_IOS)
NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedHigh);
if (profile != nil) {
return profile;
}
#endif
return kRTCLevel31ConstrainedHigh;
}
} // namespace
@implementation RTCVideoCodecInfo
@synthesize name = _name;
@synthesize parameters = _parameters;
- (instancetype)initWithName:(NSString *)name {
return [self initWithName:name parameters:nil];
}
- (instancetype)initWithName:(NSString *)name
parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
if (self = [super init]) {
_name = name;
_parameters = (parameters ? parameters : @{});
}
return self;
}
- (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format {
NSMutableDictionary *params = [NSMutableDictionary dictionary];
for (auto it = format.parameters.begin(); it != format.parameters.end(); ++it) {
[params setObject:[NSString stringForStdString:it->second]
forKey:[NSString stringForStdString:it->first]];
}
return [self initWithName:[NSString stringForStdString:format.name] parameters:params];
}
- (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info {
if (!info ||
![self.name isEqualToString:info.name] ||
![self.parameters isEqualToDictionary:info.parameters]) {
return NO;
}
return YES;
}
- (BOOL)isEqual:(id)object {
if (self == object)
return YES;
if (![object isKindOfClass:[self class]])
return NO;
return [self isEqualToCodecInfo:object];
}
- (NSUInteger)hash {
return [self.name hash] ^ [self.parameters hash];
}
- (webrtc::SdpVideoFormat)nativeSdpVideoFormat {
std::map<std::string, std::string> parameters;
for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey];
std::string value = [NSString stdStringForString:_parameters[paramKey]];
parameters[key] = value;
}
return webrtc::SdpVideoFormat([NSString stdStringForString:_name], parameters);
}
#pragma mark - NSCoding
- (instancetype)initWithCoder:(NSCoder *)decoder {
return [self initWithName:[decoder decodeObjectForKey:@"name"]
parameters:[decoder decodeObjectForKey:@"parameters"]];
}
- (void)encodeWithCoder:(NSCoder *)encoder {
[encoder encodeObject:_name forKey:@"name"];
[encoder encodeObject:_parameters forKey:@"parameters"];
}
@end
@implementation RTCVideoEncoderQpThresholds
@synthesize low = _low;
@synthesize high = _high;
- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high {
if (self = [super init]) {
_low = low;
_high = high;
}
return self;
}
@end

66
sdk/objc/Framework/Classes/PeerConnection/RTCVideoEncoderSettings.mm
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@ -1,66 +0,0 @@
/*
* Copyright 2017 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/RTCVideoCodec.h"
#import "NSString+StdString.h"
#import "RTCVideoCodec+Private.h"
#import "WebRTC/RTCVideoCodecFactory.h"
@implementation RTCVideoEncoderSettings
@synthesize name = _name;
@synthesize width = _width;
@synthesize height = _height;
@synthesize startBitrate = _startBitrate;
@synthesize maxBitrate = _maxBitrate;
@synthesize minBitrate = _minBitrate;
@synthesize targetBitrate = _targetBitrate;
@synthesize maxFramerate = _maxFramerate;
@synthesize qpMax = _qpMax;
@synthesize mode = _mode;
- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec {
if (self = [super init]) {
if (videoCodec) {
const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
_name = [NSString stringWithUTF8String:codecName];
_width = videoCodec->width;
_height = videoCodec->height;
_startBitrate = videoCodec->startBitrate;
_maxBitrate = videoCodec->maxBitrate;
_minBitrate = videoCodec->minBitrate;
_targetBitrate = videoCodec->targetBitrate;
_maxFramerate = videoCodec->maxFramerate;
_qpMax = videoCodec->qpMax;
_mode = (RTCVideoCodecMode)videoCodec->mode;
}
}
return self;
}
- (webrtc::VideoCodec)nativeVideoCodec {
webrtc::VideoCodec videoCodec;
videoCodec.width = _width;
videoCodec.height = _height;
videoCodec.startBitrate = _startBitrate;
videoCodec.maxBitrate = _maxBitrate;
videoCodec.minBitrate = _minBitrate;
videoCodec.targetBitrate = _targetBitrate;
videoCodec.maxBitrate = _maxBitrate;
videoCodec.qpMax = _qpMax;
videoCodec.mode = (webrtc::VideoCodecMode)_mode;
return videoCodec;
}
@end

14
sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
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@ -8,16 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCVideoViewShading.h"
NS_ASSUME_NONNULL_BEGIN
/** Default RTCVideoViewShading that will be used in RTCNSGLVideoView and
* RTCEAGLVideoView if no external shader is specified. This shader will render
* the video in a rectangle without any color or geometric transformations.
*/
@interface RTCDefaultShader : NSObject<RTCVideoViewShading>
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCDefaultShader.h"

22
sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
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@ -8,24 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <GLKit/GLKit.h>
@class RTCVideoFrame;
NS_ASSUME_NONNULL_BEGIN
@interface RTCNV12TextureCache : NSObject
@property(nonatomic, readonly) GLuint yTexture;
@property(nonatomic, readonly) GLuint uvTexture;
- (instancetype)init NS_UNAVAILABLE;
- (nullable instancetype)initWithContext:(EAGLContext *)context NS_DESIGNATED_INITIALIZER;
- (BOOL)uploadFrameToTextures:(RTCVideoFrame *)frame;
- (void)releaseTextures;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCNV12TextureCache.h"

107
sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
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@ -6,111 +6,6 @@
* 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_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
#include "modules/video_coding/codecs/h264/include/h264.h"
#include <CoreMedia/CoreMedia.h>
#include <vector>
#include "common_video/h264/h264_common.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/buffer.h"
using webrtc::H264::NaluIndex;
namespace webrtc {
// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
// needs to be in Annex B format. Data is written directly to |annexb_buffer|
// and a new RTPFragmentationHeader is returned in |out_header|.
bool H264CMSampleBufferToAnnexBBuffer(
CMSampleBufferRef avcc_sample_buffer,
bool is_keyframe,
rtc::Buffer* annexb_buffer,
std::unique_ptr<RTPFragmentationHeader>* out_header);
// Converts a buffer received from RTP into a sample buffer suitable for the
// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
// buffer is in avcc format.
// If |is_keyframe| is true then |video_format| is ignored since the format will
// be read from the buffer. Otherwise |video_format| must be provided.
// Caller is responsible for releasing the created sample buffer.
bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
size_t annexb_buffer_size,
CMVideoFormatDescriptionRef video_format,
CMSampleBufferRef* out_sample_buffer);
// Returns a video format description created from the sps/pps information in
// the Annex B buffer. If there is no such information, nullptr is returned.
// The caller is responsible for releasing the description.
CMVideoFormatDescriptionRef CreateVideoFormatDescription(
const uint8_t* annexb_buffer,
size_t annexb_buffer_size);
// Helper class for reading NALUs from an RTP Annex B buffer.
class AnnexBBufferReader final {
public:
AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length);
~AnnexBBufferReader();
AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
void operator=(const AnnexBBufferReader& other) = delete;
// Returns a pointer to the beginning of the next NALU slice without the
// header bytes and its length. Returns false if no more slices remain.
bool ReadNalu(const uint8_t** out_nalu, size_t* out_length);
// Returns the number of unread NALU bytes, including the size of the header.
// If the buffer has no remaining NALUs this will return zero.
size_t BytesRemaining() const;
// Reset the reader to start reading from the first NALU
void SeekToStart();
// Seek to the next position that holds a NALU of the desired type,
// or the end if no such NALU is found.
// Return true if a NALU of the desired type is found, false if we
// reached the end instead
bool SeekToNextNaluOfType(H264::NaluType type);
private:
// Returns the the next offset that contains NALU data.
size_t FindNextNaluHeader(const uint8_t* start,
size_t length,
size_t offset) const;
const uint8_t* const start_;
std::vector<NaluIndex> offsets_;
std::vector<NaluIndex>::iterator offset_;
const size_t length_;
};
// Helper class for writing NALUs using avcc format into a buffer.
class AvccBufferWriter final {
public:
AvccBufferWriter(uint8_t* const avcc_buffer, size_t length);
~AvccBufferWriter() {}
AvccBufferWriter(const AvccBufferWriter& other) = delete;
void operator=(const AvccBufferWriter& other) = delete;
// Writes the data slice into the buffer. Returns false if there isn't
// enough space left.
bool WriteNalu(const uint8_t* data, size_t data_size);
// Returns the unused bytes in the buffer.
size_t BytesRemaining() const;
private:
uint8_t* const start_;
size_t offset_;
const size_t length_;
};
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
#import "components/video_codec/nalu_rewriter.h"

240
sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
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@ -8,242 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCAudioSessionErrorDomain;
/** Method that requires lock was called without lock. */
extern NSInteger const kRTCAudioSessionErrorLockRequired;
/** Unknown configuration error occurred. */
extern NSInteger const kRTCAudioSessionErrorConfiguration;
@class RTCAudioSession;
@class RTCAudioSessionConfiguration;
// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
// from AVAudioSession and handle them before calling these delegate methods,
// at which point applications can perform additional processing if required.
RTC_EXPORT
@protocol RTCAudioSessionDelegate <NSObject>
@optional
/** Called on a system notification thread when AVAudioSession starts an
* interruption event.
*/
- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
/** Called on a system notification thread when AVAudioSession ends an
* interruption event.
*/
- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
shouldResumeSession:(BOOL)shouldResumeSession;
/** Called on a system notification thread when AVAudioSession changes the
* route.
*/
- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
reason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
/** Called on a system notification thread when AVAudioSession media server
* terminates.
*/
- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
/** Called on a system notification thread when AVAudioSession media server
* restarts.
*/
- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
/** Called on a WebRTC thread when the audio device is notified to begin
* playback or recording.
*/
- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
/** Called on a WebRTC thread when the audio device is notified to stop
* playback or recording.
*/
- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
/** Called when the AVAudioSession output volume value changes. */
- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
/** Called when the audio device detects a playout glitch. The argument is the
* number of glitches detected so far in the current audio playout session.
*/
- (void)audioSession:(RTCAudioSession *)audioSession
didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
/** Called when the audio session is about to change the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
/** Called after the audio session sucessfully changed the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
/** Called after the audio session failed to change the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession
failedToSetActive:(BOOL)active
error:(NSError *)error;
@end
/** This is a protocol used to inform RTCAudioSession when the audio session
* activation state has changed outside of RTCAudioSession. The current known use
* case of this is when CallKit activates the audio session for the application
*/
RTC_EXPORT
@protocol RTCAudioSessionActivationDelegate <NSObject>
/** Called when the audio session is activated outside of the app by iOS. */
- (void)audioSessionDidActivate:(AVAudioSession *)session;
/** Called when the audio session is deactivated outside of the app by iOS. */
- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
@end
/** Proxy class for AVAudioSession that adds a locking mechanism similar to
* AVCaptureDevice. This is used to that interleaving configurations between
* WebRTC and the application layer are avoided.
*
* RTCAudioSession also coordinates activation so that the audio session is
* activated only once. See |setActive:error:|.
*/
RTC_EXPORT
@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
/** Convenience property to access the AVAudioSession singleton. Callers should
* not call setters on AVAudioSession directly, but other method invocations
* are fine.
*/
@property(nonatomic, readonly) AVAudioSession *session;
/** Our best guess at whether the session is active based on results of calls to
* AVAudioSession.
*/
@property(nonatomic, readonly) BOOL isActive;
/** Whether RTCAudioSession is currently locked for configuration. */
@property(nonatomic, readonly) BOOL isLocked;
/** If YES, WebRTC will not initialize the audio unit automatically when an
* audio track is ready for playout or recording. Instead, applications should
* call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
* as soon as an audio track is ready for playout or recording.
*/
@property(nonatomic, assign) BOOL useManualAudio;
/** This property is only effective if useManualAudio is YES.
* Represents permission for WebRTC to initialize the VoIP audio unit.
* When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
* stopped and uninitialized. This will stop incoming and outgoing audio.
* When set to YES, WebRTC will initialize and start the audio unit when it is
* needed (e.g. due to establishing an audio connection).
* This property was introduced to work around an issue where if an AVPlayer is
* playing audio while the VoIP audio unit is initialized, its audio would be
* either cut off completely or played at a reduced volume. By preventing
* the audio unit from being initialized until after the audio has completed,
* we are able to prevent the abrupt cutoff.
*/
@property(nonatomic, assign) BOOL isAudioEnabled;
// Proxy properties.
@property(readonly) NSString *category;
@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
@property(readonly) NSString *mode;
@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
@property(readonly) AVAudioSessionRouteDescription *currentRoute;
@property(readonly) NSInteger maximumInputNumberOfChannels;
@property(readonly) NSInteger maximumOutputNumberOfChannels;
@property(readonly) float inputGain;
@property(readonly) BOOL inputGainSettable;
@property(readonly) BOOL inputAvailable;
@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
@property(readonly) double sampleRate;
@property(readonly) double preferredSampleRate;
@property(readonly) NSInteger inputNumberOfChannels;
@property(readonly) NSInteger outputNumberOfChannels;
@property(readonly) float outputVolume;
@property(readonly) NSTimeInterval inputLatency;
@property(readonly) NSTimeInterval outputLatency;
@property(readonly) NSTimeInterval IOBufferDuration;
@property(readonly) NSTimeInterval preferredIOBufferDuration;
/** Default constructor. */
+ (instancetype)sharedInstance;
- (instancetype)init NS_UNAVAILABLE;
/** Adds a delegate, which is held weakly. */
- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Removes an added delegate. */
- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Request exclusive access to the audio session for configuration. This call
* will block if the lock is held by another object.
*/
- (void)lockForConfiguration;
/** Relinquishes exclusive access to the audio session. */
- (void)unlockForConfiguration;
/** If |active|, activates the audio session if it isn't already active.
* Successful calls must be balanced with a setActive:NO when activation is no
* longer required. If not |active|, deactivates the audio session if one is
* active and this is the last balanced call. When deactivating, the
* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
* AVAudioSession.
*/
- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
// The following methods are proxies for the associated methods on
// AVAudioSession. |lockForConfiguration| must be called before using them
// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
- (BOOL)setCategory:(NSString *)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
@end
@interface RTCAudioSession (Configuration)
/** Applies the configuration to the current session. Attempts to set all
* properties even if previous ones fail. Only the last error will be
* returned.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
/** Convenience method that calls both setConfiguration and setActive.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
active:(BOOL)active
error:(NSError **)outError;
@end
NS_ASSUME_NONNULL_END
#import "components/audio/RTCAudioSession.h"

39
sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
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@ -8,41 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import "WebRTC/RTCMacros.h"
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
// Struct to hold configuration values.
RTC_EXPORT
@interface RTCAudioSessionConfiguration : NSObject
@property(nonatomic, strong) NSString *category;
@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
@property(nonatomic, strong) NSString *mode;
@property(nonatomic, assign) double sampleRate;
@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
@property(nonatomic, assign) NSInteger inputNumberOfChannels;
@property(nonatomic, assign) NSInteger outputNumberOfChannels;
/** Initializes configuration to defaults. */
- (instancetype)init NS_DESIGNATED_INITIALIZER;
/** Returns the current configuration of the audio session. */
+ (instancetype)currentConfiguration;
/** Returns the configuration that WebRTC needs. */
+ (instancetype)webRTCConfiguration;
/** Provide a way to override the default configuration. */
+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
@end
NS_ASSUME_NONNULL_END
#import "components/audio/RTCAudioSessionConfiguration.h"

23
sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
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@ -8,25 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCAudioSource : RTCMediaSource
- (instancetype)init NS_UNAVAILABLE;
// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
// [0, 10].
// Temporary fix to be able to modify volume of remote audio tracks.
// TODO(kthelgason): Property stays here temporarily until a proper volume-api
// is available on the surface exposed by webrtc.
@property(nonatomic, assign) double volume;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCAudioSource.h"

19
sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSource;
RTC_EXPORT
@interface RTCAudioTrack : RTCMediaStreamTrack
- (instancetype)init NS_UNAVAILABLE;
/** The audio source for this audio track. */
@property(nonatomic, readonly) RTCAudioSource *source;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCAudioTrack.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCVPixelBuffer.h Normal file
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@ -0,0 +1,11 @@
/*
* 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 "components/video_frame_buffer/RTCCVPixelBuffer.h"

26
sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h
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@ -8,28 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCLogging.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and forwards them to a registered block.
// This class is not threadsafe.
RTC_EXPORT
@interface RTCCallbackLogger : NSObject
// The severity level to capture. The default is kRTCLoggingSeverityInfo.
@property(nonatomic, assign) RTCLoggingSeverity severity;
// The callback will be called on the same thread that does the logging, so
// if the logging callback can be slow it may be a good idea to implement
// dispatching to some other queue.
- (void)start:(nullable void (^)(NSString*))callback;
- (void)stop;
@end
NS_ASSUME_NONNULL_END
#import "api/logging/RTCCallbackLogger.h"

21
sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
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@ -8,23 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
@class AVCaptureSession;
/** RTCCameraPreviewView is a view that renders local video from an
* AVCaptureSession.
*/
RTC_EXPORT
@interface RTCCameraPreviewView : UIView
/** The capture session being rendered in the view. Capture session
* is assigned to AVCaptureVideoPreviewLayer async in the same
* queue that the AVCaptureSession is started/stopped.
*/
@property(nonatomic, strong) AVCaptureSession* captureSession;
@end
#import "helpers/RTCCameraPreviewView.h"

47
sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
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@ -8,49 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
// Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
// (usually RTCVideoSource).
NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
@interface RTCCameraVideoCapturer : RTCVideoCapturer
// Capture session that is used for capturing. Valid from initialization to dealloc.
@property(readonly, nonatomic) AVCaptureSession *captureSession;
// Returns list of available capture devices that support video capture.
+ (NSArray<AVCaptureDevice *> *)captureDevices;
// Returns list of formats that are supported by this class for this device.
+ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
// Returns the most efficient supported output pixel format for this capturer.
- (FourCharCode)preferredOutputPixelFormat;
// Starts the capture session asynchronously and notifies callback on completion.
// The device will capture video in the format given in the `format` parameter. If the pixel format
// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps
completionHandler:(nullable void (^)(NSError *))completionHandler;
// Stops the capture session asynchronously and notifies callback on completion.
- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
// Starts the capture session asynchronously.
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps;
// Stops the capture session asynchronously.
- (void)stopCapture;
@end
NS_ASSUME_NONNULL_END
#import "components/capturer/RTCCameraVideoCapturer.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCCertificate : NSObject <NSCopying>
/** Private key in PEM. */
@property(nonatomic, readonly, copy) NSString *private_key;
/** Public key in an x509 cert encoded in PEM. */
@property(nonatomic, readonly, copy) NSString *certificate;
/**
* Initialize an RTCCertificate with PEM strings for private_key and certificate.
*/
- (instancetype)initWithPrivateKey:(NSString *)private_key
certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
- (instancetype)init NS_UNAVAILABLE;
/** Generate a new certificate for 're' use.
*
* Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
* provided.
* - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
*/
+ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCCertificate.h"

165
sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
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@ -8,167 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCCertificate.h>
#import <WebRTC/RTCMacros.h>
@class RTCIceServer;
@class RTCIntervalRange;
/**
* Represents the ice transport policy. This exposes the same states in C++,
* which include one more state than what exists in the W3C spec.
*/
typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
RTCIceTransportPolicyNone,
RTCIceTransportPolicyRelay,
RTCIceTransportPolicyNoHost,
RTCIceTransportPolicyAll
};
/** Represents the bundle policy. */
typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
RTCBundlePolicyBalanced,
RTCBundlePolicyMaxCompat,
RTCBundlePolicyMaxBundle
};
/** Represents the rtcp mux policy. */
typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
/** Represents the tcp candidate policy. */
typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
RTCTcpCandidatePolicyEnabled,
RTCTcpCandidatePolicyDisabled
};
/** Represents the candidate network policy. */
typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
RTCCandidateNetworkPolicyAll,
RTCCandidateNetworkPolicyLowCost
};
/** Represents the continual gathering policy. */
typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
RTCContinualGatheringPolicyGatherOnce,
RTCContinualGatheringPolicyGatherContinually
};
/** Represents the encryption key type. */
typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
RTCEncryptionKeyTypeRSA,
RTCEncryptionKeyTypeECDSA,
};
/** Represents the chosen SDP semantics for the RTCPeerConnection. */
typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
RTCSdpSemanticsPlanB,
RTCSdpSemanticsUnifiedPlan,
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCConfiguration : NSObject
/** An array of Ice Servers available to be used by ICE. */
@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
/** An RTCCertificate for 're' use. */
@property(nonatomic, nullable) RTCCertificate *certificate;
/** Which candidates the ICE agent is allowed to use. The W3C calls it
* |iceTransportPolicy|, while in C++ it is called |type|. */
@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
/** The media-bundling policy to use when gathering ICE candidates. */
@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
/** The rtcp-mux policy to use when gathering ICE candidates. */
@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
/** By default, the PeerConnection will use a limited number of IPv6 network
* interfaces, in order to avoid too many ICE candidate pairs being created
* and delaying ICE completion.
*
* Can be set to INT_MAX to effectively disable the limit.
*/
@property(nonatomic, assign) int maxIPv6Networks;
/** Exclude link-local network interfaces
* from considertaion for gathering ICE candidates.
* Defaults to NO.
*/
@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
@property(nonatomic, assign) int audioJitterBufferMaxPackets;
@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
@property(nonatomic, assign) int iceConnectionReceivingTimeout;
@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
/** Key type used to generate SSL identity. Default is ECDSA. */
@property(nonatomic, assign) RTCEncryptionKeyType keyType;
/** ICE candidate pool size as defined in JSEP. Default is 0. */
@property(nonatomic, assign) int iceCandidatePoolSize;
/** Prune turn ports on the same network to the same turn server.
* Default is NO.
*/
@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
/** If set to YES, this means the ICE transport should presume TURN-to-TURN
* candidate pairs will succeed, even before a binding response is received.
*/
@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
/** If set to non-nil, controls the minimal interval between consecutive ICE
* check packets.
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
/** ICE Periodic Regathering
* If set, WebRTC will periodically create and propose candidates without
* starting a new ICE generation. The regathering happens continuously with
* interval specified in milliseconds by the uniform distribution [a, b].
*/
@property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
/** Configure the SDP semantics used by this PeerConnection. Note that the
* WebRTC 1.0 specification requires UnifiedPlan semantics. The
* RTCRtpTransceiver API is only available with UnifiedPlan semantics.
*
* PlanB will cause RTCPeerConnection to create offers and answers with at
* most one audio and one video m= section with multiple RTCRtpSenders and
* RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
* will also cause RTCPeerConnection to ignore all but the first m= section of
* the same media type.
*
* UnifiedPlan will cause RTCPeerConnection to create offers and answers with
* multiple m= sections where each m= section maps to one RTCRtpSender and one
* RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
* will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
* that form a Plan B stream.
*
* For users who wish to send multiple audio/video streams and need to stay
* interoperable with legacy WebRTC implementations or use legacy APIs,
* specify PlanB.
*
* For all other users, specify UnifiedPlan.
*/
@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
/** Actively reset the SRTP parameters when the DTLS transports underneath are
* changed after offer/answer negotiation. This is only intended to be a
* workaround for crbug.com/835958
*/
@property(nonatomic, assign) BOOL activeResetSrtpParams;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCConfiguration.h"

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sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
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@ -8,123 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCDataBuffer : NSObject
/** NSData representation of the underlying buffer. */
@property(nonatomic, readonly) NSData *data;
/** Indicates whether |data| contains UTF-8 or binary data. */
@property(nonatomic, readonly) BOOL isBinary;
- (instancetype)init NS_UNAVAILABLE;
/**
* Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
* contains UTF-8 or binary data.
*/
- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
@end
@class RTCDataChannel;
RTC_EXPORT
@protocol RTCDataChannelDelegate <NSObject>
/** The data channel state changed. */
- (void)dataChannelDidChangeState:(RTCDataChannel *)dataChannel;
/** The data channel successfully received a data buffer. */
- (void)dataChannel:(RTCDataChannel *)dataChannel
didReceiveMessageWithBuffer:(RTCDataBuffer *)buffer;
@optional
/** The data channel's |bufferedAmount| changed. */
- (void)dataChannel:(RTCDataChannel *)dataChannel didChangeBufferedAmount:(uint64_t)amount;
@end
/** Represents the state of the data channel. */
typedef NS_ENUM(NSInteger, RTCDataChannelState) {
RTCDataChannelStateConnecting,
RTCDataChannelStateOpen,
RTCDataChannelStateClosing,
RTCDataChannelStateClosed,
};
RTC_EXPORT
@interface RTCDataChannel : NSObject
/**
* A label that can be used to distinguish this data channel from other data
* channel objects.
*/
@property(nonatomic, readonly) NSString *label;
/** Whether the data channel can send messages in unreliable mode. */
@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
/** Returns whether this data channel is ordered or not. */
@property(nonatomic, readonly) BOOL isOrdered;
/** Deprecated. Use maxPacketLifeTime. */
@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
/**
* The length of the time window (in milliseconds) during which transmissions
* and retransmissions may occur in unreliable mode.
*/
@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
/**
* The maximum number of retransmissions that are attempted in unreliable mode.
*/
@property(nonatomic, readonly) uint16_t maxRetransmits;
/**
* The name of the sub-protocol used with this data channel, if any. Otherwise
* this returns an empty string.
*/
@property(nonatomic, readonly) NSString *protocol;
/**
* Returns whether this data channel was negotiated by the application or not.
*/
@property(nonatomic, readonly) BOOL isNegotiated;
/** Deprecated. Use channelId. */
@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
/** The identifier for this data channel. */
@property(nonatomic, readonly) int channelId;
/** The state of the data channel. */
@property(nonatomic, readonly) RTCDataChannelState readyState;
/**
* The number of bytes of application data that have been queued using
* |sendData:| but that have not yet been transmitted to the network.
*/
@property(nonatomic, readonly) uint64_t bufferedAmount;
/** The delegate for this data channel. */
@property(nonatomic, weak) id<RTCDataChannelDelegate> delegate;
- (instancetype)init NS_UNAVAILABLE;
/** Closes the data channel. */
- (void)close;
/** Attempt to send |data| on this data channel's underlying data transport. */
- (BOOL)sendData:(RTCDataBuffer *)data;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCDataChannel.h"

43
sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
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@ -8,45 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCDataChannelConfiguration : NSObject
/** Set to YES if ordered delivery is required. */
@property(nonatomic, assign) BOOL isOrdered;
/** Deprecated. Use maxPacketLifeTime. */
@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
/**
* Max period in milliseconds in which retransmissions will be sent. After this
* time, no more retransmissions will be sent. -1 if unset.
*/
@property(nonatomic, assign) int maxPacketLifeTime;
/** The max number of retransmissions. -1 if unset. */
@property(nonatomic, assign) int maxRetransmits;
/** Set to YES if the channel has been externally negotiated and we do not send
* an in-band signalling in the form of an "open" message.
*/
@property(nonatomic, assign) BOOL isNegotiated;
/** Deprecated. Use channelId. */
@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
/** The id of the data channel. */
@property(nonatomic, assign) int channelId;
/** Set by the application and opaque to the WebRTC implementation. */
@property(nonatomic) NSString* protocol;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCDataChannelConfiguration.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoDecoderFactory.h Normal file
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@ -0,0 +1,11 @@
/*
* Copyright 2017 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 "components/video_codec/RTCDefaultVideoDecoderFactory.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoEncoderFactory.h Normal file
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@ -0,0 +1,11 @@
/*
* Copyright 2017 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 "components/video_codec/RTCDefaultVideoEncoderFactory.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
// Main dispatcher queue.
RTCDispatcherTypeMain,
// Used for starting/stopping AVCaptureSession, and assigning
// capture session to AVCaptureVideoPreviewLayer.
RTCDispatcherTypeCaptureSession,
// Used for operations on AVAudioSession.
RTCDispatcherTypeAudioSession,
};
/** Dispatcher that asynchronously dispatches blocks to a specific
* shared dispatch queue.
*/
RTC_EXPORT
@interface RTCDispatcher : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Dispatch the block asynchronously on the queue for dispatchType.
* @param dispatchType The queue type to dispatch on.
* @param block The block to dispatch asynchronously.
*/
+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
/** Returns YES if run on queue for the dispatchType otherwise NO.
* Useful for asserting that a method is run on a correct queue.
*/
+ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
@end
#import "helpers/RTCDispatcher.h"

61
sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
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@ -8,63 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@protocol RTCDtmfSender <NSObject>
/**
* Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
* returns false. To be able to send DTMF, the associated RTCRtpSender must be
* able to send packets, and a "telephone-event" codec must be negotiated.
*/
@property(nonatomic, readonly) BOOL canInsertDtmf;
/**
* Queues a task that sends the DTMF tones. The tones parameter is treated
* as a series of characters. The characters 0 through 9, A through D, #, and *
* generate the associated DTMF tones. The characters a to d are equivalent
* to A to D. The character ',' indicates a delay of 2 seconds before
* processing the next character in the tones parameter.
*
* Unrecognized characters are ignored.
*
* @param duration The parameter indicates the duration to use for each
* character passed in the tones parameter. The duration cannot be more
* than 6000 or less than 70 ms.
*
* @param interToneGap The parameter indicates the gap between tones.
* This parameter must be at least 50 ms but should be as short as
* possible.
*
* If InsertDtmf is called on the same object while an existing task for this
* object to generate DTMF is still running, the previous task is canceled.
* Returns true on success and false on failure.
*/
- (BOOL)insertDtmf:(nonnull NSString *)tones
duration:(NSTimeInterval)duration
interToneGap:(NSTimeInterval)interToneGap;
/** The tones remaining to be played out */
- (nonnull NSString *)remainingTones;
/**
* The current tone duration value. This value will be the value last set via the
* insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
*/
- (NSTimeInterval)duration;
/**
* The current value of the between-tone gap. This value will be the value last set
* via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
* called.
*/
- (NSTimeInterval)interToneGap;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCDtmfSender.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCVideoViewShading.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCEAGLVideoView;
RTC_EXPORT
@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
@end
/**
* RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
* bounds using OpenGLES 2.0 or OpenGLES 3.0.
*/
RTC_EXPORT
NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
- (instancetype)initWithFrame:(CGRect)frame
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithCoder:(NSCoder *)aDecoder
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCEAGLVideoView.h"

37
sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
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@ -8,39 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
RTC_EXTERN NSString * const kRTCFieldTrialImprovedBitrateEstimateKey;
RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
/** The valid value for field trials above. */
RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
/** Use a string returned by RTCFieldTrialMedianSlopeFilterValue as the value. */
RTC_EXTERN NSString * const kRTCFieldTrialMedianSlopeFilterKey;
RTC_EXTERN NSString *RTCFieldTrialMedianSlopeFilterValue(
size_t windowSize, double thresholdGain);
/** Use a string returned by RTCFieldTrialTrendlineFilterValue as the value. */
RTC_EXTERN NSString * const kRTCFieldTrialTrendlineFilterKey;
/** Returns a valid value for kRTCFieldTrialTrendlineFilterKey. */
RTC_EXTERN NSString *RTCFieldTrialTrendlineFilterValue(
size_t windowSize, double smoothingCoeff, double thresholdGain);
/** Initialize field trials using a dictionary mapping field trial keys to their values. See above
* for valid keys and values.
* Must be called before any other call into WebRTC. See:
* webrtc/system_wrappers/include/field_trial_default.h
*/
RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
#import "api/peerconnection/RTCFieldTrials.h"

65
sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
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@ -8,67 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
RTCFileLoggerSeverityVerbose,
RTCFileLoggerSeverityInfo,
RTCFileLoggerSeverityWarning,
RTCFileLoggerSeverityError
};
typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
RTCFileLoggerTypeCall,
RTCFileLoggerTypeApp,
};
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and saves them to a file. The file size
// will not exceed the given maximum bytesize. When the maximum bytesize is
// reached, logs are rotated according to the rotationType specified.
// For kRTCFileLoggerTypeCall, logs from the beginning and the end
// are preserved while the middle section is overwritten instead.
// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
// This class is not threadsafe.
RTC_EXPORT
@interface RTCFileLogger : NSObject
// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
@property(nonatomic, assign) RTCFileLoggerSeverity severity;
// The rotation type for this file logger. The default is
// kRTCFileLoggerTypeCall.
@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
// Disables buffering disk writes. Should be set before |start|. Buffering
// is enabled by default for performance.
@property(nonatomic, assign) BOOL shouldDisableBuffering;
// Default constructor provides default settings for dir path, file size and
// rotation type.
- (instancetype)init;
// Create file logger with default rotation type.
- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
- (instancetype)initWithDirPath:(NSString *)dirPath
maxFileSize:(NSUInteger)maxFileSize
rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
// Starts writing WebRTC logs to disk if not already started. Overwrites any
// existing file(s).
- (void)start;
// Stops writing WebRTC logs to disk. This method is also called on dealloc.
- (void)stop;
// Returns the current contents of the logs, or nil if start has been called
// without a stop.
- (nullable NSData *)logData;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCFileLogger.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
View File

@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
/**
* Error passing block.
*/
typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
/**
* Captures buffers from bundled video file.
*
* See @c RTCVideoCapturer for more info on capturers.
*/
RTC_EXPORT
NS_CLASS_AVAILABLE_IOS(10)
@interface RTCFileVideoCapturer : RTCVideoCapturer
/**
* Starts asynchronous capture of frames from video file.
*
* Capturing is not started if error occurs. Underlying error will be
* relayed in the errorBlock if one is provided.
* Successfully captured video frames will be passed to the delegate.
*
* @param nameOfFile The name of the bundled video file to be read.
* @errorBlock block to be executed upon error.
*/
- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
/**
* Immediately stops capture.
*/
- (void)stopCapture;
@end
NS_ASSUME_NONNULL_END
#import "components/capturer/RTCFileVideoCapturer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCH264ProfileLevelId.h Normal file
View File

@ -0,0 +1,11 @@
/*
* Copyright 2017 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 "components/video_codec/RTCH264ProfileLevelId.h"

40
sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
View File

@ -8,42 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCIceCandidate : NSObject
/**
* If present, the identifier of the "media stream identification" for the media
* component this candidate is associated with.
*/
@property(nonatomic, readonly, nullable) NSString *sdpMid;
/**
* The index (starting at zero) of the media description this candidate is
* associated with in the SDP.
*/
@property(nonatomic, readonly) int sdpMLineIndex;
/** The SDP string for this candidate. */
@property(nonatomic, readonly) NSString *sdp;
/** The URL of the ICE server which this candidate is gathered from. */
@property(nonatomic, readonly, nullable) NSString *serverUrl;
- (instancetype)init NS_UNAVAILABLE;
/**
* Initialize an RTCIceCandidate from SDP.
*/
- (instancetype)initWithSdp:(NSString *)sdp
sdpMLineIndex:(int)sdpMLineIndex
sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIceCandidate.h"

105
sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
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@ -8,107 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
RTCTlsCertPolicySecure,
RTCTlsCertPolicyInsecureNoCheck
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCIceServer : NSObject
/** URI(s) for this server represented as NSStrings. */
@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
/** Username to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *username;
/** Credential to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *credential;
/**
* TLS certificate policy to use if this RTCIceServer object is a TURN server.
*/
@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
/**
If the URIs in |urls| only contain IP addresses, this field can be used
to indicate the hostname, which may be necessary for TLS (using the SNI
extension). If |urls| itself contains the hostname, this isn't necessary.
*/
@property(nonatomic, readonly, nullable) NSString *hostname;
/** List of protocols to be used in the TLS ALPN extension. */
@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
/**
List elliptic curves to be used in the TLS elliptic curves extension.
Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
*/
@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
- (nonnull instancetype)init NS_UNAVAILABLE;
/** Convenience initializer for a server with no authentication (e.g. STUN). */
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and credentialType.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and TLS cert policy.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy and hostname.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname and ALPN protocols.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname, ALPN protocols and
* elliptic curves.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIceServer.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
NS_ASSUME_NONNULL_BEGIN
@interface RTCIntervalRange : NSObject
@property(nonatomic, readonly) NSInteger min;
@property(nonatomic, readonly) NSInteger max;
- (instancetype)init;
- (instancetype)initWithMin:(NSInteger)min max:(NSInteger)max NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIntervalRange.h"

28
sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** This does not currently conform to the spec. */
RTC_EXPORT
@interface RTCLegacyStatsReport : NSObject
/** Time since 1970-01-01T00:00:00Z in milliseconds. */
@property(nonatomic, readonly) CFTimeInterval timestamp;
/** The type of stats held by this object. */
@property(nonatomic, readonly) NSString *type;
/** The identifier for this object. */
@property(nonatomic, readonly) NSString *reportId;
/** A dictionary holding the actual stats. */
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCLegacyStatsReport.h"

57
sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
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@ -8,59 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
// Subset of rtc::LoggingSeverity.
typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
RTCLoggingSeverityVerbose,
RTCLoggingSeverityInfo,
RTCLoggingSeverityWarning,
RTCLoggingSeverityError,
};
// Wrapper for C++ RTC_LOG(sev) macros.
// Logs the log string to the webrtc logstream for the given severity.
RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
// Wrapper for rtc::LogMessage::LogToDebug.
// Sets the minimum severity to be logged to console.
RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
// Returns the filename with the path prefix removed.
RTC_EXTERN NSString* RTCFileName(const char* filePath);
// Some convenience macros.
#define RTCLogString(format, ...) \
[NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
__LINE__, __FUNCTION__, ##__VA_ARGS__]
#define RTCLogFormat(severity, format, ...) \
do { \
NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
RTCLogEx(severity, log_string); \
} while (false)
#define RTCLogVerbose(format, ...) \
RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
#define RTCLogInfo(format, ...) \
RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
#define RTCLogWarning(format, ...) \
RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
#define RTCLogError(format, ...) \
RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
#if !defined(NDEBUG)
#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
#else
#define RTCLogDebug(format, ...) \
do { \
} while (false)
#endif
#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
#import "base/RTCLogging.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
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@ -8,13 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Cocoa/Cocoa.h>
#import "WebRTC/RTCVideoRenderer.h"
NS_AVAILABLE_MAC(10.11)
@interface RTCMTLNSVideoView : NSView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
+ (BOOL)isMetalAvailable;
@end
#import "components/renderer/metal/RTCMTLNSVideoView.h"

42
sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
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@ -8,44 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import "WebRTC/RTCVideoFrame.h"
#import "WebRTC/RTCVideoRenderer.h"
// Check if metal is supported in WebRTC.
// NOTE: Currently arm64 == Metal.
#if defined(__aarch64__)
#define RTC_SUPPORTS_METAL
#endif
NS_ASSUME_NONNULL_BEGIN
/**
* RTCMTLVideoView is thin wrapper around MTKView.
*
* It has id<RTCVideoRenderer> property that renders video frames in the view's
* bounds using Metal.
* NOTE: always check if metal is available on the running device via
* RTC_SUPPORTS_METAL macro before initializing this class.
*/
NS_CLASS_AVAILABLE_IOS(9)
RTC_EXPORT
@interface RTCMTLVideoView : UIView<RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
@property(nonatomic) UIViewContentMode videoContentMode;
/** @abstract Enables/disables rendering.
*/
@property(nonatomic, getter=isEnabled) BOOL enabled;
/** @abstract Wrapped RTCVideoRotation, or nil.
*/
@property(nonatomic, nullable) NSValue* rotationOverride;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/metal/RTCMTLVideoView.h"

19
sdk/objc/Framework/Headers/WebRTC/RTCMacros.h
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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#define SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#define RTC_EXPORT __attribute__((visibility("default")))
#if defined(__cplusplus)
#define RTC_EXTERN extern "C" RTC_EXPORT
#else
#define RTC_EXTERN extern RTC_EXPORT
#endif
#ifdef __OBJC__
#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
#else
#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
#endif
#endif // SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#import "base/RTCMacros.h"

45
sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
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@ -8,47 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Constraint keys for media sources. */
RTC_EXTERN NSString *const kRTCMediaConstraintsMinAspectRatio;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxAspectRatio;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxWidth;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinWidth;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxHeight;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinHeight;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxFrameRate;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinFrameRate;
/** The value for this key should be a base64 encoded string containing
* the data from the serialized configuration proto.
*/
RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
/** Constraint keys for generating offers and answers. */
RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
/** Constraint values for Boolean parameters. */
RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
RTC_EXPORT
@interface RTCMediaConstraints : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Initialize with mandatory and/or optional constraints. */
- (instancetype)
initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaConstraints.h"

25
sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
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@ -8,27 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCSourceState) {
RTCSourceStateInitializing,
RTCSourceStateLive,
RTCSourceStateEnded,
RTCSourceStateMuted,
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCMediaSource : NSObject
/** The current state of the RTCMediaSource. */
@property(nonatomic, readonly) RTCSourceState state;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaSource.h"

40
sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
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@ -8,42 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioTrack;
@class RTCPeerConnectionFactory;
@class RTCVideoTrack;
RTC_EXPORT
@interface RTCMediaStream : NSObject
/** The audio tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTCAudioTrack *> *audioTracks;
/** The video tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTCVideoTrack *> *videoTracks;
/** An identifier for this media stream. */
@property(nonatomic, readonly) NSString *streamId;
- (instancetype)init NS_UNAVAILABLE;
/** Adds the given audio track to this media stream. */
- (void)addAudioTrack:(RTCAudioTrack *)audioTrack;
/** Adds the given video track to this media stream. */
- (void)addVideoTrack:(RTCVideoTrack *)videoTrack;
/** Removes the given audio track to this media stream. */
- (void)removeAudioTrack:(RTCAudioTrack *)audioTrack;
/** Removes the given video track to this media stream. */
- (void)removeVideoTrack:(RTCVideoTrack *)videoTrack;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaStream.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
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@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the state of the track. This exposes the same states in C++.
*/
typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
RTCMediaStreamTrackStateLive,
RTCMediaStreamTrackStateEnded
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
RTC_EXPORT
@interface RTCMediaStreamTrack : NSObject
/**
* The kind of track. For example, "audio" if this track represents an audio
* track and "video" if this track represents a video track.
*/
@property(nonatomic, readonly) NSString *kind;
/** An identifier string. */
@property(nonatomic, readonly) NSString *trackId;
/** The enabled state of the track. */
@property(nonatomic, assign) BOOL isEnabled;
/** The state of the track. */
@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaStreamTrack.h"

14
sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
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@ -8,16 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMetricsSampleInfo.h>
/**
* Enables gathering of metrics (which can be fetched with
* RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
*/
RTC_EXTERN void RTCEnableMetrics(void);
/** Gets and clears native histograms. */
RTC_EXTERN NSArray<RTCMetricsSampleInfo*>* RTCGetAndResetMetrics(void);
#import "api/peerconnection/RTCMetrics.h"

39
sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
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@ -8,41 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCMetricsSampleInfo : NSObject
/**
* Example of RTCMetricsSampleInfo:
* name: "WebRTC.Video.InputFramesPerSecond"
* min: 1
* max: 100
* bucketCount: 50
* samples: [29]:2 [30]:1
*/
/** The name of the histogram. */
@property(nonatomic, readonly) NSString *name;
/** The minimum bucket value. */
@property(nonatomic, readonly) int min;
/** The maximum bucket value. */
@property(nonatomic, readonly) int max;
/** The number of buckets. */
@property(nonatomic, readonly) int bucketCount;
/** A dictionary holding the samples <value, # of events>. */
@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMetricsSampleInfo.h"

30
sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
View File

@ -8,32 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#if !TARGET_OS_IPHONE
#import <AppKit/NSOpenGLView.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCVideoViewShading.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCNSGLVideoView;
@protocol RTCNSGLVideoViewDelegate <RTCVideoViewDelegate>
@end
@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
- (instancetype)initWithFrame:(NSRect)frameRect
pixelFormat:(NSOpenGLPixelFormat *)format
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#endif
#import "components/renderer/opengl/RTCNSGLVideoView.h"

307
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
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@ -8,309 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
@class RTCConfiguration;
@class RTCDataChannel;
@class RTCDataChannelConfiguration;
@class RTCIceCandidate;
@class RTCMediaConstraints;
@class RTCMediaStream;
@class RTCMediaStreamTrack;
@class RTCPeerConnectionFactory;
@class RTCRtpReceiver;
@class RTCRtpSender;
@class RTCRtpTransceiver;
@class RTCRtpTransceiverInit;
@class RTCSessionDescription;
@class RTCLegacyStatsReport;
typedef NS_ENUM(NSInteger, RTCRtpMediaType);
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCPeerConnectionErrorDomain;
extern int const kRTCSessionDescriptionErrorCode;
/** Represents the signaling state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCSignalingState) {
RTCSignalingStateStable,
RTCSignalingStateHaveLocalOffer,
RTCSignalingStateHaveLocalPrAnswer,
RTCSignalingStateHaveRemoteOffer,
RTCSignalingStateHaveRemotePrAnswer,
// Not an actual state, represents the total number of states.
RTCSignalingStateClosed,
};
/** Represents the ice connection state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
RTCIceConnectionStateNew,
RTCIceConnectionStateChecking,
RTCIceConnectionStateConnected,
RTCIceConnectionStateCompleted,
RTCIceConnectionStateFailed,
RTCIceConnectionStateDisconnected,
RTCIceConnectionStateClosed,
RTCIceConnectionStateCount,
};
/** Represents the ice gathering state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
RTCIceGatheringStateNew,
RTCIceGatheringStateGathering,
RTCIceGatheringStateComplete,
};
/** Represents the stats output level. */
typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
RTCStatsOutputLevelStandard,
RTCStatsOutputLevelDebug,
};
@class RTCPeerConnection;
RTC_EXPORT
@protocol RTCPeerConnectionDelegate <NSObject>
/** Called when the SignalingState changed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeSignalingState:(RTCSignalingState)stateChanged;
/** Called when media is received on a new stream from remote peer. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream;
/** Called when a remote peer closes a stream.
* This is not called when RTCSdpSemanticsUnifiedPlan is specified.
*/
- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream;
/** Called when negotiation is needed, for example ICE has restarted. */
- (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection;
/** Called any time the IceConnectionState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeIceConnectionState:(RTCIceConnectionState)newState;
/** Called any time the IceGatheringState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeIceGatheringState:(RTCIceGatheringState)newState;
/** New ice candidate has been found. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didGenerateIceCandidate:(RTCIceCandidate *)candidate;
/** Called when a group of local Ice candidates have been removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didRemoveIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
/** New data channel has been opened. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didOpenDataChannel:(RTCDataChannel *)dataChannel;
/** Called when signaling indicates a transceiver will be receiving media from
* the remote endpoint.
* This is only called with RTCSdpSemanticsUnifiedPlan specified.
*/
@optional
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver;
/** Called when a receiver and its track are created. */
@optional
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didAddReceiver:(RTCRtpReceiver *)rtpReceiver
streams:(NSArray<RTCMediaStream *> *)mediaStreams;
/** Called when the receiver and its track are removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didRemoveReceiver:(RTCRtpReceiver *)rtpReceiver;
@end
RTC_EXPORT
@interface RTCPeerConnection : NSObject
/** The object that will be notifed about events such as state changes and
* streams being added or removed.
*/
@property(nonatomic, weak, nullable) id<RTCPeerConnectionDelegate> delegate;
/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
* |senders| instead.
*/
@property(nonatomic, readonly) NSArray<RTCMediaStream *> *localStreams;
@property(nonatomic, readonly, nullable) RTCSessionDescription *localDescription;
@property(nonatomic, readonly, nullable) RTCSessionDescription *remoteDescription;
@property(nonatomic, readonly) RTCSignalingState signalingState;
@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
@property(nonatomic, readonly, copy) RTCConfiguration *configuration;
/** Gets all RTCRtpSenders associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpSender.
* Use isEqual: instead of == to compare RTCRtpSender instances.
*/
@property(nonatomic, readonly) NSArray<RTCRtpSender *> *senders;
/** Gets all RTCRtpReceivers associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpReceiver.
* Use isEqual: instead of == to compare RTCRtpReceiver instances.
*/
@property(nonatomic, readonly) NSArray<RTCRtpReceiver *> *receivers;
/** Gets all RTCRtpTransceivers associated with this peer connection.
* Note: reading this property returns different instances of
* RTCRtpTransceiver. Use isEqual: instead of == to compare RTCRtpTransceiver
* instances.
* This is only available with RTCSdpSemanticsUnifiedPlan specified.
*/
@property(nonatomic, readonly) NSArray<RTCRtpTransceiver *> *transceivers;
- (instancetype)init NS_UNAVAILABLE;
/** Sets the PeerConnection's global configuration to |configuration|.
* Any changes to STUN/TURN servers or ICE candidate policy will affect the
* next gathering phase, and cause the next call to createOffer to generate
* new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
* cannot be changed with this method.
*/
- (BOOL)setConfiguration:(RTCConfiguration *)configuration;
/** Terminate all media and close the transport. */
- (void)close;
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTCIceCandidate *)candidate;
/** Remove a group of remote candidates from the ICE Agent. */
- (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
/** Add a new media stream to be sent on this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTrack instead.
*/
- (void)addStream:(RTCMediaStream *)stream;
/** Remove the given media stream from this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* removeTrack instead.
*/
- (void)removeStream:(RTCMediaStream *)stream;
/** Add a new media stream track to be sent on this peer connection, and return
* the newly created RTCRtpSender. The RTCRtpSender will be associated with
* the streams specified in the |streamIds| list.
*
* Errors: If an error occurs, returns nil. An error can occur if:
* - A sender already exists for the track.
* - The peer connection is closed.
*/
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds;
/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
*
* With UnifiedPlan semantics, sets sender's track to null and removes the
* send component from the associated RTCRtpTransceiver's direction.
*
* Returns YES on success.
*/
- (BOOL)removeTrack:(RTCRtpSender *)sender;
/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
* transceivers. Adding a transceiver will cause future calls to CreateOffer
* to add a media description for the corresponding transceiver.
*
* The initial value of |mid| in the returned transceiver is nil. Setting a
* new session description may change it to a non-nil value.
*
* https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
*
* Optionally, an RtpTransceiverInit structure can be specified to configure
* the transceiver from construction. If not specified, the transceiver will
* default to having a direction of kSendRecv and not be part of any streams.
*
* These methods are only available when Unified Plan is enabled (see
* RTCConfiguration).
*/
/** Adds a transceiver with a sender set to transmit the given track. The kind
* of the transceiver (and sender/receiver) will be derived from the kind of
* the track.
*/
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track;
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
init:(RTCRtpTransceiverInit *)init;
/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
* or RTCRtpMediaTypeVideo.
*/
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTCRtpTransceiverInit *)init;
/** Generate an SDP offer. */
- (void)offerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
NSError *_Nullable error))completionHandler;
/** Generate an SDP answer. */
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the local description. */
- (void)setLocalDescription:(RTCSessionDescription *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the remote description. */
- (void)setRemoteDescription:(RTCSessionDescription *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Nil parameters will be unchanged. Setting
* |currentBitrateBps| will force the available bitrate estimate to the given
* value. Returns YES if the parameters were successfully updated.
*/
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
/** Start or stop recording an Rtc EventLog. */
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
- (void)stopRtcEventLog;
@end
@interface RTCPeerConnection (Media)
/** Create an RTCRtpSender with the specified kind and media stream ID.
* See RTCMediaStreamTrack.h for available kinds.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTransceiver instead.
*/
- (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
@end
@interface RTCPeerConnection (DataChannel)
/** Create a new data channel with the given label and configuration. */
- (nullable RTCDataChannel *)dataChannelForLabel:(NSString *)label
configuration:(RTCDataChannelConfiguration *)configuration;
@end
@interface RTCPeerConnection (Stats)
/** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
* statistics are gathered for all tracks.
*/
- (void)statsForTrack:(nullable RTCMediaStreamTrack *)mediaStreamTrack
statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
completionHandler:(nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnection.h"

72
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
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@ -8,74 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSource;
@class RTCAudioTrack;
@class RTCConfiguration;
@class RTCMediaConstraints;
@class RTCMediaStream;
@class RTCPeerConnection;
@class RTCVideoSource;
@class RTCVideoTrack;
@class RTCPeerConnectionFactoryOptions;
@protocol RTCPeerConnectionDelegate;
@protocol RTCVideoDecoderFactory;
@protocol RTCVideoEncoderFactory;
RTC_EXPORT
@interface RTCPeerConnectionFactory : NSObject
/* Initialize object with default H264 video encoder/decoder factories */
- (instancetype)init;
/* Initialize object with injectable video encoder/decoder factories */
- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory;
/** Initialize an RTCAudioSource with constraints. */
- (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints;
/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source with no
* constraints.
*/
- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId;
/** Initialize an RTCAudioTrack with a source and an id. */
- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source trackId:(NSString *)trackId;
/** Initialize a generic RTCVideoSource. The RTCVideoSource should be passed to a RTCVideoCapturer
* implementation, e.g. RTCCameraVideoCapturer, in order to produce frames.
*/
- (RTCVideoSource *)videoSource;
/** Initialize an RTCVideoTrack with a source and an id. */
- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source trackId:(NSString *)trackId;
/** Initialize an RTCMediaStream with an id. */
- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:
(nullable id<RTCPeerConnectionDelegate>)delegate;
/** Set the options to be used for subsequently created RTCPeerConnections */
- (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options;
/** Start an AecDump recording. This API call will likely change in the future. */
- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
/* Stop an active AecDump recording */
- (void)stopAecDump;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactory.h"

33
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
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@ -8,35 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCPeerConnectionFactoryOptions : NSObject
@property(nonatomic, assign) BOOL disableEncryption;
@property(nonatomic, assign) BOOL disableNetworkMonitor;
@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
@property(nonatomic, assign) BOOL enableAes128Sha1_32CryptoCipher;
@property(nonatomic, assign) BOOL enableGcmCryptoSuites;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactoryOptions.h"

21
sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
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@ -8,23 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtcpParameters : NSObject
/** The Canonical Name used by RTCP. */
@property(nonatomic, readonly, copy) NSString *cname;
/** Whether reduced size RTCP is configured or compound RTCP. */
@property(nonatomic, assign) BOOL isReducedSize;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtcpParameters.h"

64
sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
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@ -8,66 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const NSString *const kRTCRtxCodecName;
RTC_EXTERN const NSString *const kRTCRedCodecName;
RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
RTC_EXTERN const NSString *const kRTCOpusCodecName;
RTC_EXTERN const NSString *const kRTCIsacCodecName;
RTC_EXTERN const NSString *const kRTCL16CodecName;
RTC_EXTERN const NSString *const kRTCG722CodecName;
RTC_EXTERN const NSString *const kRTCIlbcCodecName;
RTC_EXTERN const NSString *const kRTCPcmuCodecName;
RTC_EXTERN const NSString *const kRTCPcmaCodecName;
RTC_EXTERN const NSString *const kRTCDtmfCodecName;
RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
RTC_EXTERN const NSString *const kRTCVp8CodecName;
RTC_EXTERN const NSString *const kRTCVp9CodecName;
RTC_EXTERN const NSString *const kRTCH264CodecName;
/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
RTC_EXPORT
@interface RTCRtpCodecParameters : NSObject
/** The RTP payload type. */
@property(nonatomic, assign) int payloadType;
/**
* The codec MIME subtype. Valid types are listed in:
* http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
*
* Several supported types are represented by the constants above.
*/
@property(nonatomic, readonly, nonnull) NSString *name;
/**
* The media type of this codec. Equivalent to MIME top-level type.
*
* Valid values are kRTCMediaStreamTrackKindAudio and
* kRTCMediaStreamTrackKindVideo.
*/
@property(nonatomic, readonly, nonnull) NSString *kind;
/** The codec clock rate expressed in Hertz. */
@property(nonatomic, readonly, nullable) NSNumber *clockRate;
/**
* The number of channels (mono=1, stereo=2).
* Set to null for video codecs.
**/
@property(nonatomic, readonly, nullable) NSNumber *numChannels;
/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpCodecParameters.h"

33
sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
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@ -8,35 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpEncodingParameters : NSObject
/** Controls whether the encoding is currently transmitted. */
@property(nonatomic, assign) BOOL isActive;
/** The maximum bitrate to use for the encoding, or nil if there is no
* limit.
*/
@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
/** The minimum bitrate to use for the encoding, or nil if there is no
* limit.
*
* Not implemented.
*/
@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
/** The SSRC being used by this encoding. */
@property(nonatomic, readonly, nullable) NSNumber *ssrc;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpEncodingParameters.h"

24
sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h
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@ -8,26 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpHeaderExtension : NSObject
/** The URI of the RTP header extension, as defined in RFC5285. */
@property(nonatomic, readonly, copy) NSString *uri;
/** The value put in the RTP packet to identify the header extension. */
@property(nonatomic, readonly) int id;
/** Whether the header extension is encrypted or not. */
@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpHeaderExtension.h"

34
sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
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@ -8,36 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtcpParameters.h>
#import <WebRTC/RTCRtpCodecParameters.h>
#import <WebRTC/RTCRtpEncodingParameters.h>
#import <WebRTC/RTCRtpHeaderExtension.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpParameters : NSObject
/** A unique identifier for the last set of parameters applied. */
@property(nonatomic, copy) NSString *transactionId;
/** Parameters used for RTCP. */
@property(nonatomic, readonly, copy) RTCRtcpParameters *rtcp;
/** An array containing parameters for RTP header extensions. */
@property(nonatomic, readonly, copy) NSArray<RTCRtpHeaderExtension *> *headerExtensions;
/** The currently active encodings in the order of preference. */
@property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;
/** The negotiated set of send codecs in order of preference. */
@property(nonatomic, copy) NSArray<RTCRtpCodecParameters *> *codecs;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpParameters.h"

73
sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
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@ -8,75 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
/** Represents the media type of the RtpReceiver. */
typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
RTCRtpMediaTypeAudio,
RTCRtpMediaTypeVideo,
RTCRtpMediaTypeData,
};
@class RTCRtpReceiver;
RTC_EXPORT
@protocol RTCRtpReceiverDelegate <NSObject>
/** Called when the first RTP packet is received.
*
* Note: Currently if there are multiple RtpReceivers of the same media type,
* they will all call OnFirstPacketReceived at once.
*
* For example, if we create three audio receivers, A/B/C, they will listen to
* the same signal from the underneath network layer. Whenever the first audio packet
* is received, the underneath signal will be fired. All the receivers A/B/C will be
* notified and the callback of the receiver's delegate will be called.
*
* The process is the same for video receivers.
*/
- (void)rtpReceiver:(RTCRtpReceiver *)rtpReceiver
didReceiveFirstPacketForMediaType:(RTCRtpMediaType)mediaType;
@end
RTC_EXPORT
@protocol RTCRtpReceiver <NSObject>
/** A unique identifier for this receiver. */
@property(nonatomic, readonly) NSString *receiverId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
*
* The WebRTC specification only defines RTCRtpParameters in terms of senders,
* but this API also applies them to receivers, similar to ORTC:
* http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
*/
@property(nonatomic, readonly) RTCRtpParameters *parameters;
/** The RTCMediaStreamTrack associated with the receiver.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, readonly, nullable) RTCMediaStreamTrack *track;
/** The delegate for this RtpReceiver. */
@property(nonatomic, weak) id<RTCRtpReceiverDelegate> delegate;
@end
RTC_EXPORT
@interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpReceiver.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
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@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCDtmfSender.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@protocol RTCRtpSender <NSObject>
/** 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.
*/
@property(nonatomic, copy) RTCRtpParameters *parameters;
/** The RTCMediaStreamTrack associated with the sender.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, copy, nullable) RTCMediaStreamTrack *track;
/** The RTCDtmfSender accociated with the RTP sender. */
@property(nonatomic, readonly, nullable) id<RTCDtmfSender> dtmfSender;
@end
RTC_EXPORT
@interface RTCRtpSender : NSObject <RTCRtpSender>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpSender.h"

120
sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
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@ -8,122 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtpReceiver.h>
#import <WebRTC/RTCRtpSender.h>
NS_ASSUME_NONNULL_BEGIN
/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
RTCRtpTransceiverDirectionSendRecv,
RTCRtpTransceiverDirectionSendOnly,
RTCRtpTransceiverDirectionRecvOnly,
RTCRtpTransceiverDirectionInactive,
};
/** Structure for initializing an RTCRtpTransceiver in a call to
* RTCPeerConnection.addTransceiver.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
*/
RTC_EXPORT
@interface RTCRtpTransceiverInit : NSObject
/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The added RTCRtpTransceiver will be added to these streams. */
@property(nonatomic) NSArray<NSString *> *streamIds;
/** TODO(bugs.webrtc.org/7600): Not implemented. */
@property(nonatomic) NSArray<RTCRtpEncodingParameters *> *sendEncodings;
@end
@class RTCRtpTransceiver;
/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
* specification. A transceiver represents a combination of an RTCRtpSender
* and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
* RTCRtpTransceiver is said to be associated with a media description if its
* mid property is non-nil; otherwise, it is said to be disassociated.
* JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
*
* Note that RTCRtpTransceivers are only supported when using
* RTCPeerConnection with Unified Plan SDP.
*
* WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
*/
RTC_EXPORT
@protocol RTCRtpTransceiver <NSObject>
/** Media type of the transceiver. The sender and receiver will also have this
* type.
*/
@property(nonatomic, readonly) RTCRtpMediaType mediaType;
/** The mid attribute is the mid negotiated and present in the local and
* remote descriptions. Before negotiation is complete, the mid value may be
* nil. After rollbacks, the value may change from a non-nil value to nil.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
*/
@property(nonatomic, readonly) NSString *mid;
/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
* media that may be sent with the transceiver's mid. The sender is always
* present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
*/
@property(nonatomic, readonly) RTCRtpSender *sender;
/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
* media that may be received with the transceiver's mid. The receiver is
* always present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
*/
@property(nonatomic, readonly) RTCRtpReceiver *receiver;
/** The isStopped attribute indicates that the sender of this transceiver will
* no longer send, and that the receiver will no longer receive. It is true if
* either stop has been called or if setting the local or remote description
* has caused the RTCRtpTransceiver to be stopped.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
*/
@property(nonatomic, readonly) BOOL isStopped;
/** The direction attribute indicates the preferred direction of this
* transceiver, which will be used in calls to createOffer and createAnswer.
* An update of directionality does not take effect immediately. Instead,
* future calls to createOffer and createAnswer mark the corresponding media
* descriptions as sendrecv, sendonly, recvonly, or inactive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The currentDirection attribute indicates the current direction negotiated
* for this transceiver. If this transceiver has never been represented in an
* offer/answer exchange, or if the transceiver is stopped, the value is not
* present and this method returns NO.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
*/
- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
* this transceiver will no longer send, the receiver will no longer receive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
*/
- (void)stop;
@end
RTC_EXPORT
@interface RTCRtpTransceiver : NSObject <RTCRtpTransceiver>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpTransceiver.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
View File

@ -8,13 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Initialize and clean up the SSL library. Failure is fatal. These call the
* corresponding functions in webrtc/rtc_base/ssladapter.h.
*/
RTC_EXTERN BOOL RTCInitializeSSL(void);
RTC_EXTERN BOOL RTCCleanupSSL(void);
#import "api/peerconnection/RTCSSLAdapter.h"

38
sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
View File

@ -8,40 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the session description type. This exposes the same types that are
* in C++, which doesn't include the rollback type that is in the W3C spec.
*/
typedef NS_ENUM(NSInteger, RTCSdpType) {
RTCSdpTypeOffer,
RTCSdpTypePrAnswer,
RTCSdpTypeAnswer,
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCSessionDescription : NSObject
/** The type of session description. */
@property(nonatomic, readonly) RTCSdpType type;
/** The SDP string representation of this session description. */
@property(nonatomic, readonly) NSString *sdp;
- (instancetype)init NS_UNAVAILABLE;
/** Initialize a session description with a type and SDP string. */
- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+ (NSString *)stringForType:(RTCSdpType)type;
+ (RTCSdpType)typeForString:(NSString *)string;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCSessionDescription.h"

12
sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
View File

@ -8,14 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
RTC_EXTERN void RTCSetupInternalTracer(void);
/** Starts capture to specified file. Must be a valid writable path.
* Returns YES if capture starts.
*/
RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
RTC_EXTERN void RTCStopInternalCapture(void);
RTC_EXTERN void RTCShutdownInternalTracer(void);
#import "api/peerconnection/RTCTracing.h"

22
sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
View File

@ -8,24 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCVideoCapturer;
RTC_EXPORT
@protocol RTCVideoCapturerDelegate <NSObject>
- (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame;
@end
RTC_EXPORT
@interface RTCVideoCapturer : NSObject
@property(nonatomic, weak) id<RTCVideoCapturerDelegate> delegate;
- (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoCapturer.h"

186
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
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@ -8,179 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT extern NSString *const kRTCVideoCodecVp8Name;
RTC_EXPORT extern NSString *const kRTCVideoCodecVp9Name;
RTC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
/** Represents an encoded frame's type. */
typedef NS_ENUM(NSUInteger, RTCFrameType) {
RTCFrameTypeEmptyFrame = 0,
RTCFrameTypeAudioFrameSpeech = 1,
RTCFrameTypeAudioFrameCN = 2,
RTCFrameTypeVideoFrameKey = 3,
RTCFrameTypeVideoFrameDelta = 4,
};
typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
RTCVideoContentTypeUnspecified,
RTCVideoContentTypeScreenshare,
};
/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
RTC_EXPORT
@interface RTCEncodedImage : NSObject
@property(nonatomic, strong) NSData *buffer;
@property(nonatomic, assign) int32_t encodedWidth;
@property(nonatomic, assign) int32_t encodedHeight;
@property(nonatomic, assign) uint32_t timeStamp;
@property(nonatomic, assign) int64_t captureTimeMs;
@property(nonatomic, assign) int64_t ntpTimeMs;
@property(nonatomic, assign) uint8_t flags;
@property(nonatomic, assign) int64_t encodeStartMs;
@property(nonatomic, assign) int64_t encodeFinishMs;
@property(nonatomic, assign) RTCFrameType frameType;
@property(nonatomic, assign) RTCVideoRotation rotation;
@property(nonatomic, assign) BOOL completeFrame;
@property(nonatomic, strong) NSNumber *qp;
@property(nonatomic, assign) RTCVideoContentType contentType;
@end
/** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
RTC_EXPORT
@interface RTCRtpFragmentationHeader : NSObject
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationLength;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationTimeDiff;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationPlType;
@end
/** Implement this protocol to pass codec specific info from the encoder.
* Corresponds to webrtc::CodecSpecificInfo.
*/
RTC_EXPORT
@protocol RTCCodecSpecificInfo <NSObject>
@end
/** Callback block for encoder. */
typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
id<RTCCodecSpecificInfo> info,
RTCRtpFragmentationHeader *header);
/** Callback block for decoder. */
typedef void (^RTCVideoDecoderCallback)(RTCVideoFrame *frame);
typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
RTCVideoCodecModeRealtimeVideo,
RTCVideoCodecModeScreensharing,
};
/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
RTC_EXPORT
@interface RTCVideoCodecInfo : NSObject <NSCoding>
- (instancetype)init NS_UNAVAILABLE;
- (instancetype)initWithName:(NSString *)name;
- (instancetype)initWithName:(NSString *)name
parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
NS_DESIGNATED_INITIALIZER;
- (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info;
@property(nonatomic, readonly) NSString *name;
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
@end
/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
RTC_EXPORT
@interface RTCVideoEncoderSettings : NSObject
@property(nonatomic, strong) NSString *name;
@property(nonatomic, assign) unsigned short width;
@property(nonatomic, assign) unsigned short height;
@property(nonatomic, assign) unsigned int startBitrate; // kilobits/sec.
@property(nonatomic, assign) unsigned int maxBitrate;
@property(nonatomic, assign) unsigned int minBitrate;
@property(nonatomic, assign) unsigned int targetBitrate;
@property(nonatomic, assign) uint32_t maxFramerate;
@property(nonatomic, assign) unsigned int qpMax;
@property(nonatomic, assign) RTCVideoCodecMode mode;
@end
/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
RTC_EXPORT
@interface RTCVideoEncoderQpThresholds : NSObject
- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
@property(nonatomic, readonly) NSInteger low;
@property(nonatomic, readonly) NSInteger high;
@end
/** Protocol for encoder implementations. */
RTC_EXPORT
@protocol RTCVideoEncoder <NSObject>
- (void)setCallback:(RTCVideoEncoderCallback)callback;
- (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores;
- (NSInteger)releaseEncoder;
- (NSInteger)encode:(RTCVideoFrame *)frame
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
frameTypes:(NSArray<NSNumber *> *)frameTypes;
- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
- (NSString *)implementationName;
/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
* keep the QP from the encoded images within the given range. Returning nil from this function
* disables quality scaling. */
- (RTCVideoEncoderQpThresholds *)scalingSettings;
@end
/** Protocol for decoder implementations. */
RTC_EXPORT
@protocol RTCVideoDecoder <NSObject>
- (void)setCallback:(RTCVideoDecoderCallback)callback;
- (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores
DEPRECATED_MSG_ATTRIBUTE("use startDecodeWithNumberOfCores: instead");
- (NSInteger)releaseDecoder;
- (NSInteger)decode:(RTCEncodedImage *)encodedImage
missingFrames:(BOOL)missingFrames
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
renderTimeMs:(int64_t)renderTimeMs;
- (NSString *)implementationName;
// TODO(andersc): Make non-optional when `startDecodeWithSettings:numberOfCores:` is removed.
@optional
- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
@end
NS_ASSUME_NONNULL_END
#import "api/video_codec/RTCVideoCodecConstants.h"
#import "base/RTCCodecSpecificInfo.h"
#import "base/RTCEncodedImage.h"
#import "base/RTCRtpFragmentationHeader.h"
#import "base/RTCVideoCodecInfo.h"
#import "base/RTCVideoDecoder.h"
#import "base/RTCVideoEncoder.h"
#import "base/RTCVideoEncoderQpThresholds.h"
#import "base/RTCVideoEncoderSettings.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"

48
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
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@ -8,47 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
RTC_EXPORT
@protocol RTCVideoEncoderFactory <NSObject>
- (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@end
/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
RTC_EXPORT
@protocol RTCVideoDecoderFactory <NSObject>
- (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@end
#pragma mark - Default factories
/** These codec factories include support for all codecs bundled with WebRTC. If using custom
* codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
*/
RTC_EXPORT
@interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
@property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
+ (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;
@end
RTC_EXPORT
@interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoDecoderFactory.h"
#import "base/RTCVideoEncoderFactory.h"
#import "components/video_codec/RTCDefaultVideoDecoderFactory.h"
#import "components/video_codec/RTCDefaultVideoEncoderFactory.h"

87
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
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@ -8,84 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodecFactory.h>
/** Class for H264 specific config. */
typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
RTCH264PacketizationModeNonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
RTCH264PacketizationModeSingleNalUnit // Mode 0 - only single NALU allowed
};
RTC_EXPORT
@interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
@end
/** H264 Profiles and levels. */
typedef NS_ENUM(NSUInteger, RTCH264Profile) {
RTCH264ProfileConstrainedBaseline,
RTCH264ProfileBaseline,
RTCH264ProfileMain,
RTCH264ProfileConstrainedHigh,
RTCH264ProfileHigh,
};
typedef NS_ENUM(NSUInteger, RTCH264Level) {
RTCH264Level1_b = 0,
RTCH264Level1 = 10,
RTCH264Level1_1 = 11,
RTCH264Level1_2 = 12,
RTCH264Level1_3 = 13,
RTCH264Level2 = 20,
RTCH264Level2_1 = 21,
RTCH264Level2_2 = 22,
RTCH264Level3 = 30,
RTCH264Level3_1 = 31,
RTCH264Level3_2 = 32,
RTCH264Level4 = 40,
RTCH264Level4_1 = 41,
RTCH264Level4_2 = 42,
RTCH264Level5 = 50,
RTCH264Level5_1 = 51,
RTCH264Level5_2 = 52
};
RTC_EXPORT
@interface RTCH264ProfileLevelId : NSObject
@property(nonatomic, readonly) RTCH264Profile profile;
@property(nonatomic, readonly) RTCH264Level level;
@property(nonatomic, readonly) NSString *hexString;
- (instancetype)initWithHexString:(NSString *)hexString;
- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
@end
/** Encoder. */
RTC_EXPORT
@interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
@end
/** Decoder. */
RTC_EXPORT
@interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
@end
/** Encoder factory. */
RTC_EXPORT
@interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
@end
/** Decoder factory. */
RTC_EXPORT
@interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
@end
#import "components/video_codec/RTCCodecSpecificInfoH264.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoDecoderH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderH264.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecInfo.h Normal file
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@ -0,0 +1,11 @@
/*
* Copyright 2017 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 "base/RTCVideoCodecInfo.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoDecoderVP8 : NSObject
/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoDecoder>)vp8Decoder;
@end
#import "api/video_codec/RTCVideoDecoderVP8.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoDecoderVP9 : NSObject
/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoDecoder>)vp9Decoder;
@end
#import "api/video_codec/RTCVideoDecoderVP9.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoEncoderVP8 : NSObject
/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoEncoder>)vp8Encoder;
@end
#import "api/video_codec/RTCVideoEncoderVP8.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoEncoderVP9 : NSObject
/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoEncoder>)vp9Encoder;
@end
#import "api/video_codec/RTCVideoEncoderVP9.h"

76
sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
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@ -8,78 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef NS_ENUM(NSInteger, RTCVideoRotation) {
RTCVideoRotation_0 = 0,
RTCVideoRotation_90 = 90,
RTCVideoRotation_180 = 180,
RTCVideoRotation_270 = 270,
};
@protocol RTCVideoFrameBuffer;
// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
RTC_EXPORT
@interface RTCVideoFrame : NSObject
/** Width without rotation applied. */
@property(nonatomic, readonly) int width;
/** Height without rotation applied. */
@property(nonatomic, readonly) int height;
@property(nonatomic, readonly) RTCVideoRotation rotation;
/** Timestamp in nanoseconds. */
@property(nonatomic, readonly) int64_t timeStampNs;
/** Timestamp 90 kHz. */
@property(nonatomic, assign) int32_t timeStamp;
@property(nonatomic, readonly) id<RTCVideoFrameBuffer> buffer;
- (instancetype)init NS_UNAVAILABLE;
- (instancetype) new NS_UNAVAILABLE;
/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
* Deprecated - initialize with a RTCCVPixelBuffer instead
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
* scaling. Cropping will be applied first on the pixel buffer, followed by
* scaling to the final resolution of scaledWidth x scaledHeight.
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
scaledWidth:(int)scaledWidth
scaledHeight:(int)scaledHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
*/
- (instancetype)initWithBuffer:(id<RTCVideoFrameBuffer>)frameBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs;
/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
* the YUV data on it.
*/
- (RTCVideoFrame *)newI420VideoFrame;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoFrame.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
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@ -8,109 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTCI420Buffer;
// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
RTC_EXPORT
@protocol RTCVideoFrameBuffer <NSObject>
@property(nonatomic, readonly) int width;
@property(nonatomic, readonly) int height;
- (id<RTCI420Buffer>)toI420;
@end
/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
@protocol RTCYUVPlanarBuffer <RTCVideoFrameBuffer>
@property(nonatomic, readonly) int chromaWidth;
@property(nonatomic, readonly) int chromaHeight;
@property(nonatomic, readonly) const uint8_t *dataY;
@property(nonatomic, readonly) const uint8_t *dataU;
@property(nonatomic, readonly) const uint8_t *dataV;
@property(nonatomic, readonly) int strideY;
@property(nonatomic, readonly) int strideU;
@property(nonatomic, readonly) int strideV;
- (instancetype)initWithWidth:(int)width
height:(int)height
dataY:(const uint8_t *)dataY
dataU:(const uint8_t *)dataU
dataV:(const uint8_t *)dataV;
- (instancetype)initWithWidth:(int)width height:(int)height;
- (instancetype)initWithWidth:(int)width
height:(int)height
strideY:(int)strideY
strideU:(int)strideU
strideV:(int)strideV;
@end
/** Extension of the YUV planar data buffer with mutable data access */
@protocol RTCMutableYUVPlanarBuffer <RTCYUVPlanarBuffer>
@property(nonatomic, readonly) uint8_t *mutableDataY;
@property(nonatomic, readonly) uint8_t *mutableDataU;
@property(nonatomic, readonly) uint8_t *mutableDataV;
@end
/** Protocol for RTCYUVPlanarBuffers containing I420 data */
@protocol RTCI420Buffer <RTCYUVPlanarBuffer>
@end
/** Extension of the I420 buffer with mutable data access */
@protocol RTCMutableI420Buffer <RTCI420Buffer, RTCMutableYUVPlanarBuffer>
@end
/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
RTC_EXPORT
@interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
@property(nonatomic, readonly) int cropX;
@property(nonatomic, readonly) int cropY;
@property(nonatomic, readonly) int cropWidth;
@property(nonatomic, readonly) int cropHeight;
+ (NSSet<NSNumber *> *)supportedPixelFormats;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
adaptedWidth:(int)adaptedWidth
adaptedHeight:(int)adaptedHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY;
- (BOOL)requiresCropping;
- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
* bufferSizeForCroppingAndScalingToWidth:height: method.
* If that size is 0, the |tmpBuffer| may be nil.
*/
- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
withTempBuffer:(nullable uint8_t *)tmpBuffer;
@end
/** RTCI420Buffer implements the RTCI420Buffer protocol */
RTC_EXPORT
@interface RTCI420Buffer : NSObject <RTCI420Buffer>
@end
/** Mutable version of RTCI420Buffer */
RTC_EXPORT
@interface RTCMutableI420Buffer : RTCI420Buffer <RTCMutableI420Buffer>
@end
NS_ASSUME_NONNULL_END
#import "api/video_frame_buffer/RTCI420Buffer.h"
#import "api/video_frame_buffer/RTCMutableI420Buffer.h"
#import "base/RTCI420Buffer.h"
#import "base/RTCMutableI420Buffer.h"
#import "base/RTCMutableYUVPlanarBuffer.h"
#import "base/RTCVideoFrameBuffer.h"
#import "base/RTCYUVPlanarBuffer.h"
#import "components/video_frame_buffer/RTCCVPixelBuffer.h"

31
sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
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@ -8,33 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#if TARGET_OS_IPHONE
#import <UIKit/UIKit.h>
#endif
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCVideoFrame;
RTC_EXPORT
@protocol RTCVideoRenderer <NSObject>
/** The size of the frame. */
- (void)setSize:(CGSize)size;
/** The frame to be displayed. */
- (void)renderFrame:(nullable RTCVideoFrame *)frame;
@end
RTC_EXPORT
@protocol RTCVideoViewDelegate
- (void)videoView:(id<RTCVideoRenderer>)videoView didChangeVideoSize:(CGSize)size;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoRenderer.h"

28
sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCVideoSource : RTCMediaSource <RTCVideoCapturerDelegate>
- (instancetype)init NS_UNAVAILABLE;
/**
* Calling this function will cause frames to be scaled down to the
* requested resolution. Also, frames will be cropped to match the
* requested aspect ratio, and frames will be dropped to match the
* requested fps. The requested aspect ratio is orientation agnostic and
* will be adjusted to maintain the input orientation, so it doesn't
* matter if e.g. 1280x720 or 720x1280 is requested.
*/
- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCVideoSource.h"

28
sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTCVideoRenderer;
@class RTCPeerConnectionFactory;
@class RTCVideoSource;
RTC_EXPORT
@interface RTCVideoTrack : RTCMediaStreamTrack
/** The video source for this video track. */
@property(nonatomic, readonly) RTCVideoSource *source;
- (instancetype)init NS_UNAVAILABLE;
/** Register a renderer that will render all frames received on this track. */
- (void)addRenderer:(id<RTCVideoRenderer>)renderer;
/** Deregister a renderer. */
- (void)removeRenderer:(id<RTCVideoRenderer>)renderer;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCVideoTrack.h"

32
sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
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@ -8,34 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/**
* RTCVideoViewShading provides a way for apps to customize the OpenGL(ES) shaders used in
* rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
*/
RTC_EXPORT
@protocol RTCVideoViewShading <NSObject>
/** Callback for I420 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth:(int)width
height:(int)height
rotation:(RTCVideoRotation)rotation
yPlane:(GLuint)yPlane
uPlane:(GLuint)uPlane
vPlane:(GLuint)vPlane;
/** Callback for NV12 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth:(int)width
height:(int)height
rotation:(RTCVideoRotation)rotation
yPlane:(GLuint)yPlane
uvPlane:(GLuint)uvPlane;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCVideoViewShading.h"

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sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
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@ -8,74 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <UIKit/UIKit.h>
typedef NS_ENUM(NSInteger, RTCDeviceType) {
RTCDeviceTypeUnknown,
RTCDeviceTypeIPhone1G,
RTCDeviceTypeIPhone3G,
RTCDeviceTypeIPhone3GS,
RTCDeviceTypeIPhone4,
RTCDeviceTypeIPhone4Verizon,
RTCDeviceTypeIPhone4S,
RTCDeviceTypeIPhone5GSM,
RTCDeviceTypeIPhone5GSM_CDMA,
RTCDeviceTypeIPhone5CGSM,
RTCDeviceTypeIPhone5CGSM_CDMA,
RTCDeviceTypeIPhone5SGSM,
RTCDeviceTypeIPhone5SGSM_CDMA,
RTCDeviceTypeIPhone6Plus,
RTCDeviceTypeIPhone6,
RTCDeviceTypeIPhone6S,
RTCDeviceTypeIPhone6SPlus,
RTCDeviceTypeIPhone7,
RTCDeviceTypeIPhone7Plus,
RTCDeviceTypeIPhoneSE,
RTCDeviceTypeIPhone8,
RTCDeviceTypeIPhone8Plus,
RTCDeviceTypeIPhoneX,
RTCDeviceTypeIPodTouch1G,
RTCDeviceTypeIPodTouch2G,
RTCDeviceTypeIPodTouch3G,
RTCDeviceTypeIPodTouch4G,
RTCDeviceTypeIPodTouch5G,
RTCDeviceTypeIPodTouch6G,
RTCDeviceTypeIPad,
RTCDeviceTypeIPad2Wifi,
RTCDeviceTypeIPad2GSM,
RTCDeviceTypeIPad2CDMA,
RTCDeviceTypeIPad2Wifi2,
RTCDeviceTypeIPadMiniWifi,
RTCDeviceTypeIPadMiniGSM,
RTCDeviceTypeIPadMiniGSM_CDMA,
RTCDeviceTypeIPad3Wifi,
RTCDeviceTypeIPad3GSM_CDMA,
RTCDeviceTypeIPad3GSM,
RTCDeviceTypeIPad4Wifi,
RTCDeviceTypeIPad4GSM,
RTCDeviceTypeIPad4GSM_CDMA,
RTCDeviceTypeIPad5,
RTCDeviceTypeIPad6,
RTCDeviceTypeIPadAirWifi,
RTCDeviceTypeIPadAirCellular,
RTCDeviceTypeIPadAirWifiCellular,
RTCDeviceTypeIPadAir2,
RTCDeviceTypeIPadMini2GWifi,
RTCDeviceTypeIPadMini2GCellular,
RTCDeviceTypeIPadMini2GWifiCellular,
RTCDeviceTypeIPadMini3,
RTCDeviceTypeIPadMini4,
RTCDeviceTypeIPadPro9Inch,
RTCDeviceTypeIPadPro12Inch,
RTCDeviceTypeIPadPro12Inch2,
RTCDeviceTypeIPadPro10Inch,
RTCDeviceTypeSimulatori386,
RTCDeviceTypeSimulatorx86_64,
};
@interface UIDevice (RTCDevice)
+ (RTCDeviceType)deviceType;
+ (BOOL)isIOS11OrLater;
@end
#import "helpers/UIDevice+RTCDevice.h"

6
sdk/objc/Framework/Modules/module.modulemap
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@ -1,6 +0,0 @@
framework module WebRTC {
umbrella header "WebRTC.h"
export *
module * { export * }
}

15
sdk/objc/Framework/Native/api/audio_device_module.h
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@ -8,17 +8,4 @@
* 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_
#import "native/api/audio_device_module.h"

18
sdk/objc/Framework/Native/api/video_decoder_factory.h
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@ -8,20 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
#include <memory>
#import "WebRTC/RTCVideoCodecFactory.h"
#include "api/video_codecs/video_decoder_factory.h"
namespace webrtc {
std::unique_ptr<VideoDecoderFactory> ObjCToNativeVideoDecoderFactory(
id<RTCVideoDecoderFactory> objc_video_decoder_factory);
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
#import "native/api/video_decoder_factory.h"

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sdk/objc/Framework/Native/api/video_encoder_factory.h
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@ -8,20 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
#include <memory>
#import "WebRTC/RTCVideoCodecFactory.h"
#include "api/video_codecs/video_encoder_factory.h"
namespace webrtc {
std::unique_ptr<VideoEncoderFactory> ObjCToNativeVideoEncoderFactory(
id<RTCVideoEncoderFactory> objc_video_encoder_factory);
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
#import "native/api/video_encoder_factory.h"

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sdk/objc/Framework/Native/api/video_frame_buffer.h
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@ -1,5 +1,5 @@
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2015 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
@ -8,22 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
#import "WebRTC/RTCVideoFrameBuffer.h"
#include "common_video/include/video_frame_buffer.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
rtc::scoped_refptr<VideoFrameBuffer> ObjCToNativeVideoFrameBuffer(
id<RTCVideoFrameBuffer> objc_video_frame_buffer);
id<RTCVideoFrameBuffer> NativeToObjCVideoFrameBuffer(
const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
#import "native/api/video_frame_buffer.h"

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sdk/objc/Framework/Native/src/objc_video_decoder_factory.h
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@ -8,31 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
#include "api/video_codecs/video_decoder_factory.h"
#include "media/base/codec.h"
@protocol RTCVideoDecoderFactory;
namespace webrtc {
class ObjCVideoDecoderFactory : public VideoDecoderFactory {
public:
explicit ObjCVideoDecoderFactory(id<RTCVideoDecoderFactory>);
~ObjCVideoDecoderFactory() override;
id<RTCVideoDecoderFactory> wrapped_decoder_factory() const;
std::vector<SdpVideoFormat> GetSupportedFormats() const override;
std::unique_ptr<VideoDecoder> CreateVideoDecoder(
const SdpVideoFormat& format) override;
private:
id<RTCVideoDecoderFactory> decoder_factory_;
};
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
#import "native/src/objc_video_decoder_factory.h"

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sdk/objc/Framework/Native/src/objc_video_encoder_factory.h
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@ -8,33 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
#import <Foundation/Foundation.h>
#include "api/video_codecs/video_encoder_factory.h"
@protocol RTCVideoEncoderFactory;
namespace webrtc {
class ObjCVideoEncoderFactory : public VideoEncoderFactory {
public:
explicit ObjCVideoEncoderFactory(id<RTCVideoEncoderFactory>);
~ObjCVideoEncoderFactory() override;
id<RTCVideoEncoderFactory> wrapped_encoder_factory() const;
std::vector<SdpVideoFormat> GetSupportedFormats() const override;
std::unique_ptr<VideoEncoder> CreateVideoEncoder(
const SdpVideoFormat& format) override;
CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
private:
id<RTCVideoEncoderFactory> encoder_factory_;
};
} // namespace webrtc
#endif // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
#import "native/src/objc_video_encoder_factory.h"

0
sdk/objc/Framework/Info.plist → sdk/objc/Info.plist
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# WebRTC Obj-C SDK
This directory contains the Obj-C SDK for WebRTC. This includes wrappers for the
C++ PeerConnection API and some platform specific components for iOS and macOS.
## Organization
- api/
Wrappers around classes and functions in the C++ API for creating and
configuring peer connections, etc.
- base/
This directory contains some base protocols and classes that are used by both
the platform specific components and the SDK wrappers.
- components/
These are the platform specific components. Contains components for handling
audio, capturing and rendering video, encoding and decoding using the
platform's hardware codec implementation and for representing video frames
in the platform's native format.
- helpers/
These files are not WebRTC specific, but are general helper classes and
utilities for the Cocoa platforms.
- native/
APIs for wrapping the platform specific components and using them with the
C++ API.
- unittests/
This directory contains the tests.

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sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h → sdk/objc/api/RTCVideoRendererAdapter+Private.h
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@ -10,7 +10,7 @@
#import "RTCVideoRendererAdapter.h"
#import "WebRTC/RTCVideoRenderer.h"
#import "base/RTCVideoRenderer.h"
#include "api/mediastreaminterface.h"

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* be found in the AUTHORS file in the root of the source tree.
*/
#import "RTCI420Buffer+Private.h"
#import "RTCVideoRendererAdapter+Private.h"
#import "WebRTC/RTCVideoFrame.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
#import "api/video_frame_buffer/RTCI420Buffer+Private.h"
#import "base/RTCVideoFrame.h"
#import "base/RTCVideoFrameBuffer.h"
#include <memory>
#include "sdk/objc/Framework/Native/api/video_frame.h"
#include "sdk/objc/native/api/video_frame.h"
namespace webrtc {

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/*
* 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 <Foundation/Foundation.h>
#import "RTCLogging.h"
#import "RTCMacros.h"
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and forwards them to a registered block.
// This class is not threadsafe.
RTC_EXPORT
@interface RTCCallbackLogger : NSObject
// The severity level to capture. The default is kRTCLoggingSeverityInfo.
@property(nonatomic, assign) RTCLoggingSeverity severity;
// The callback will be called on the same thread that does the logging, so
// if the logging callback can be slow it may be a good idea to implement
// dispatching to some other queue.
- (void)start:(nullable void (^)(NSString*))callback;
- (void)stop;
@end
NS_ASSUME_NONNULL_END

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCCallbackLogger.h"
#import "RTCCallbackLogger.h"
#include <memory>

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sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h → sdk/objc/api/peerconnection/RTCAudioSource+Private.h
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCAudioSource.h"
#import "RTCAudioSource.h"
#import "RTCMediaSource+Private.h"

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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 <Foundation/Foundation.h>
#import "RTCMacros.h"
#import "RTCMediaSource.h"
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCAudioSource : RTCMediaSource
- (instancetype)init NS_UNAVAILABLE;
// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
// [0, 10].
// Temporary fix to be able to modify volume of remote audio tracks.
// TODO(kthelgason): Property stays here temporarily until a proper volume-api
// is available on the surface exposed by webrtc.
@property(nonatomic, assign) double volume;
@end
NS_ASSUME_NONNULL_END

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* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCAudioTrack.h"
#import "RTCAudioTrack.h"
#include "api/mediastreaminterface.h"

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/*
* Copyright 2015 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 "RTCMacros.h"
#import "RTCMediaStreamTrack.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSource;
RTC_EXPORT
@interface RTCAudioTrack : RTCMediaStreamTrack
- (instancetype)init NS_UNAVAILABLE;
/** The audio source for this audio track. */
@property(nonatomic, readonly) RTCAudioSource *source;
@end
NS_ASSUME_NONNULL_END

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#import "RTCAudioTrack+Private.h"
#import "NSString+StdString.h"
#import "RTCAudioSource+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "helpers/NSString+StdString.h"
#include "rtc_base/checks.h"

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