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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm | 121 |
1 files changed, 121 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm new file mode 100644 index 0000000000..2baf0ecd80 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm @@ -0,0 +1,121 @@ +/* + * 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 "RTCRtpParameters+Private.h" + +#import "RTCRtcpParameters+Private.h" +#import "RTCRtpCodecParameters+Private.h" +#import "RTCRtpEncodingParameters+Private.h" +#import "RTCRtpHeaderExtension+Private.h" +#import "helpers/NSString+StdString.h" + +@implementation RTC_OBJC_TYPE (RTCRtpParameters) + +@synthesize transactionId = _transactionId; +@synthesize rtcp = _rtcp; +@synthesize headerExtensions = _headerExtensions; +@synthesize encodings = _encodings; +@synthesize codecs = _codecs; +@synthesize degradationPreference = _degradationPreference; + +- (instancetype)init { + webrtc::RtpParameters nativeParameters; + return [self initWithNativeParameters:nativeParameters]; +} + +- (instancetype)initWithNativeParameters: + (const webrtc::RtpParameters &)nativeParameters { + if (self = [super init]) { + _transactionId = [NSString stringForStdString:nativeParameters.transaction_id]; + _rtcp = + [[RTC_OBJC_TYPE(RTCRtcpParameters) alloc] initWithNativeParameters:nativeParameters.rtcp]; + + NSMutableArray *headerExtensions = [[NSMutableArray alloc] init]; + for (const auto &headerExtension : nativeParameters.header_extensions) { + [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtension) alloc] + initWithNativeParameters:headerExtension]]; + } + _headerExtensions = headerExtensions; + + NSMutableArray *encodings = [[NSMutableArray alloc] init]; + for (const auto &encoding : nativeParameters.encodings) { + [encodings addObject:[[RTC_OBJC_TYPE(RTCRtpEncodingParameters) alloc] + initWithNativeParameters:encoding]]; + } + _encodings = encodings; + + NSMutableArray *codecs = [[NSMutableArray alloc] init]; + for (const auto &codec : nativeParameters.codecs) { + [codecs + addObject:[[RTC_OBJC_TYPE(RTCRtpCodecParameters) alloc] initWithNativeParameters:codec]]; + } + _codecs = codecs; + + _degradationPreference = [RTC_OBJC_TYPE(RTCRtpParameters) + degradationPreferenceFromNativeDegradationPreference:nativeParameters + .degradation_preference]; + } + return self; +} + +- (webrtc::RtpParameters)nativeParameters { + webrtc::RtpParameters parameters; + parameters.transaction_id = [NSString stdStringForString:_transactionId]; + parameters.rtcp = [_rtcp nativeParameters]; + for (RTC_OBJC_TYPE(RTCRtpHeaderExtension) * headerExtension in _headerExtensions) { + parameters.header_extensions.push_back(headerExtension.nativeParameters); + } + for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * encoding in _encodings) { + parameters.encodings.push_back(encoding.nativeParameters); + } + for (RTC_OBJC_TYPE(RTCRtpCodecParameters) * codec in _codecs) { + parameters.codecs.push_back(codec.nativeParameters); + } + if (_degradationPreference) { + parameters.degradation_preference = [RTC_OBJC_TYPE(RTCRtpParameters) + nativeDegradationPreferenceFromDegradationPreference:(RTCDegradationPreference) + _degradationPreference.intValue]; + } + return parameters; +} + ++ (webrtc::DegradationPreference)nativeDegradationPreferenceFromDegradationPreference: + (RTCDegradationPreference)degradationPreference { + switch (degradationPreference) { + case RTCDegradationPreferenceDisabled: + return webrtc::DegradationPreference::DISABLED; + case RTCDegradationPreferenceMaintainFramerate: + return webrtc::DegradationPreference::MAINTAIN_FRAMERATE; + case RTCDegradationPreferenceMaintainResolution: + return webrtc::DegradationPreference::MAINTAIN_RESOLUTION; + case RTCDegradationPreferenceBalanced: + return webrtc::DegradationPreference::BALANCED; + } +} + ++ (NSNumber *)degradationPreferenceFromNativeDegradationPreference: + (absl::optional<webrtc::DegradationPreference>)nativeDegradationPreference { + if (!nativeDegradationPreference.has_value()) { + return nil; + } + + switch (*nativeDegradationPreference) { + case webrtc::DegradationPreference::DISABLED: + return @(RTCDegradationPreferenceDisabled); + case webrtc::DegradationPreference::MAINTAIN_FRAMERATE: + return @(RTCDegradationPreferenceMaintainFramerate); + case webrtc::DegradationPreference::MAINTAIN_RESOLUTION: + return @(RTCDegradationPreferenceMaintainResolution); + case webrtc::DegradationPreference::BALANCED: + return @(RTCDegradationPreferenceBalanced); + } +} + +@end |