diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
---|---|---|
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/RTCRtpEncodingParameters.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/RTCRtpEncodingParameters.mm | 128 |
1 files changed, 128 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm new file mode 100644 index 0000000000..d6087dafb0 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm @@ -0,0 +1,128 @@ +/* + * 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 "RTCRtpEncodingParameters+Private.h" + +#import "helpers/NSString+StdString.h" + +@implementation RTC_OBJC_TYPE (RTCRtpEncodingParameters) + +@synthesize rid = _rid; +@synthesize isActive = _isActive; +@synthesize maxBitrateBps = _maxBitrateBps; +@synthesize minBitrateBps = _minBitrateBps; +@synthesize maxFramerate = _maxFramerate; +@synthesize numTemporalLayers = _numTemporalLayers; +@synthesize scaleResolutionDownBy = _scaleResolutionDownBy; +@synthesize ssrc = _ssrc; +@synthesize bitratePriority = _bitratePriority; +@synthesize networkPriority = _networkPriority; +@synthesize adaptiveAudioPacketTime = _adaptiveAudioPacketTime; + +- (instancetype)init { + webrtc::RtpEncodingParameters nativeParameters; + return [self initWithNativeParameters:nativeParameters]; +} + +- (instancetype)initWithNativeParameters: + (const webrtc::RtpEncodingParameters &)nativeParameters { + if (self = [super init]) { + if (!nativeParameters.rid.empty()) { + _rid = [NSString stringForStdString:nativeParameters.rid]; + } + _isActive = nativeParameters.active; + if (nativeParameters.max_bitrate_bps) { + _maxBitrateBps = + [NSNumber numberWithInt:*nativeParameters.max_bitrate_bps]; + } + if (nativeParameters.min_bitrate_bps) { + _minBitrateBps = + [NSNumber numberWithInt:*nativeParameters.min_bitrate_bps]; + } + if (nativeParameters.max_framerate) { + _maxFramerate = [NSNumber numberWithInt:*nativeParameters.max_framerate]; + } + if (nativeParameters.num_temporal_layers) { + _numTemporalLayers = [NSNumber numberWithInt:*nativeParameters.num_temporal_layers]; + } + if (nativeParameters.scale_resolution_down_by) { + _scaleResolutionDownBy = + [NSNumber numberWithDouble:*nativeParameters.scale_resolution_down_by]; + } + if (nativeParameters.ssrc) { + _ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc]; + } + _bitratePriority = nativeParameters.bitrate_priority; + _networkPriority = [RTC_OBJC_TYPE(RTCRtpEncodingParameters) + priorityFromNativePriority:nativeParameters.network_priority]; + _adaptiveAudioPacketTime = nativeParameters.adaptive_ptime; + } + return self; +} + +- (webrtc::RtpEncodingParameters)nativeParameters { + webrtc::RtpEncodingParameters parameters; + if (_rid != nil) { + parameters.rid = [NSString stdStringForString:_rid]; + } + parameters.active = _isActive; + if (_maxBitrateBps != nil) { + parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue); + } + if (_minBitrateBps != nil) { + parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue); + } + if (_maxFramerate != nil) { + parameters.max_framerate = absl::optional<int>(_maxFramerate.intValue); + } + if (_numTemporalLayers != nil) { + parameters.num_temporal_layers = absl::optional<int>(_numTemporalLayers.intValue); + } + if (_scaleResolutionDownBy != nil) { + parameters.scale_resolution_down_by = + absl::optional<double>(_scaleResolutionDownBy.doubleValue); + } + if (_ssrc != nil) { + parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue); + } + parameters.bitrate_priority = _bitratePriority; + parameters.network_priority = + [RTC_OBJC_TYPE(RTCRtpEncodingParameters) nativePriorityFromPriority:_networkPriority]; + parameters.adaptive_ptime = _adaptiveAudioPacketTime; + return parameters; +} + ++ (webrtc::Priority)nativePriorityFromPriority:(RTCPriority)networkPriority { + switch (networkPriority) { + case RTCPriorityVeryLow: + return webrtc::Priority::kVeryLow; + case RTCPriorityLow: + return webrtc::Priority::kLow; + case RTCPriorityMedium: + return webrtc::Priority::kMedium; + case RTCPriorityHigh: + return webrtc::Priority::kHigh; + } +} + ++ (RTCPriority)priorityFromNativePriority:(webrtc::Priority)nativePriority { + switch (nativePriority) { + case webrtc::Priority::kVeryLow: + return RTCPriorityVeryLow; + case webrtc::Priority::kLow: + return RTCPriorityLow; + case webrtc::Priority::kMedium: + return RTCPriorityMedium; + case webrtc::Priority::kHigh: + return RTCPriorityHigh; + } +} + +@end |