webrtc_m130/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
Boris Tsirkin 536c19a64d Format /sdk/objc/api folder
There are a lot of changes in /sdk so I'm splitting it

Formatting done via:

git ls-files | grep -E '^sdk\/objc\/api\/.*\.(h|cc|mm)' | xargs clang-format -i

No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: Ieebcd026e77db31f94df2b5dd5cd18ccc4f06674
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373883
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43682}
2025-01-08 08:20:49 -08:00

132 lines
4.4 KiB
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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 "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 {
self = [super init];
if (self) {
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 = std::optional<int>(_maxBitrateBps.intValue);
}
if (_minBitrateBps != nil) {
parameters.min_bitrate_bps = std::optional<int>(_minBitrateBps.intValue);
}
if (_maxFramerate != nil) {
parameters.max_framerate = std::optional<int>(_maxFramerate.intValue);
}
if (_numTemporalLayers != nil) {
parameters.num_temporal_layers =
std::optional<int>(_numTemporalLayers.intValue);
}
if (_scaleResolutionDownBy != nil) {
parameters.scale_resolution_down_by =
std::optional<double>(_scaleResolutionDownBy.doubleValue);
}
if (_ssrc != nil) {
parameters.ssrc = std::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