From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 7 Apr 2024 21:33:14 +0200
Subject: Adding upstream version 115.7.0esr.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../api/peerconnection/RTCRtpEncodingParameters.mm | 128 +++++++++++++++++++++
 1 file changed, 128 insertions(+)
 create mode 100644 third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm

(limited to 'third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm')

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
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