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e1301a8b3a
This reverts commit 806e06d136.
Reason for revert: Breaks WebRTC roll to Chromium. https://chromium-review.googlesource.com/c/chromium/src/+/1375538
02:52:35.346 7748 [6936:11248:1213/025234.206:ERROR:mediaengine.cc(80)] Attempted to set RtpParameters with modified codecPayloadType (INVALID_MODIFICATION)
Original change's description:
> Implement read-only codecPayloadType in RtpParameters
>
> Bug: webrtc:7580
> Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
> Reviewed-on: https://webrtc-review.googlesource.com/c/113944
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25993}
TBR=steveanton@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,shampson@webrtc.org,orphis@webrtc.org
Change-Id: I157f9a79ae7133395431891e15e2c053559d359b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7580
Reviewed-on: https://webrtc-review.googlesource.com/c/114300
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26000}
72 lines
2.3 KiB
Text
72 lines
2.3 KiB
Text
/*
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* Copyright 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import "RTCRtpEncodingParameters+Private.h"
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@implementation RTCRtpEncodingParameters
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@synthesize isActive = _isActive;
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@synthesize maxBitrateBps = _maxBitrateBps;
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@synthesize minBitrateBps = _minBitrateBps;
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@synthesize maxFramerate = _maxFramerate;
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@synthesize numTemporalLayers = _numTemporalLayers;
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@synthesize ssrc = _ssrc;
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- (instancetype)init {
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return [super init];
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}
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- (instancetype)initWithNativeParameters:
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(const webrtc::RtpEncodingParameters &)nativeParameters {
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if (self = [self init]) {
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_isActive = nativeParameters.active;
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if (nativeParameters.max_bitrate_bps) {
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_maxBitrateBps =
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[NSNumber numberWithInt:*nativeParameters.max_bitrate_bps];
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}
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if (nativeParameters.min_bitrate_bps) {
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_minBitrateBps =
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[NSNumber numberWithInt:*nativeParameters.min_bitrate_bps];
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}
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if (nativeParameters.max_framerate) {
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_maxFramerate = [NSNumber numberWithInt:*nativeParameters.max_framerate];
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}
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if (nativeParameters.num_temporal_layers) {
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_numTemporalLayers = [NSNumber numberWithInt:*nativeParameters.num_temporal_layers];
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}
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if (nativeParameters.ssrc) {
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_ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
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}
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}
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return self;
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}
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- (webrtc::RtpEncodingParameters)nativeParameters {
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webrtc::RtpEncodingParameters parameters;
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parameters.active = _isActive;
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if (_maxBitrateBps != nil) {
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parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
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}
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if (_minBitrateBps != nil) {
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parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
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}
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if (_maxFramerate != nil) {
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parameters.max_framerate = absl::optional<int>(_maxFramerate.intValue);
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}
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if (_numTemporalLayers != nil) {
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parameters.num_temporal_layers = absl::optional<int>(_numTemporalLayers.intValue);
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}
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if (_ssrc != nil) {
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parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
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}
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return parameters;
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}
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@end
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