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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:43:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:43:14 +0000 |
commit | 8dd16259287f58f9273002717ec4d27e97127719 (patch) | |
tree | 3863e62a53829a84037444beab3abd4ed9dfc7d0 /third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm | |
parent | Releasing progress-linux version 126.0.1-1~progress7.99u1. (diff) | |
download | firefox-8dd16259287f58f9273002717ec4d27e97127719.tar.xz firefox-8dd16259287f58f9273002717ec4d27e97127719.zip |
Merging upstream version 127.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm | 116 |
1 files changed, 116 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm new file mode 100644 index 0000000000..2dc1e5dc4b --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm @@ -0,0 +1,116 @@ +/* + * Copyright 2024 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 "RTCRtpCodecCapability+Private.h" + +#import "RTCMediaStreamTrack.h" +#import "helpers/NSString+StdString.h" + +#include "media/base/media_constants.h" +#include "rtc_base/checks.h" + +@implementation RTC_OBJC_TYPE (RTCRtpCodecCapability) + +@synthesize preferredPayloadType = _preferredPayloadType; +@synthesize name = _name; +@synthesize kind = _kind; +@synthesize clockRate = _clockRate; +@synthesize numChannels = _numChannels; +@synthesize parameters = _parameters; +@synthesize mimeType = _mimeType; + +- (instancetype)init { + webrtc::RtpCodecCapability rtpCodecCapability; + return [self initWithNativeRtpCodecCapability:rtpCodecCapability]; +} + +- (instancetype)initWithNativeRtpCodecCapability: + (const webrtc::RtpCodecCapability &)nativeRtpCodecCapability { + if (self = [super init]) { + if (nativeRtpCodecCapability.preferred_payload_type) { + _preferredPayloadType = + [NSNumber numberWithInt:*nativeRtpCodecCapability.preferred_payload_type]; + } + _name = [NSString stringForStdString:nativeRtpCodecCapability.name]; + switch (nativeRtpCodecCapability.kind) { + case cricket::MEDIA_TYPE_AUDIO: + _kind = kRTCMediaStreamTrackKindAudio; + break; + case cricket::MEDIA_TYPE_VIDEO: + _kind = kRTCMediaStreamTrackKindVideo; + break; + case cricket::MEDIA_TYPE_DATA: + RTC_DCHECK_NOTREACHED(); + break; + case cricket::MEDIA_TYPE_UNSUPPORTED: + RTC_DCHECK_NOTREACHED(); + break; + } + if (nativeRtpCodecCapability.clock_rate) { + _clockRate = [NSNumber numberWithInt:*nativeRtpCodecCapability.clock_rate]; + } + if (nativeRtpCodecCapability.num_channels) { + _numChannels = [NSNumber numberWithInt:*nativeRtpCodecCapability.num_channels]; + } + NSMutableDictionary *parameters = [NSMutableDictionary dictionary]; + for (const auto ¶meter : nativeRtpCodecCapability.parameters) { + [parameters setObject:[NSString stringForStdString:parameter.second] + forKey:[NSString stringForStdString:parameter.first]]; + } + _parameters = parameters; + _mimeType = [NSString stringForStdString:nativeRtpCodecCapability.mime_type()]; + } + return self; +} + +- (NSString *)description { + return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpCodecCapability) {\n " + @"preferredPayloadType: %@\n name: %@\n kind: %@\n " + @"clockRate: %@\n numChannels: %@\n parameters: %@\n " + @"mimeType: %@\n}", + _preferredPayloadType, + _name, + _kind, + _clockRate, + _numChannels, + _parameters, + _mimeType]; +} + +- (webrtc::RtpCodecCapability)nativeRtpCodecCapability { + webrtc::RtpCodecCapability rtpCodecCapability; + if (_preferredPayloadType != nil) { + rtpCodecCapability.preferred_payload_type = absl::optional<int>(_preferredPayloadType.intValue); + } + rtpCodecCapability.name = [NSString stdStringForString:_name]; + // NSString pointer comparison is safe here since "kind" is readonly and only + // populated above. + if (_kind == kRTCMediaStreamTrackKindAudio) { + rtpCodecCapability.kind = cricket::MEDIA_TYPE_AUDIO; + } else if (_kind == kRTCMediaStreamTrackKindVideo) { + rtpCodecCapability.kind = cricket::MEDIA_TYPE_VIDEO; + } else { + RTC_DCHECK_NOTREACHED(); + } + if (_clockRate != nil) { + rtpCodecCapability.clock_rate = absl::optional<int>(_clockRate.intValue); + } + if (_numChannels != nil) { + rtpCodecCapability.num_channels = absl::optional<int>(_numChannels.intValue); + } + for (NSString *paramKey in _parameters.allKeys) { + std::string key = [NSString stdStringForString:paramKey]; + std::string value = [NSString stdStringForString:_parameters[paramKey]]; + rtpCodecCapability.parameters[key] = value; + } + return rtpCodecCapability; +} + +@end |