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/*
* 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 "RTCRtpSource.h"
#import "RTCRtpSource+Private.h"
#import "base/RTCLogging.h"
#import "helpers/NSString+StdString.h"
#include <optional>
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
@implementation RTC_OBJC_TYPE (RTCRtpSource) {
std::optional<webrtc::RtpSource> _nativeRtpSource;
}
- (uint32_t)sourceId {
return _nativeRtpSource.value().source_id();
}
- (CFTimeInterval)timestampUs {
return _nativeRtpSource.value().timestamp().us();
}
- (uint32_t)rtpTimestamp {
return _nativeRtpSource.value().rtp_timestamp();
}
- (RTCRtpSourceType)sourceType {
return [RTC_OBJC_TYPE(RTCRtpSource)
rtpSourceTypeForNativeRtpSourceType:_nativeRtpSource.value()
.source_type()];
}
- (NSNumber *)audioLevel {
std::optional<uint8_t> level = _nativeRtpSource.value().audio_level();
if (!level.has_value()) {
return nil;
}
// Converted according to equation defined here:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource-audiolevel
uint8_t rfcLevel = level.value();
if (rfcLevel > 127u) {
rfcLevel = 127u;
}
if (rfcLevel == 127u) {
return @(0.0);
}
return @(std::pow(10.0, -(double)rfcLevel / 20.0));
}
- (NSString *)description {
return [NSString
stringWithFormat:
@"RTC_OBJC_TYPE(RTCRtpSource) {\n sourceId: %d, sourceType: %@\n}",
self.sourceId,
[RTC_OBJC_TYPE(RTCRtpSource) stringForRtpSourceType:self.sourceType]];
}
- (instancetype)initWithNativeRtpSource:
(const webrtc::RtpSource &)nativeRtpSource {
self = [super init];
if (self) {
_nativeRtpSource = nativeRtpSource;
}
return self;
}
+ (RTCRtpSourceType)rtpSourceTypeForNativeRtpSourceType:
(webrtc::RtpSourceType)nativeRtpSourceType {
switch (nativeRtpSourceType) {
case webrtc::RtpSourceType::SSRC:
return RTCRtpSourceTypeSSRC;
case webrtc::RtpSourceType::CSRC:
return RTCRtpSourceTypeCSRC;
}
}
+ (NSString *)stringForRtpSourceType:(RTCRtpSourceType)mediaType {
switch (mediaType) {
case RTCRtpSourceTypeSSRC:
return @"SSRC";
case RTCRtpSourceTypeCSRC:
return @"CSRC";
}
}
@end
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