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// Copyright (c) 2017 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
#include <cmath>
#include "base/logging.h"
#include "base/time/time.h"
#include "third_party/webrtc/api/scoped_refptr.h"
namespace blink {
RTCRtpSource::RTCRtpSource(const webrtc::RtpSource& source) : source_(source) {}
RTCRtpSource::~RTCRtpSource() {}
RTCRtpSource::Type RTCRtpSource::SourceType() const {
switch (source_.source_type()) {
case webrtc::RtpSourceType::SSRC:
return RTCRtpSource::Type::kSSRC;
case webrtc::RtpSourceType::CSRC:
return RTCRtpSource::Type::kCSRC;
default:
NOTREACHED();
return RTCRtpSource::Type::kSSRC;
}
}
base::TimeTicks RTCRtpSource::Timestamp() const {
return base::TimeTicks() +
base::TimeDelta::FromMilliseconds(source_.timestamp_ms());
}
uint32_t RTCRtpSource::Source() const {
return source_.source_id();
}
base::Optional<double> RTCRtpSource::AudioLevel() const {
if (!source_.audio_level())
return base::nullopt;
// Converted according to equation defined here:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource-audiolevel
uint8_t rfc_level = *source_.audio_level();
if (rfc_level > 127u)
rfc_level = 127u;
if (rfc_level == 127u)
return 0.0;
return std::pow(10.0, -(double)rfc_level / 20.0);
}
uint32_t RTCRtpSource::RtpTimestamp() const {
return source_.rtp_timestamp();
}
} // namespace blink
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