blob: 615a31dd107c0ea342b74e54accaf4652b1fe185 [file] [log] [blame]
/*
* Copyright 2018 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.
*/
#include "api/rtp_receiver_interface.h"
namespace webrtc {
RtpSource::RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type) {}
RtpSource::RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
uint8_t audio_level)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type),
audio_level_(audio_level) {}
RtpSource::RtpSource(const RtpSource&) = default;
RtpSource& RtpSource::operator=(const RtpSource&) = default;
RtpSource::~RtpSource() = default;
std::vector<std::string> RtpReceiverInterface::stream_ids() const {
return {};
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>>
RtpReceiverInterface::streams() const {
return {};
}
std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
return {};
}
void RtpReceiverInterface::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {}
rtc::scoped_refptr<FrameDecryptorInterface>
RtpReceiverInterface::GetFrameDecryptor() const {
return nullptr;
}
rtc::scoped_refptr<DtlsTransportInterface>
RtpReceiverInterface::dtls_transport() const {
return nullptr;
}
void RtpReceiverInterface::SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) {}
} // namespace webrtc