mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

This change injects the FrameEncryptorInterface and the FrameDecryptorInterface into the RtpSenderInterface and RtpReceiverInterface respectively. This is the second stage of the injection. In a follow up CL non owning pointers to these values will be passed down into the media channel. This change also updates the corresponding mock files. Bug: webrtc:9681 Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d Reviewed-on: https://webrtc-review.googlesource.com/96625 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24489}
56 lines
1.6 KiB
C++
56 lines
1.6 KiB
C++
/*
|
|
* 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/rtpreceiverinterface.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;
|
|
}
|
|
|
|
} // namespace webrtc
|