blob: 94fa83b60e6699ae3c0cb1373a4c1143af43371e [file] [log] [blame]
* Copyright (c) 2017 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 "webrtc/call/rtp_stream_receiver_controller.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/ptr_util.h"
namespace webrtc {
RtpStreamReceiverController* controller,
uint32_t ssrc,
RtpPacketSinkInterface* sink)
: controller_(controller), sink_(sink) {
const bool sink_added = controller_->AddSink(ssrc, sink_);
if (!sink_added) {
LOG(LS_ERROR) << "RtpStreamReceiverController::Receiver::Receiver: Sink "
<< "could not be added for SSRC=" << ssrc << ".";
RtpStreamReceiverController::Receiver::~Receiver() {
// Don't require return value > 0, since for RTX we currently may
// have multiple Receiver objects with the same sink.
// TODO(nisse): Consider adding a DCHECK when RtxReceiveStream is wired up.
RtpStreamReceiverController::RtpStreamReceiverController() = default;
RtpStreamReceiverController::~RtpStreamReceiverController() = default;
uint32_t ssrc,
RtpPacketSinkInterface* sink) {
return rtc::MakeUnique<Receiver>(this, ssrc, sink);
bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
rtc::CritScope cs(&lock_);
return demuxer_.OnRtpPacket(packet);
bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
RtpPacketSinkInterface* sink) {
rtc::CritScope cs(&lock_);
return demuxer_.AddSink(ssrc, sink);
size_t RtpStreamReceiverController::RemoveSink(
const RtpPacketSinkInterface* sink) {
rtc::CritScope cs(&lock_);
return demuxer_.RemoveSink(sink);
} // namespace webrtc