This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
210 lines
7.3 KiB
C++
210 lines
7.3 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/channel_send_proxy.h"
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#include <utility>
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#include "api/crypto/frameencryptorinterface.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace voe {
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ChannelSendProxy::ChannelSendProxy() {}
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ChannelSendProxy::ChannelSendProxy(std::unique_ptr<ChannelSend> channel)
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: channel_(std::move(channel)) {
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RTC_DCHECK(channel_);
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module_process_thread_checker_.DetachFromThread();
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}
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ChannelSendProxy::~ChannelSendProxy() {}
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void ChannelSendProxy::SetLocalSSRC(uint32_t ssrc) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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int error = channel_->SetLocalSSRC(ssrc);
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RTC_DCHECK_EQ(0, error);
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}
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void ChannelSendProxy::SetNACKStatus(bool enable, int max_packets) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetNACKStatus(enable, max_packets);
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}
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CallSendStatistics ChannelSendProxy::GetRTCPStatistics() const {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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CallSendStatistics stats = {0};
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int error = channel_->GetRTPStatistics(stats);
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RTC_DCHECK_EQ(0, error);
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return stats;
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}
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void ChannelSendProxy::RegisterTransport(Transport* transport) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->RegisterTransport(transport);
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}
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bool ChannelSendProxy::ReceivedRTCPPacket(const uint8_t* packet,
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size_t length) {
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// May be called on either worker thread or network thread.
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return channel_->ReceivedRTCPPacket(packet, length) == 0;
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}
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bool ChannelSendProxy::SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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return channel_->SetEncoder(payload_type, std::move(encoder));
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}
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void ChannelSendProxy::ModifyEncoder(
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rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->ModifyEncoder(modifier);
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}
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void ChannelSendProxy::SetRTCPStatus(bool enable) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetRTCPStatus(enable);
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}
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void ChannelSendProxy::SetMid(const std::string& mid, int extension_id) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetMid(mid, extension_id);
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}
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void ChannelSendProxy::SetRTCP_CNAME(const std::string& c_name) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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// Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
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std::string c_name_limited = c_name.substr(0, 255);
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int error = channel_->SetRTCP_CNAME(c_name_limited.c_str());
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RTC_DCHECK_EQ(0, error);
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}
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void ChannelSendProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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int error = channel_->SetSendAudioLevelIndicationStatus(enable, id);
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RTC_DCHECK_EQ(0, error);
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}
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void ChannelSendProxy::EnableSendTransportSequenceNumber(int id) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->EnableSendTransportSequenceNumber(id);
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}
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void ChannelSendProxy::RegisterSenderCongestionControlObjects(
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RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->RegisterSenderCongestionControlObjects(transport,
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bandwidth_observer);
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}
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void ChannelSendProxy::ResetSenderCongestionControlObjects() {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->ResetSenderCongestionControlObjects();
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}
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std::vector<ReportBlock> ChannelSendProxy::GetRemoteRTCPReportBlocks() const {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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std::vector<webrtc::ReportBlock> blocks;
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int error = channel_->GetRemoteRTCPReportBlocks(&blocks);
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RTC_DCHECK_EQ(0, error);
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return blocks;
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}
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ANAStats ChannelSendProxy::GetANAStatistics() const {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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return channel_->GetANAStatistics();
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}
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bool ChannelSendProxy::SetSendTelephoneEventPayloadType(int payload_type,
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int payload_frequency) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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return channel_->SetSendTelephoneEventPayloadType(payload_type,
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payload_frequency) == 0;
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}
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bool ChannelSendProxy::SendTelephoneEventOutband(int event, int duration_ms) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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return channel_->SendTelephoneEventOutband(event, duration_ms) == 0;
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}
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void ChannelSendProxy::SetBitrate(int bitrate_bps,
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int64_t probing_interval_ms) {
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// This method can be called on the worker thread, module process thread
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// or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
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// TODO(solenberg): Figure out a good way to check this or enforce calling
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// rules.
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// RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
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// module_process_thread_checker_.CalledOnValidThread());
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channel_->SetBitRate(bitrate_bps, probing_interval_ms);
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}
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int ChannelSendProxy::GetBitrate() const {
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return channel_->GetBitRate();
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}
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void ChannelSendProxy::SetInputMute(bool muted) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetInputMute(muted);
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}
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void ChannelSendProxy::ProcessAndEncodeAudio(
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std::unique_ptr<AudioFrame> audio_frame) {
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RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
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return channel_->ProcessAndEncodeAudio(std::move(audio_frame));
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}
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void ChannelSendProxy::SetTransportOverhead(int transport_overhead_per_packet) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetTransportOverhead(transport_overhead_per_packet);
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}
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RtpRtcp* ChannelSendProxy::GetRtpRtcp() const {
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RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
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return channel_->GetRtpRtcp();
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}
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void ChannelSendProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->OnTwccBasedUplinkPacketLossRate(packet_loss_rate);
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}
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void ChannelSendProxy::OnRecoverableUplinkPacketLossRate(
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float recoverable_packet_loss_rate) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
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}
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void ChannelSendProxy::StartSend() {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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int error = channel_->StartSend();
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RTC_DCHECK_EQ(0, error);
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}
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void ChannelSendProxy::StopSend() {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->StopSend();
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}
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ChannelSend* ChannelSendProxy::GetChannel() const {
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return channel_.get();
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}
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void ChannelSendProxy::SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetFrameEncryptor(frame_encryptor);
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}
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} // namespace voe
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} // namespace webrtc
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