webrtc_m130/webrtc/voice_engine/channel_proxy.cc

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/*
* Copyright (c) 2015 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/voice_engine/channel_proxy.h"
#include <utility>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/call/rtp_transport_controller_send_interface.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/safe_minmax.h"
#include "webrtc/voice_engine/channel.h"
namespace webrtc {
namespace voe {
ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
channel_owner_(channel_owner) {
RTC_CHECK(channel_owner_.channel());
module_process_thread_checker_.DetachFromThread();
}
ChannelProxy::~ChannelProxy() {}
bool ChannelProxy::SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetEncoder(payload_type, std::move(encoder));
}
void ChannelProxy::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->ModifyEncoder(modifier);
}
void ChannelProxy::SetRTCPStatus(bool enable) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRTCPStatus(enable);
}
void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetLocalSSRC(ssrc);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
std::string c_name_limited = c_name.substr(0, 255);
int error = channel()->SetRTCP_CNAME(c_name_limited.c_str());
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetNACKStatus(bool enable, int max_packets) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetNACKStatus(enable, max_packets);
}
void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->EnableReceiveTransportSequenceNumber(id);
}
void ChannelProxy::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->RegisterSenderCongestionControlObjects(transport,
bandwidth_observer);
}
void ChannelProxy::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->RegisterReceiverCongestionControlObjects(packet_router);
}
void ChannelProxy::ResetSenderCongestionControlObjects() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->ResetSenderCongestionControlObjects();
}
void ChannelProxy::ResetReceiverCongestionControlObjects() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->ResetReceiverCongestionControlObjects();
}
CallStatistics ChannelProxy::GetRTCPStatistics() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
CallStatistics stats = {0};
int error = channel()->GetRTPStatistics(stats);
RTC_DCHECK_EQ(0, error);
return stats;
}
std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
std::vector<webrtc::ReportBlock> blocks;
int error = channel()->GetRemoteRTCPReportBlocks(&blocks);
RTC_DCHECK_EQ(0, error);
return blocks;
}
NetworkStatistics ChannelProxy::GetNetworkStatistics() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
NetworkStatistics stats = {0};
int error = channel()->GetNetworkStatistics(stats);
RTC_DCHECK_EQ(0, error);
return stats;
}
AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
AudioDecodingCallStats stats;
channel()->GetDecodingCallStatistics(&stats);
return stats;
}
int ChannelProxy::GetSpeechOutputLevel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetSpeechOutputLevel();
}
int ChannelProxy::GetSpeechOutputLevelFullRange() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetSpeechOutputLevelFullRange();
}
double ChannelProxy::GetTotalOutputEnergy() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetTotalOutputEnergy();
}
double ChannelProxy::GetTotalOutputDuration() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetTotalOutputDuration();
}
uint32_t ChannelProxy::GetDelayEstimate() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
module_process_thread_checker_.CalledOnValidThread());
return channel()->GetDelayEstimate();
}
bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetSendTelephoneEventPayloadType(payload_type,
payload_frequency) == 0;
}
bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
}
void ChannelProxy::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) {
// This method can be called on the worker thread, module process thread
// or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
// TODO(solenberg): Figure out a good way to check this or enforce calling
// rules.
// RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
// module_process_thread_checker_.CalledOnValidThread());
channel()->SetBitRate(bitrate_bps, probing_interval_ms);
}
void ChannelProxy::SetRecPayloadType(int payload_type,
const SdpAudioFormat& format) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const int result = channel()->SetRecPayloadType(payload_type, format);
RTC_DCHECK_EQ(0, result);
}
void ChannelProxy::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetReceiveCodecs(codecs);
}
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
}
void ChannelProxy::SetInputMute(bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetInputMute(muted);
}
void ChannelProxy::RegisterExternalTransport(Transport* transport) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->RegisterExternalTransport(transport);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::DeRegisterExternalTransport() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->DeRegisterExternalTransport();
}
void ChannelProxy::OnRtpPacket(const RtpPacketReceived& packet) {
// May be called on either worker thread or network thread.
channel()->OnRtpPacket(packet);
}
bool ChannelProxy::ReceivedRTCPPacket(const uint8_t* packet, size_t length) {
// May be called on either worker thread or network thread.
return channel()->ReceivedRTCPPacket(packet, length) == 0;
}
const rtc::scoped_refptr<AudioDecoderFactory>&
ChannelProxy::GetAudioDecoderFactory() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetAudioDecoderFactory();
}
void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetChannelOutputVolumeScaling(scaling);
}
void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRtcEventLog(event_log);
}
AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
return channel()->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
}
int ChannelProxy::NeededFrequency() const {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
return static_cast<int>(channel()->NeededFrequency(-1));
}
void ChannelProxy::SetTransportOverhead(int transport_overhead_per_packet) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetTransportOverhead(transport_overhead_per_packet);
}
void ChannelProxy::AssociateSendChannel(
const ChannelProxy& send_channel_proxy) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->set_associate_send_channel(send_channel_proxy.channel_owner_);
}
void ChannelProxy::DisassociateSendChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->set_associate_send_channel(ChannelOwner(nullptr));
}
void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver) const {
RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
RTC_DCHECK(rtp_rtcp);
RTC_DCHECK(rtp_receiver);
int error = channel()->GetRtpRtcp(rtp_rtcp, rtp_receiver);
RTC_DCHECK_EQ(0, error);
}
uint32_t ChannelProxy::GetPlayoutTimestamp() const {
RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
unsigned int timestamp = 0;
int error = channel()->GetPlayoutTimestamp(timestamp);
RTC_DCHECK(!error || timestamp == 0);
return timestamp;
}
void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
// Limit to range accepted by both VoE and ACM, so we're at least getting as
// close as possible, instead of failing.
delay_ms = rtc::SafeClamp(delay_ms, 0, 10000);
int error = channel()->SetMinimumPlayoutDelay(delay_ms);
if (0 != error) {
LOG(LS_WARNING) << "Error setting minimum playout delay.";
}
}
void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRtcpRttStats(rtcp_rtt_stats);
}
bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetRecCodec(*codec_inst) == 0;
}
void ChannelProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
// TODO(eladalon): This fails in UT; fix and uncomment.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7405
// RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->OnTwccBasedUplinkPacketLossRate(packet_loss_rate);
}
void ChannelProxy::OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate) {
// TODO(eladalon): This fails in UT; fix and uncomment.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7405
// RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
}
void ChannelProxy::RegisterLegacyReceiveCodecs() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->RegisterLegacyReceiveCodecs();
}
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
std::vector<RtpSource> ChannelProxy::GetSources() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetSources();
}
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
}
} // namespace voe
} // namespace webrtc