webrtc_m130/webrtc/voice_engine/channel_proxy.cc
ivoc 9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00

214 lines
7.1 KiB
C++

/*
* 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/audio_sink.h"
#include "webrtc/base/checks.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());
}
ChannelProxy::~ChannelProxy() {}
void ChannelProxy::SetRTCPStatus(bool enable) {
channel()->SetRTCPStatus(enable);
}
void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetLocalSSRC(ssrc);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
RTC_DCHECK(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(thread_checker_.CalledOnValidThread());
channel()->SetNACKStatus(enable, max_packets);
}
void ChannelProxy::SetSendAbsoluteSenderTimeStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAbsoluteSenderTimeStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableReceiveTransportSequenceNumber(id);
}
void ChannelProxy::RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->RegisterSenderCongestionControlObjects(
rtp_packet_sender, transport_feedback_observer, packet_router);
}
void ChannelProxy::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->RegisterReceiverCongestionControlObjects(packet_router);
}
void ChannelProxy::ResetCongestionControlObjects() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->ResetCongestionControlObjects();
}
CallStatistics ChannelProxy::GetRTCPStatistics() const {
RTC_DCHECK(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(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(thread_checker_.CalledOnValidThread());
NetworkStatistics stats = {0};
int error = channel()->GetNetworkStatistics(stats);
RTC_DCHECK_EQ(0, error);
return stats;
}
AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
AudioDecodingCallStats stats;
channel()->GetDecodingCallStatistics(&stats);
return stats;
}
int32_t ChannelProxy::GetSpeechOutputLevelFullRange() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
uint32_t level = 0;
int error = channel()->GetSpeechOutputLevelFullRange(level);
RTC_DCHECK_EQ(0, error);
return static_cast<int32_t>(level);
}
uint32_t ChannelProxy::GetDelayEstimate() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel()->GetDelayEstimate();
}
bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
}
bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
}
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
}
void ChannelProxy::SetInputMute(bool muted) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetInputMute(muted);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::RegisterExternalTransport(Transport* transport) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->RegisterExternalTransport(transport);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::DeRegisterExternalTransport() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->DeRegisterExternalTransport();
}
bool ChannelProxy::ReceivedRTPPacket(const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
// May be called on either worker thread or network thread.
return channel()->ReceivedRTPPacket(packet, length, packet_time) == 0;
}
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(thread_checker_.CalledOnValidThread());
return channel()->GetAudioDecoderFactory();
}
void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetChannelOutputVolumeScaling(scaling);
RTC_DCHECK_EQ(0, error);
}
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
}
} // namespace voe
} // namespace webrtc