Files
platform-external-webrtc/webrtc/voice_engine/channel_proxy.cc
Stefan Holmer b86d4e4a8d Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.

BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1479023002 .

Cr-Commit-Position: refs/heads/master@{#10909}
2015-12-07 09:26:32 +00:00

144 lines
4.7 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 "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());
}
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::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::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
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::SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->SetCongestionControlObjects(
rtp_packet_sender, transport_feedback_observer, packet_router);
}
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(uint8_t event,
uint32_t duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return
channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
}
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
}
} // namespace voe
} // namespace webrtc