admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Wire up audio packet loss to BWE.

Browse Source 
BUG=webtrc:5079

Review-Url: https://codereview.webrtc.org/2658233002
Cr-Commit-Position: refs/heads/master@{#16474}
This commit is contained in:
stefan 2017-02-07 07:14:08 -08:00 committed by Commit bot
parent 2bc6864278
commit 7de8d64f89
10 changed files with 108 additions and 40 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/audio/BUILD.gn
View File

@ -37,6 +37,7 @@ rtc_static_library("audio") {
"../common_audio",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller:bitrate_controller",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",

10
webrtc/audio/audio_send_stream.cc
View File

@ -19,6 +19,7 @@
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -62,9 +63,6 @@ AudioSendStream::AudioSendStream(
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
channel_proxy_->RegisterSenderCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(), packet_router);
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
@ -81,10 +79,16 @@ AudioSendStream::AudioSendStream(
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
congestion_controller->EnablePeriodicAlrProbing(true);
bandwidth_observer_.reset(congestion_controller->GetBitrateController()
->CreateRtcpBandwidthObserver());
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";
}
}
channel_proxy_->RegisterSenderCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(), packet_router,
bandwidth_observer_.get());
if (!SetupSendCodec()) {
LOG(LS_ERROR) << "Failed to set up send codec state.";
}

2
webrtc/audio/audio_send_stream.h
View File

@ -23,6 +23,7 @@ namespace webrtc {
class CongestionController;
class VoiceEngine;
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class PacketRouter;
@ -77,6 +78,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
BitrateAllocator* const bitrate_allocator_;
CongestionController* const congestion_controller_;
std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};

78
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -32,6 +32,8 @@ namespace test {
namespace {
using testing::_;
using testing::Eq;
using testing::Ne;
using testing::Return;
const int kChannelId = 1;
@ -62,7 +64,7 @@ class MockLimitObserver : public BitrateAllocator::LimitObserver {
};
struct ConfigHelper {
ConfigHelper()
explicit ConfigHelper(bool audio_bwe_enabled)
: simulated_clock_(123456),
stream_config_(nullptr),
congestion_controller_(&simulated_clock_,
@ -87,7 +89,7 @@ struct ConfigHelper {
config.audio_mixer = AudioMixerImpl::Create();
audio_state_ = AudioState::Create(config);
SetupDefaultChannelProxy();
SetupDefaultChannelProxy(audio_bwe_enabled);
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
.WillOnce(Invoke([this](int channel_id) {
@ -102,8 +104,12 @@ struct ConfigHelper {
stream_config_.rtp.c_name = kCName;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
stream_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
if (audio_bwe_enabled) {
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberId));
stream_config_.send_codec_spec.transport_cc_enabled = true;
}
// Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
// calls from the default ctor behavior.
stream_config_.send_codec_spec.codec_inst = kIsacCodec;
@ -123,7 +129,7 @@ struct ConfigHelper {
RtcEventLog* event_log() { return &event_log_; }
MockVoiceEngine* voice_engine() { return &voice_engine_; }
void SetupDefaultChannelProxy() {
void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
using testing::StrEq;
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
@ -133,15 +139,25 @@ struct ConfigHelper {
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
RegisterSenderCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
packet_router()))
.Times(1);
if (audio_bwe_enabled) {
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
RegisterSenderCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
packet_router(), Ne(nullptr)))
.Times(1);
} else {
EXPECT_CALL(*channel_proxy_,
RegisterSenderCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
packet_router(), Eq(nullptr)))
.Times(1);
}
EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()).Times(1);
EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
@ -271,7 +287,7 @@ TEST(AudioSendStreamTest, ConfigToString) {
}
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -279,7 +295,7 @@ TEST(AudioSendStreamTest, ConstructDestruct) {
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -291,7 +307,7 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
}
TEST(AudioSendStreamTest, SetMuted) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -300,8 +316,24 @@ TEST(AudioSendStreamTest, SetMuted) {
send_stream.SetMuted(true);
}
TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(true);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -332,7 +364,7 @@ TEST(AudioSendStreamTest, GetStats) {
}
TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -351,7 +383,7 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
}
TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
ConfigHelper helper;
ConfigHelper helper(false);
auto stream_config = helper.config();
const CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000};
stream_config.send_codec_spec.codec_inst = kOpusCodec;
@ -396,7 +428,7 @@ TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
// VAD is applied when codec is mono and the CNG frequency matches the codec
// sample rate.
TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
ConfigHelper helper;
ConfigHelper helper(false);
auto stream_config = helper.config();
const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000};
stream_config.send_codec_spec.codec_inst = kG722Codec;
@ -411,7 +443,7 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
}
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),
@ -423,7 +455,7 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
}
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
ConfigHelper helper;
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.packet_router(), helper.congestion_controller(),

5
webrtc/test/mock_voe_channel_proxy.h
View File

@ -29,10 +29,11 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));
MOCK_METHOD3(RegisterSenderCongestionControlObjects,
MOCK_METHOD4(RegisterSenderCongestionControlObjects,
void(RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router));
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer));
MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
void(PacketRouter* packet_router));
MOCK_METHOD0(ResetCongestionControlObjects, void());

2
webrtc/video/video_quality_test.cc
View File

@ -927,6 +927,8 @@ std::string VideoQualityTest::GenerateGraphTitle() const {
}
void VideoQualityTest::CheckParams() {
if (!params_.video.enabled)
return;
// Add a default stream in none specified.
if (params_.ss.streams.empty())
params_.ss.streams.push_back(VideoQualityTest::DefaultVideoStream(params_));

27
webrtc/voice_engine/channel.cc
View File

@ -333,16 +333,32 @@ class StatisticsProxy : public RtcpStatisticsCallback {
class VoERtcpObserver : public RtcpBandwidthObserver {
public:
explicit VoERtcpObserver(Channel* owner) : owner_(owner) {}
explicit VoERtcpObserver(Channel* owner)
: owner_(owner), bandwidth_observer_(nullptr) {}
virtual ~VoERtcpObserver() {}
void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) {
rtc::CritScope lock(&crit_);
bandwidth_observer_ = bandwidth_observer;
}
void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
// Not used for Voice Engine.
rtc::CritScope lock(&crit_);
if (bandwidth_observer_) {
bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
}
}
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) override {
{
rtc::CritScope lock(&crit_);
if (bandwidth_observer_) {
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt,
now_ms);
}
}
// TODO(mflodman): Do we need to aggregate reports here or can we jut send
// what we get? I.e. do we ever get multiple reports bundled into one RTCP
// report for VoiceEngine?
@ -384,6 +400,8 @@ class VoERtcpObserver : public RtcpBandwidthObserver {
Channel* owner_;
// Maps remote side ssrc to extended highest sequence number received.
std::map<uint32_t, uint32_t> extended_max_sequence_number_;
rtc::CriticalSection crit_;
RtcpBandwidthObserver* bandwidth_observer_ GUARDED_BY(crit_);
};
int32_t Channel::SendData(FrameType frameType,
@ -2424,10 +2442,12 @@ void Channel::EnableReceiveTransportSequenceNumber(int id) {
void Channel::RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK(rtp_packet_sender);
RTC_DCHECK(transport_feedback_observer);
RTC_DCHECK(packet_router && !packet_router_);
rtcp_observer_->SetBandwidthObserver(bandwidth_observer);
feedback_observer_proxy_->SetTransportFeedbackObserver(
transport_feedback_observer);
seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
@ -2447,6 +2467,7 @@ void Channel::RegisterReceiverCongestionControlObjects(
void Channel::ResetCongestionControlObjects() {
RTC_DCHECK(packet_router_);
_rtpRtcpModule->SetStorePacketsStatus(false, 600);
rtcp_observer_->SetBandwidthObserver(nullptr);
feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr);
seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr);
packet_router_->RemoveRtpModule(_rtpRtcpModule.get());

13
webrtc/voice_engine/channel.h
View File

@ -304,12 +304,13 @@ class Channel
void EnableSendTransportSequenceNumber(int id);
void EnableReceiveTransportSequenceNumber(int id);
void RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router);
void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
void ResetCongestionControlObjects();
void RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer);
void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
void ResetCongestionControlObjects();
void SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);

6
webrtc/voice_engine/channel_proxy.cc
View File

@ -75,10 +75,12 @@ void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
void ChannelProxy::RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->RegisterSenderCongestionControlObjects(
rtp_packet_sender, transport_feedback_observer, packet_router);
rtp_packet_sender, transport_feedback_observer, packet_router,
bandwidth_observer);
}
void ChannelProxy::RegisterReceiverCongestionControlObjects(

4
webrtc/voice_engine/channel_proxy.h
View File

@ -27,6 +27,7 @@ namespace webrtc {
class AudioSinkInterface;
class PacketRouter;
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class RtpPacketSender;
class RtpReceiver;
@ -62,7 +63,8 @@ class ChannelProxy {
virtual void RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router);
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer);
virtual void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router);
virtual void ResetCongestionControlObjects();