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-rw-r--r--talk/media/webrtc/webrtcvoiceengine.cc1
-rw-r--r--webrtc/audio/audio_send_stream.cc15
-rw-r--r--webrtc/audio/audio_send_stream.h4
-rw-r--r--webrtc/audio/audio_send_stream_unittest.cc52
-rw-r--r--webrtc/audio_send_stream.h2
-rw-r--r--webrtc/call/call.cc4
-rw-r--r--webrtc/call/call_perf_tests.cc91
-rw-r--r--webrtc/config.h6
-rw-r--r--webrtc/modules/pacing/paced_sender.cc14
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_sender.cc10
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc10
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc6
-rw-r--r--webrtc/test/mock_voe_channel_proxy.h5
-rw-r--r--webrtc/voice_engine/BUILD.gn1
-rw-r--r--webrtc/voice_engine/channel.cc309
-rw-r--r--webrtc/voice_engine/channel.h19
-rw-r--r--webrtc/voice_engine/channel_proxy.cc15
-rw-r--r--webrtc/voice_engine/channel_proxy.h10
-rw-r--r--webrtc/voice_engine/voice_engine.gyp1
19 files changed, 431 insertions, 144 deletions
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index d70c86495d..1ffc66b8fa 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -530,6 +530,7 @@ void WebRtcVoiceEngine::Construct() {
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
options_ = GetDefaultEngineOptions();
+ voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 2ff388bbca..35a65521dd 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -17,6 +17,9 @@
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_codec.h"
@@ -55,22 +58,31 @@ std::string AudioSendStream::Config::ToString() const {
namespace internal {
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
- const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ CongestionController* congestion_controller)
: config_(config), audio_state_(audio_state) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
+ RTC_DCHECK(congestion_controller);
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+ channel_proxy_->SetCongestionControlObjects(
+ congestion_controller->pacer(),
+ congestion_controller->GetTransportFeedbackObserver(),
+ congestion_controller->packet_router());
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+
for (const auto& extension : config.rtp.extensions) {
if (extension.name == RtpExtension::kAbsSendTime) {
channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
} else if (extension.name == RtpExtension::kAudioLevel) {
channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
+ } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+ channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";
}
@@ -80,6 +92,7 @@ AudioSendStream::AudioSendStream(
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+ channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
}
void AudioSendStream::Start() {
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 88304fd702..8b96350590 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -17,6 +17,7 @@
#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
+class CongestionController;
class VoiceEngine;
namespace voe {
@@ -27,7 +28,8 @@ namespace internal {
class AudioSendStream final : public webrtc::AudioSendStream {
public:
AudioSendStream(const webrtc::AudioSendStream::Config& config,
- const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ CongestionController* congestion_controller);
~AudioSendStream() override;
// webrtc::SendStream implementation.
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index c3620b294c..f90757584e 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -16,8 +16,12 @@
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video_engine/call_stats.h"
namespace webrtc {
namespace test {
@@ -31,6 +35,7 @@ const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
const int kAbsSendTimeId = 3;
+const int kTransportSequenceNumberId = 4;
const int kEchoDelayMedian = 254;
const int kEchoDelayStdDev = -3;
const int kEchoReturnLoss = -65;
@@ -45,7 +50,12 @@ const uint8_t kTelephoneEventCode = 45;
const uint32_t kTelephoneEventDuration = 6789;
struct ConfigHelper {
- ConfigHelper() : stream_config_(nullptr) {
+ ConfigHelper()
+ : stream_config_(nullptr),
+ process_thread_(ProcessThread::Create("AudioTestThread")),
+ congestion_controller_(process_thread_.get(),
+ &call_stats_,
+ &bitrate_observer_) {
using testing::Invoke;
using testing::StrEq;
@@ -68,6 +78,18 @@ struct ConfigHelper {
SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
+ EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
+ kTransportSequenceNumberId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(
+ congestion_controller_.pacer(),
+ congestion_controller_.GetTransportFeedbackObserver(),
+ congestion_controller_.packet_router()))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ .Times(1);
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
@@ -77,10 +99,15 @@ struct ConfigHelper {
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+ stream_config_.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
}
AudioSendStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+ CongestionController* congestion_controller() {
+ return &congestion_controller_;
+ }
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
@@ -126,10 +153,21 @@ struct ConfigHelper {
}
private:
+ class NullBitrateObserver : public BitrateObserver {
+ public:
+ virtual void OnNetworkChanged(uint32_t bitrate_bps,
+ uint8_t fraction_loss,
+ int64_t rtt_ms) {}
+ };
+
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+ CallStats call_stats_;
+ NullBitrateObserver bitrate_observer_;
+ rtc::scoped_ptr<ProcessThread> process_thread_;
+ CongestionController congestion_controller_;
};
} // namespace
@@ -152,12 +190,14 @@ TEST(AudioSendStreamTest, ConfigToString) {
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventCode, kTelephoneEventDuration));
@@ -165,7 +205,8 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -192,7 +233,8 @@ TEST(AudioSendStreamTest, GetStats) {
TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
helper.SetupMockForGetStats();
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index dd8d9e96ea..d1af9e0103 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -64,7 +64,7 @@ class AudioSendStream : public SendStream {
// Sender SSRC.
uint32_t ssrc = 0;
- // RTP header extensions used for the received stream.
+ // RTP header extensions used for the sent stream.
std::vector<RtpExtension> extensions;
// RTCP CNAME, see RFC 3550.
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 4156765d74..9209c7c97a 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -300,8 +300,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- AudioSendStream* send_stream =
- new AudioSendStream(config, config_.audio_state);
+ AudioSendStream* send_stream = new AudioSendStream(
+ config, config_.audio_state, congestion_controller_.get());
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
{
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 44ecae325e..caa5482273 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -18,6 +18,8 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -189,6 +191,8 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
const char* kSyncGroup = "av_sync";
+ const uint32_t kAudioSendSsrc = 1234;
+ const uint32_t kAudioRecvSsrc = 5678;
class AudioPacketReceiver : public PacketReceiver {
public:
AudioPacketReceiver(int channel, VoENetwork* voe_network)
@@ -228,37 +232,45 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
- int channel = voe_base->CreateChannel();
+ Config voe_config;
+ voe_config.Set<VoicePacing>(new VoicePacing(true));
+ int send_channel_id = voe_base->CreateChannel(voe_config);
+ int recv_channel_id = voe_base->CreateChannel();
SyncRtcpObserver audio_observer;
- AudioState::Config audio_state_config;
- audio_state_config.voice_engine = voice_engine;
+ AudioState::Config send_audio_state_config;
+ send_audio_state_config.voice_engine = voice_engine;
+ Call::Config sender_config;
+ sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config;
- receiver_config.audio_state = AudioState::Create(audio_state_config);
- CreateCalls(Call::Config(), receiver_config);
+ receiver_config.audio_state = sender_config.audio_state;
+ CreateCalls(sender_config, receiver_config);
- CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
- EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
-
- AudioPacketReceiver voe_packet_receiver(channel, voe_network);
+ AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
+ AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
net_config.loss_percent = 5;
test::PacketTransport audio_send_transport(
nullptr, &audio_observer, test::PacketTransport::kSender, net_config);
- audio_send_transport.SetReceiver(&voe_packet_receiver);
+ audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
test::PacketTransport audio_receive_transport(
nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);
- audio_receive_transport.SetReceiver(&voe_packet_receiver);
+ audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
- internal::TransportAdapter transport_adapter(&audio_send_transport);
- transport_adapter.Enable();
- EXPECT_EQ(0,
- voe_network->RegisterExternalTransport(channel, transport_adapter));
+ internal::TransportAdapter send_transport_adapter(&audio_send_transport);
+ send_transport_adapter.Enable();
+ EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
+ send_transport_adapter));
- VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), channel,
+ internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
+ recv_transport_adapter.Enable();
+ EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
+ recv_transport_adapter));
+
+ VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), recv_channel_id,
voe_sync, &audio_observer);
test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
@@ -275,6 +287,15 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
CreateSendConfig(1, &sync_send_transport);
CreateMatchingReceiveConfigs(&sync_receive_transport);
+ AudioSendStream::Config audio_send_config(&audio_send_transport);
+ audio_send_config.voe_channel_id = send_channel_id;
+ audio_send_config.rtp.ssrc = kAudioSendSsrc;
+ AudioSendStream* audio_send_stream =
+ sender_call_->CreateAudioSendStream(audio_send_config);
+
+ CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
+ EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
+
send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec) {
send_config_.rtp.fec.red_payload_type = kRedPayloadType;
@@ -286,20 +307,22 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
receive_configs_[0].renderer = &observer;
receive_configs_[0].sync_group = kSyncGroup;
- AudioReceiveStream::Config audio_config;
- audio_config.voe_channel_id = channel;
- audio_config.sync_group = kSyncGroup;
+ AudioReceiveStream::Config audio_recv_config;
+ audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
+ audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
+ audio_recv_config.voe_channel_id = recv_channel_id;
+ audio_recv_config.sync_group = kSyncGroup;
- AudioReceiveStream* audio_receive_stream = nullptr;
+ AudioReceiveStream* audio_receive_stream;
if (create_audio_first) {
audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
+ receiver_call_->CreateAudioReceiveStream(audio_recv_config);
CreateStreams();
} else {
CreateStreams();
audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
+ receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
CreateFrameGeneratorCapturer();
@@ -307,16 +330,16 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
Start();
fake_audio_device.Start();
- EXPECT_EQ(0, voe_base->StartPlayout(channel));
- EXPECT_EQ(0, voe_base->StartReceive(channel));
- EXPECT_EQ(0, voe_base->StartSend(channel));
+ EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
- EXPECT_EQ(0, voe_base->StopSend(channel));
- EXPECT_EQ(0, voe_base->StopReceive(channel));
- EXPECT_EQ(0, voe_base->StopPlayout(channel));
+ EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
+ EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
fake_audio_device.Stop();
Stop();
@@ -325,16 +348,18 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
audio_send_transport.StopSending();
audio_receive_transport.StopSending();
- voe_base->DeleteChannel(channel);
+ DestroyStreams();
+
+ sender_call_->DestroyAudioSendStream(audio_send_stream);
+ receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
+
+ voe_base->DeleteChannel(send_channel_id);
+ voe_base->DeleteChannel(recv_channel_id);
voe_base->Release();
voe_codec->Release();
voe_network->Release();
voe_sync->Release();
- DestroyStreams();
-
- receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
-
DestroyCalls();
VoiceEngine::Delete(voice_engine);
diff --git a/webrtc/config.h b/webrtc/config.h
index 114303e616..45a2ece416 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -138,6 +138,12 @@ struct NetEqFastAccelerate {
bool enabled;
};
+struct VoicePacing {
+ VoicePacing() : enabled(false) {}
+ explicit VoicePacing(bool value) : enabled(value) {}
+ bool enabled;
+};
+
} // namespace webrtc
#endif // WEBRTC_CONFIG_H_
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 37e8ed79da..bfa6b53b6b 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -358,10 +358,9 @@ int32_t PacedSender::Process() {
CriticalSectionScoped cs(critsect_.get());
int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
time_last_update_us_ = now_us;
- if (paused_)
- return 0;
int target_bitrate_kbps = max_bitrate_kbps_;
- if (elapsed_time_ms > 0) {
+ // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
+ if (!paused_ && elapsed_time_ms > 0) {
size_t queue_size_bytes = packets_->SizeInBytes();
if (queue_size_bytes > 0) {
// Assuming equal size packets and input/output rate, the average packet
@@ -389,7 +388,11 @@ int32_t PacedSender::Process() {
// element from the priority queue but keep it in storage, so that we can
// reinsert it if send fails.
const paced_sender::Packet& packet = packets_->BeginPop();
- if (SendPacket(packet)) {
+
+ // TODO(holmer): Because of this bug issue 5307 we have to send audio
+ // packets even when the pacer is paused. Here we assume audio packets are
+ // always high priority and that they are the only high priority packets.
+ if ((!paused_ || packet.priority == kHighPriority) && SendPacket(packet)) {
// Send succeeded, remove it from the queue.
packets_->FinalizePop(packet);
if (prober_->IsProbing())
@@ -401,7 +404,8 @@ int32_t PacedSender::Process() {
}
}
- if (!packets_->Empty())
+ // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
+ if (paused_ || !packets_->Empty())
return 0;
size_t padding_needed;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index dc544fbe69..d004cd8d4f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -469,7 +469,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
- LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+ LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+ << " not registered.";
return -1;
}
SetSendPayloadType(payload_type);
@@ -512,7 +513,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
- LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+ LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+ << static_cast<int>(payload_type) << ".";
return -1;
}
@@ -725,7 +727,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
// TickTime.
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(
- RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+ RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
corrected_capture_tims_ms, length - header.headerLength, true);
return length;
@@ -1003,7 +1005,7 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
}
size_t RTPSender::TimeToSendPadding(size_t bytes) {
- if (bytes == 0)
+ if (audio_configured_ || bytes == 0)
return 0;
{
CriticalSectionScoped cs(send_critsect_.get());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index f5df5b3b4a..3ae64117d2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -16,6 +16,7 @@
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
namespace webrtc {
@@ -368,7 +369,8 @@ int32_t RTPSenderAudio::SendAudio(
_rtpSender->Timestamp(), "seqnum",
_rtpSender->SequenceNumber());
return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
- -1, kAllowRetransmission,
+ TickTime::MillisecondTimestamp(),
+ kAllowRetransmission,
RtpPacketSender::kHighPriority);
}
@@ -476,9 +478,9 @@ RTPSenderAudio::SendTelephoneEventPacket(bool ended,
"Audio::SendTelephoneEvent", "timestamp",
dtmfTimeStamp, "seqnum",
_rtpSender->SequenceNumber());
- retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
- kAllowRetransmission,
- RtpPacketSender::kHighPriority);
+ retVal = _rtpSender->SendToNetwork(
+ dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+ kAllowRetransmission, RtpPacketSender::kHighPriority);
sendCount--;
}while (sendCount > 0 && retVal == 0);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 0209510d84..3e2a2b8034 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -104,7 +104,7 @@ void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
StorageType storage) {
if (_rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length,
capture_time_ms, storage,
- RtpPacketSender::kNormalPriority) == 0) {
+ RtpPacketSender::kLowPriority) == 0) {
_videoBitrate.Update(payload_length + rtp_header_length);
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"Video::PacketNormal", "timestamp", capture_timestamp,
@@ -150,7 +150,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
if (_rtpSender.SendToNetwork(
red_packet->data(), red_packet->length() - rtp_header_length,
rtp_header_length, capture_time_ms, media_packet_storage,
- RtpPacketSender::kNormalPriority) == 0) {
+ RtpPacketSender::kLowPriority) == 0) {
_videoBitrate.Update(red_packet->length());
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"Video::PacketRed", "timestamp", capture_timestamp,
@@ -162,7 +162,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
if (_rtpSender.SendToNetwork(
fec_packet->data(), fec_packet->length() - rtp_header_length,
rtp_header_length, capture_time_ms, fec_storage,
- RtpPacketSender::kNormalPriority) == 0) {
+ RtpPacketSender::kLowPriority) == 0) {
_fecOverheadRate.Update(fec_packet->length());
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"Video::PacketFec", "timestamp", capture_timestamp,
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index a0f0464111..b5d79c18ea 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -25,8 +25,13 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
+ MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
+ MOCK_METHOD3(SetCongestionControlObjects,
+ void(RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* seq_num_allocator));
MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 7a30a7bdbc..82cd92355c 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -106,6 +106,7 @@ source_set("voice_engine") {
"../modules/audio_processing",
"../modules/bitrate_controller",
"../modules/media_file",
+ "../modules/pacing",
"../modules/rtp_rtcp",
"../modules/utility",
"../system_wrappers",
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index fb9835664a..37dc3b685b 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -15,12 +15,14 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
@@ -44,6 +46,104 @@
namespace webrtc {
namespace voe {
+class TransportFeedbackProxy : public TransportFeedbackObserver {
+ public:
+ TransportFeedbackProxy() : feedback_observer_(nullptr) {
+ pacer_thread_.DetachFromThread();
+ network_thread_.DetachFromThread();
+ }
+
+ void SetTransportFeedbackObserver(
+ TransportFeedbackObserver* feedback_observer) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ feedback_observer_ = feedback_observer;
+ }
+
+ // Implements TransportFeedbackObserver.
+ void AddPacket(uint16_t sequence_number,
+ size_t length,
+ bool was_paced) override {
+ RTC_DCHECK(pacer_thread_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ if (feedback_observer_)
+ feedback_observer_->AddPacket(sequence_number, length, was_paced);
+ }
+ void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
+ RTC_DCHECK(network_thread_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ if (feedback_observer_)
+ feedback_observer_->OnTransportFeedback(feedback);
+ }
+
+ private:
+ rtc::CriticalSection crit_;
+ rtc::ThreadChecker thread_checker_;
+ rtc::ThreadChecker pacer_thread_;
+ rtc::ThreadChecker network_thread_;
+ TransportFeedbackObserver* feedback_observer_ GUARDED_BY(&crit_);
+};
+
+class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator {
+ public:
+ TransportSequenceNumberProxy() : seq_num_allocator_(nullptr) {
+ pacer_thread_.DetachFromThread();
+ }
+
+ void SetSequenceNumberAllocator(
+ TransportSequenceNumberAllocator* seq_num_allocator) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ seq_num_allocator_ = seq_num_allocator;
+ }
+
+ // Implements TransportSequenceNumberAllocator.
+ uint16_t AllocateSequenceNumber() override {
+ RTC_DCHECK(pacer_thread_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ if (!seq_num_allocator_)
+ return 0;
+ return seq_num_allocator_->AllocateSequenceNumber();
+ }
+
+ private:
+ rtc::CriticalSection crit_;
+ rtc::ThreadChecker thread_checker_;
+ rtc::ThreadChecker pacer_thread_;
+ TransportSequenceNumberAllocator* seq_num_allocator_ GUARDED_BY(&crit_);
+};
+
+class RtpPacketSenderProxy : public RtpPacketSender {
+ public:
+ RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {
+ }
+
+ void SetPacketSender(RtpPacketSender* rtp_packet_sender) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ rtp_packet_sender_ = rtp_packet_sender;
+ }
+
+ // Implements RtpPacketSender.
+ void InsertPacket(Priority priority,
+ uint32_t ssrc,
+ uint16_t sequence_number,
+ int64_t capture_time_ms,
+ size_t bytes,
+ bool retransmission) override {
+ rtc::CritScope lock(&crit_);
+ if (rtp_packet_sender_) {
+ rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number,
+ capture_time_ms, bytes, retransmission);
+ }
+ }
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ rtc::CriticalSection crit_;
+ RtpPacketSender* rtp_packet_sender_ GUARDED_BY(&crit_);
+};
+
// Extend the default RTCP statistics struct with max_jitter, defined as the
// maximum jitter value seen in an RTCP report block.
struct ChannelStatistics : public RtcpStatistics {
@@ -690,89 +790,97 @@ Channel::Channel(int32_t channelId,
uint32_t instanceId,
RtcEventLog* const event_log,
const Config& config)
- : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
- _instanceId(instanceId),
- _channelId(channelId),
- event_log_(event_log),
- rtp_header_parser_(RtpHeaderParser::Create()),
- rtp_payload_registry_(
- new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
- rtp_receive_statistics_(
- ReceiveStatistics::Create(Clock::GetRealTimeClock())),
- rtp_receiver_(
- RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
- this,
- this,
- this,
- rtp_payload_registry_.get())),
- telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
- _outputAudioLevel(),
- _externalTransport(false),
- _inputFilePlayerPtr(NULL),
- _outputFilePlayerPtr(NULL),
- _outputFileRecorderPtr(NULL),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
- _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
- _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
- _outputFileRecording(false),
- _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
- _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
- _outputExternalMedia(false),
- _inputExternalMediaCallbackPtr(NULL),
- _outputExternalMediaCallbackPtr(NULL),
- _timeStamp(0), // This is just an offset, RTP module will add it's own
- // random offset
- _sendTelephoneEventPayloadType(106),
- ntp_estimator_(Clock::GetRealTimeClock()),
- jitter_buffer_playout_timestamp_(0),
- playout_timestamp_rtp_(0),
- playout_timestamp_rtcp_(0),
- playout_delay_ms_(0),
- _numberOfDiscardedPackets(0),
- send_sequence_number_(0),
- ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
- capture_start_rtp_time_stamp_(-1),
- capture_start_ntp_time_ms_(-1),
- _engineStatisticsPtr(NULL),
- _outputMixerPtr(NULL),
- _transmitMixerPtr(NULL),
- _moduleProcessThreadPtr(NULL),
- _audioDeviceModulePtr(NULL),
- _voiceEngineObserverPtr(NULL),
- _callbackCritSectPtr(NULL),
- _transportPtr(NULL),
- _rxVadObserverPtr(NULL),
- _oldVadDecision(-1),
- _sendFrameType(0),
- _externalMixing(false),
- _mixFileWithMicrophone(false),
- _mute(false),
- _panLeft(1.0f),
- _panRight(1.0f),
- _outputGain(1.0f),
- _playOutbandDtmfEvent(false),
- _playInbandDtmfEvent(false),
- _lastLocalTimeStamp(0),
- _lastPayloadType(0),
- _includeAudioLevelIndication(false),
- _outputSpeechType(AudioFrame::kNormalSpeech),
- video_sync_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- _average_jitter_buffer_delay_us(0),
- _previousTimestamp(0),
- _recPacketDelayMs(20),
- _RxVadDetection(false),
- _rxAgcIsEnabled(false),
- _rxNsIsEnabled(false),
- restored_packet_in_use_(false),
- rtcp_observer_(new VoERtcpObserver(this)),
- network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
- assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- associate_send_channel_(ChannelOwner(nullptr)) {
+ : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ volume_settings_critsect_(
+ *CriticalSectionWrapper::CreateCriticalSection()),
+ _instanceId(instanceId),
+ _channelId(channelId),
+ event_log_(event_log),
+ rtp_header_parser_(RtpHeaderParser::Create()),
+ rtp_payload_registry_(
+ new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
+ rtp_receive_statistics_(
+ ReceiveStatistics::Create(Clock::GetRealTimeClock())),
+ rtp_receiver_(
+ RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
+ this,
+ this,
+ this,
+ rtp_payload_registry_.get())),
+ telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
+ _outputAudioLevel(),
+ _externalTransport(false),
+ _inputFilePlayerPtr(NULL),
+ _outputFilePlayerPtr(NULL),
+ _outputFileRecorderPtr(NULL),
+ // Avoid conflict with other channels by adding 1024 - 1026,
+ // won't use as much as 1024 channels.
+ _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
+ _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
+ _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
+ _outputFileRecording(false),
+ _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+ _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
+ _outputExternalMedia(false),
+ _inputExternalMediaCallbackPtr(NULL),
+ _outputExternalMediaCallbackPtr(NULL),
+ _timeStamp(0), // This is just an offset, RTP module will add it's own
+ // random offset
+ _sendTelephoneEventPayloadType(106),
+ ntp_estimator_(Clock::GetRealTimeClock()),
+ jitter_buffer_playout_timestamp_(0),
+ playout_timestamp_rtp_(0),
+ playout_timestamp_rtcp_(0),
+ playout_delay_ms_(0),
+ _numberOfDiscardedPackets(0),
+ send_sequence_number_(0),
+ ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
+ capture_start_rtp_time_stamp_(-1),
+ capture_start_ntp_time_ms_(-1),
+ _engineStatisticsPtr(NULL),
+ _outputMixerPtr(NULL),
+ _transmitMixerPtr(NULL),
+ _moduleProcessThreadPtr(NULL),
+ _audioDeviceModulePtr(NULL),
+ _voiceEngineObserverPtr(NULL),
+ _callbackCritSectPtr(NULL),
+ _transportPtr(NULL),
+ _rxVadObserverPtr(NULL),
+ _oldVadDecision(-1),
+ _sendFrameType(0),
+ _externalMixing(false),
+ _mixFileWithMicrophone(false),
+ _mute(false),
+ _panLeft(1.0f),
+ _panRight(1.0f),
+ _outputGain(1.0f),
+ _playOutbandDtmfEvent(false),
+ _playInbandDtmfEvent(false),
+ _lastLocalTimeStamp(0),
+ _lastPayloadType(0),
+ _includeAudioLevelIndication(false),
+ _outputSpeechType(AudioFrame::kNormalSpeech),
+ video_sync_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ _average_jitter_buffer_delay_us(0),
+ _previousTimestamp(0),
+ _recPacketDelayMs(20),
+ _RxVadDetection(false),
+ _rxAgcIsEnabled(false),
+ _rxNsIsEnabled(false),
+ restored_packet_in_use_(false),
+ rtcp_observer_(new VoERtcpObserver(this)),
+ network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
+ assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ associate_send_channel_(ChannelOwner(nullptr)),
+ pacing_enabled_(config.Get<VoicePacing>().enabled),
+ feedback_observer_proxy_(pacing_enabled_ ? new TransportFeedbackProxy()
+ : nullptr),
+ seq_num_allocator_proxy_(
+ pacing_enabled_ ? new TransportSequenceNumberProxy() : nullptr),
+ rtp_packet_sender_proxy_(pacing_enabled_ ? new RtpPacketSenderProxy()
+ : nullptr) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Channel() - ctor");
AudioCodingModule::Config acm_config;
@@ -797,6 +905,10 @@ Channel::Channel(int32_t channelId,
configuration.audio_messages = this;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.bandwidth_callback = rtcp_observer_.get();
+ configuration.paced_sender = rtp_packet_sender_proxy_.get();
+ configuration.transport_sequence_number_allocator =
+ seq_num_allocator_proxy_.get();
+ configuration.transport_feedback_callback = feedback_observer_proxy_.get();
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
@@ -2787,6 +2899,33 @@ int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
return 0;
}
+void Channel::EnableSendTransportSequenceNumber(int id) {
+ int ret =
+ SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
+ RTC_DCHECK_EQ(0, ret);
+}
+
+void Channel::SetCongestionControlObjects(
+ RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* packet_router) {
+ RTC_DCHECK(feedback_observer_proxy_.get());
+ RTC_DCHECK(seq_num_allocator_proxy_.get());
+ RTC_DCHECK(rtp_packet_sender_proxy_.get());
+ RTC_DCHECK(packet_router != nullptr || packet_router_ != nullptr);
+ feedback_observer_proxy_->SetTransportFeedbackObserver(
+ transport_feedback_observer);
+ seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
+ rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
+ _rtpRtcpModule->SetStorePacketsStatus(rtp_packet_sender != nullptr, 600);
+ if (packet_router != nullptr) {
+ packet_router->AddRtpModule(_rtpRtcpModule.get());
+ } else {
+ packet_router_->RemoveRtpModule(_rtpRtcpModule.get());
+ }
+ packet_router_ = packet_router;
+}
+
void Channel::SetRTCPStatus(bool enable) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCPStatus()");
@@ -3165,7 +3304,9 @@ bool Channel::GetCodecFECStatus() {
void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
// None of these functions can fail.
- _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
+ // If pacing is enabled we always store packets.
+ if (!pacing_enabled_)
+ _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
if (enable)
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index f26fdb23d4..d3b1b93645 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -11,13 +11,13 @@
#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
#define WEBRTC_VOICE_ENGINE_CHANNEL_H_
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_processing/rms_level.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -48,6 +48,7 @@ class AudioDeviceModule;
class Config;
class CriticalSectionWrapper;
class FileWrapper;
+class PacketRouter;
class ProcessThread;
class ReceiveStatistics;
class RemoteNtpTimeEstimator;
@@ -68,9 +69,12 @@ struct SenderInfo;
namespace voe {
class OutputMixer;
+class RtpPacketSenderProxy;
class Statistics;
class StatisticsProxy;
+class TransportFeedbackProxy;
class TransmitMixer;
+class TransportSequenceNumberProxy;
class VoERtcpObserver;
// Helper class to simplify locking scheme for members that are accessed from
@@ -321,6 +325,13 @@ public:
int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
+ void EnableSendTransportSequenceNumber(int id);
+
+ void SetCongestionControlObjects(
+ RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* packet_router);
+
void SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);
int SetRTCP_CNAME(const char cName[256]);
@@ -584,6 +595,12 @@ private:
// An associated send channel.
rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+
+ bool pacing_enabled_;
+ PacketRouter* packet_router_ = nullptr;
+ rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+ rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+ rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
};
} // namespace voe
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 1772ad5549..68fbf38863 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -52,6 +52,11 @@ void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int 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);
@@ -64,6 +69,15 @@ void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int 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};
@@ -124,5 +138,6 @@ Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
}
+
} // namespace voe
} // namespace webrtc
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 3668de4339..fa33e6caf8 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -19,6 +19,11 @@
#include <vector>
namespace webrtc {
+
+class PacketRouter;
+class RtpPacketSender;
+class TransportFeedbackObserver;
+
namespace voe {
class Channel;
@@ -41,8 +46,13 @@ class ChannelProxy {
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
+ virtual void EnableSendTransportSequenceNumber(int id);
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+ virtual void SetCongestionControlObjects(
+ RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* packet_router);
virtual CallStatistics GetRTCPStatistics() const;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp
index c9b0d859c8..ff588d8ead 100644
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@@ -23,6 +23,7 @@
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:media_file',
+ '<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-30 22:39:18 +0000