(Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/mediaconstraintsinterface.cc b/talk/app/webrtc/mediaconstraintsinterface.cc
index d59716e..a7b88aa 100644
--- a/talk/app/webrtc/mediaconstraintsinterface.cc
+++ b/talk/app/webrtc/mediaconstraintsinterface.cc
@@ -95,6 +95,8 @@
"googSuspendBelowMinBitrate";
const char MediaConstraintsInterface::kNumUnsignalledRecvStreams[] =
"googNumUnsignalledRecvStreams";
+const char MediaConstraintsInterface::kCombinedAudioVideoBwe[] =
+ "googCombinedAudioVideoBwe";
const char MediaConstraintsInterface::kScreencastMinBitrate[] =
"googScreencastMinBitrate";
// TODO(ronghuawu): Remove once cpu overuse detection is stable.
diff --git a/talk/app/webrtc/mediaconstraintsinterface.h b/talk/app/webrtc/mediaconstraintsinterface.h
index d74ffab..045da79 100644
--- a/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/talk/app/webrtc/mediaconstraintsinterface.h
@@ -117,6 +117,8 @@
// googSuspendBelowMinBitrate
static const char kNumUnsignalledRecvStreams[];
// googNumUnsignalledRecvStreams
+ // Constraint to enable combined audio+video bandwidth estimation.
+ static const char kCombinedAudioVideoBwe[]; // googCombinedAudioVideoBwe
static const char kScreencastMinBitrate[]; // googScreencastMinBitrate
static const char kCpuOveruseDetection[]; // googCpuOveruseDetection
static const char kCpuUnderuseThreshold[]; // googCpuUnderuseThreshold
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index ca022c6..a001d75 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -90,6 +90,8 @@
// warnings.
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
+static const int kMaxWaitForAudioDataMs = 10000;
+static const int kMaxWaitForRembMs = 5000;
#endif
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
@@ -154,11 +156,11 @@
}
void AddMediaStream(bool audio, bool video) {
- std::string label = kStreamLabelBase +
+ std::string stream_label = kStreamLabelBase +
rtc::ToString<int>(
static_cast<int>(peer_connection_->local_streams()->count()));
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
- peer_connection_factory_->CreateLocalMediaStream(label);
+ peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
FakeConstraints constraints;
@@ -169,13 +171,13 @@
peer_connection_factory_->CreateAudioSource(&constraints);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
+ std::string label = stream_label + kAudioTrackLabelBase;
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
- source));
+ peer_connection_factory_->CreateAudioTrack(label, source));
stream->AddTrack(audio_track);
}
if (video && can_receive_video()) {
- stream->AddTrack(CreateLocalVideoTrack(label));
+ stream->AddTrack(CreateLocalVideoTrack(stream_label));
}
EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
@@ -368,6 +370,17 @@
return observer->BytesSent();
}
+ int GetAvailableReceivedBandwidthStats() {
+ rtc::scoped_refptr<MockStatsObserver>
+ observer(new rtc::RefCountedObject<MockStatsObserver>());
+ EXPECT_TRUE(peer_connection_->GetStats(
+ observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
+ EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+ int bw = observer->AvailableReceiveBandwidth();
+ LOG(INFO) << "Available Receive Bandwidth: " << bw;
+ return bw;
+ }
+
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty() ? 1 :
@@ -447,6 +460,12 @@
webrtc::PeerConnectionInterface* pc() {
return peer_connection_.get();
}
+ void StopVideoCapturers() {
+ for (std::vector<cricket::VideoCapturer*>::iterator it =
+ video_capturers_.begin(); it != video_capturers_.end(); ++it) {
+ (*it)->Stop();
+ }
+ }
protected:
explicit PeerConnectionTestClientBase(const std::string& id)
@@ -529,10 +548,12 @@
FakeConstraints source_constraints = video_constraints_;
source_constraints.SetMandatoryMaxFrameRate(10);
+ cricket::FakeVideoCapturer* fake_capturer =
+ new webrtc::FakePeriodicVideoCapturer();
+ video_capturers_.push_back(fake_capturer);
rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
peer_connection_factory_->CreateVideoSource(
- new webrtc::FakePeriodicVideoCapturer(),
- &source_constraints);
+ fake_capturer, &source_constraints);
std::string label = stream_label + kVideoTrackLabelBase;
return peer_connection_factory_->CreateVideoTrack(label, source);
}
@@ -569,6 +590,10 @@
// For remote peer communication.
MessageReceiver* signaling_message_receiver_;
+
+ // Store references to the video capturers we've created, so that we can stop
+ // them, if required.
+ std::vector<cricket::VideoCapturer*> video_capturers_;
};
class JsepTestClient
@@ -1314,6 +1339,7 @@
// Unregister the existing observer.
receiving_client()->data_channel()->UnregisterObserver();
+
std::string data = "hello world";
SendRtpData(initializing_client()->data_channel(), data);
@@ -1437,4 +1463,80 @@
EnableVideoDecoderFactory();
LocalP2PTest();
}
+
+// Test receive bandwidth stats with only audio enabled at receiver.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsAudio) {
+ ASSERT_TRUE(CreateTestClients());
+ receiving_client()->SetReceiveAudioVideo(true, false);
+ LocalP2PTest();
+
+ // Wait until we have received some audio data.
+ StreamCollectionInterface* local_streams =
+ initializing_client()->local_streams();
+ ASSERT_GT(local_streams->count(), 0u);
+ ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
+ MediaStreamTrackInterface* local_audio_track =
+ local_streams->at(0)->GetAudioTracks()[0];
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetBytesReceivedStats(local_audio_track) > 10000,
+ kMaxWaitForAudioDataMs);
+
+ // Then wait for REMB.
+ EXPECT_EQ_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
+ kMaxWaitForRembMs);
+}
+
+// Test receive bandwidth stats with combined BWE.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsCombined) {
+ FakeConstraints setup_constraints;
+ setup_constraints.AddOptional(
+ MediaConstraintsInterface::kCombinedAudioVideoBwe, true);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ LocalP2PTest();
+
+ // Run until a non-zero bw is reported.
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats() > 40000,
+ kMaxWaitForRembMs);
+ int video_bw = receiving_client()->GetAvailableReceivedBandwidthStats();
+
+ // Halt video capturers, then run until we get a new non-zero bw which is
+ // lower than the previous value.
+ initializing_client()->StopVideoCapturers();
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats() < video_bw,
+ kMaxWaitForRembMs);
+ EXPECT_GT(receiving_client()->GetAvailableReceivedBandwidthStats(), 0);
+}
+
+// Test receive bandwidth stats with 1 video, 3 audio streams but no combined
+// BWE.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsNotCombined) {
+ FakeConstraints setup_constraints;
+ setup_constraints.AddOptional(
+ MediaConstraintsInterface::kCombinedAudioVideoBwe, false);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ LocalP2PTest();
+
+ // Run until a non-zero bw is reported.
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats() > 40000,
+ kMaxWaitForRembMs);
+
+ // Halt video capturers, then run until we get a new bw which is zero.
+ initializing_client()->StopVideoCapturers();
+ EXPECT_EQ_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
+ kMaxWaitForRembMs);
+}
+
#endif // if !defined(THREAD_SANITIZER)
diff --git a/talk/app/webrtc/test/mockpeerconnectionobservers.h b/talk/app/webrtc/test/mockpeerconnectionobservers.h
index 174b80b..0570d40 100644
--- a/talk/app/webrtc/test/mockpeerconnectionobservers.h
+++ b/talk/app/webrtc/test/mockpeerconnectionobservers.h
@@ -133,31 +133,37 @@
size_t number_of_reports() const { return reports_.size(); }
int AudioOutputLevel() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameAudioOutputLevel);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameAudioOutputLevel);
}
int AudioInputLevel() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameAudioInputLevel);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameAudioInputLevel);
}
int BytesReceived() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameBytesReceived);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameBytesReceived);
}
int BytesSent() {
- return GetSsrcStatsValue(webrtc::StatsReport::kStatsValueNameBytesSent);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameBytesSent);
+ }
+
+ int AvailableReceiveBandwidth() {
+ return GetStatsValue(StatsReport::kStatsReportTypeBwe,
+ StatsReport::kStatsValueNameAvailableReceiveBandwidth);
}
private:
- int GetSsrcStatsValue(StatsReport::StatsValueName name) {
+ int GetStatsValue(const std::string& type, StatsReport::StatsValueName name) {
if (reports_.empty()) {
return 0;
}
for (size_t i = 0; i < reports_.size(); ++i) {
- if (reports_[i].type != StatsReport::kStatsReportTypeSsrc)
+ if (reports_[i].type != type)
continue;
webrtc::StatsReport::Values::const_iterator it =
reports_[i].values.begin();
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 6d57afb..b90bf2c 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -471,14 +471,16 @@
}
WebRtcSession::~WebRtcSession() {
- if (voice_channel_.get()) {
- SignalVoiceChannelDestroyed();
- channel_manager_->DestroyVoiceChannel(voice_channel_.release());
- }
+ // Destroy video_channel_ first since it may have a pointer to the
+ // voice_channel_.
if (video_channel_.get()) {
SignalVideoChannelDestroyed();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
+ if (voice_channel_.get()) {
+ SignalVoiceChannelDestroyed();
+ channel_manager_->DestroyVoiceChannel(voice_channel_.release());
+ }
if (data_channel_.get()) {
SignalDataChannelDestroyed();
channel_manager_->DestroyDataChannel(data_channel_.release());
@@ -610,6 +612,10 @@
cricket::VideoOptions::HIGH);
}
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kCombinedAudioVideoBwe,
+ &audio_options_.combined_audio_video_bwe);
+
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,
@@ -1425,16 +1431,8 @@
void WebRtcSession::RemoveUnusedChannelsAndTransports(
const SessionDescription* desc) {
- const cricket::ContentInfo* voice_info =
- cricket::GetFirstAudioContent(desc);
- if ((!voice_info || voice_info->rejected) && voice_channel_) {
- mediastream_signaling_->OnAudioChannelClose();
- SignalVoiceChannelDestroyed();
- const std::string content_name = voice_channel_->content_name();
- channel_manager_->DestroyVoiceChannel(voice_channel_.release());
- DestroyTransportProxy(content_name);
- }
-
+ // Destroy video_channel_ first since it may have a pointer to the
+ // voice_channel_.
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(desc);
if ((!video_info || video_info->rejected) && video_channel_) {
@@ -1445,6 +1443,16 @@
DestroyTransportProxy(content_name);
}
+ const cricket::ContentInfo* voice_info =
+ cricket::GetFirstAudioContent(desc);
+ if ((!voice_info || voice_info->rejected) && voice_channel_) {
+ mediastream_signaling_->OnAudioChannelClose();
+ SignalVoiceChannelDestroyed();
+ const std::string content_name = voice_channel_->content_name();
+ channel_manager_->DestroyVoiceChannel(voice_channel_.release());
+ DestroyTransportProxy(content_name);
+ }
+
const cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(desc);
if ((!data_info || data_info->rejected) && data_channel_) {
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 206f320..2f731a9 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -3311,6 +3311,26 @@
SetAndVerifyNumUnsignalledRecvStreams(-1, 0);
}
+TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
+ true);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ SetLocalDescriptionWithoutError(offer);
+
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(voice_channel_ != NULL);
+ cricket::AudioOptions audio_options;
+ EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
+ EXPECT_TRUE(
+ audio_options.combined_audio_video_bwe.GetWithDefaultIfUnset(false));
+}
+
// Tests that we can renegotiate new media content with ICE candidates in the
// new remote SDP.
TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 62d6b61..5232e5d 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -182,6 +182,7 @@
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
dscp.SetFrom(change.dscp);
+ combined_audio_video_bwe.SetFrom(change.combined_audio_video_bwe);
}
bool operator==(const AudioOptions& o) const {
@@ -207,7 +208,8 @@
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
- dscp == o.dscp;
+ dscp == o.dscp &&
+ combined_audio_video_bwe == o.combined_audio_video_bwe;
}
std::string ToString() const {
@@ -238,6 +240,7 @@
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("dscp", dscp);
+ ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
ost << "}";
return ost.str();
}
@@ -275,6 +278,8 @@
Settable<uint32> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.
Settable<bool> dscp;
+ // Enable combined audio+bandwidth BWE.
+ Settable<bool> combined_audio_video_bwe;
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index 5cba380..ec4b9c6 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -335,6 +335,7 @@
unsigned int send_nack_bitrate_;
unsigned int send_bandwidth_;
unsigned int receive_bandwidth_;
+ // Bandwidth to deduct from estimated uplink capacity.
unsigned int reserved_transmit_bitrate_bps_;
bool suspend_below_min_bitrate_;
webrtc::CpuOveruseObserver* overuse_observer_;
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index f731b8d..fe44390 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -43,10 +43,7 @@
#ifdef USE_WEBRTC_DEV_BRANCH
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#endif
-
-namespace webrtc {
-class ViENetwork;
-}
+#include "webrtc/video_engine/include/vie_network.h"
namespace cricket {
@@ -316,6 +313,8 @@
}
webrtc::ViENetwork* GetViENetwork(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
+ // WARNING: This pointer is for verification purposes only. Calling
+ // functions on it may result in undefined behavior!
return channels_[channel]->vie_network;
}
int GetVideoChannel(int channel) {
@@ -999,6 +998,11 @@
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->vie_network = vie_network;
channels_[channel]->video_channel = video_channel;
+ if (vie_network) {
+ // The interface is released here to avoid leaks. A test should not
+ // attempt to call functions on the interface stored in the channel.
+ vie_network->Release();
+ }
return 0;
}
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 2462859..6d6ed25 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -1572,7 +1572,19 @@
bool WebRtcVideoMediaChannel::Init() {
const uint32 ssrc_key = 0;
- return CreateChannel(ssrc_key, MD_SENDRECV, &default_channel_id_);
+ bool result = CreateChannel(ssrc_key, MD_SENDRECV, &default_channel_id_);
+ if (!result) {
+ return false;
+ }
+ if (voice_channel_) {
+ WebRtcVoiceMediaChannel* voice_channel =
+ static_cast<WebRtcVoiceMediaChannel*>(voice_channel_);
+ if (!voice_channel->SetupSharedBandwidthEstimation(
+ engine()->vie()->engine(), default_channel_id_)) {
+ return false;
+ }
+ }
+ return true;
}
WebRtcVideoMediaChannel::~WebRtcVideoMediaChannel() {
@@ -1581,6 +1593,12 @@
const bool render = false;
SetRender(render);
+ if (voice_channel_) {
+ WebRtcVoiceMediaChannel* voice_channel =
+ static_cast<WebRtcVoiceMediaChannel*>(voice_channel_);
+ voice_channel->SetupSharedBandwidthEstimation(NULL, -1);
+ }
+
while (!send_channels_.empty()) {
if (!DeleteSendChannel(send_channels_.begin()->first)) {
LOG(LS_ERROR) << "Unable to delete channel with ssrc key "
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5cf53e4..8cb85b6 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -52,6 +52,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/video_engine/include/vie_network.h"
#ifdef WIN32
#include <objbase.h> // NOLINT
@@ -1759,6 +1760,8 @@
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
+ shared_bwe_vie_(NULL),
+ shared_bwe_vie_channel_(-1),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
@@ -1770,6 +1773,8 @@
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
+ ASSERT(shared_bwe_vie_ == NULL);
+ ASSERT(shared_bwe_vie_channel_ == -1);
// Remove any remaining send streams, the default channel will be deleted
// later.
@@ -1870,6 +1875,12 @@
}
}
+ // Force update of Video Engine BWE forwarding to reflect experiment setting.
+ if (!SetupSharedBandwidthEstimation(shared_bwe_vie_,
+ shared_bwe_vie_channel_)) {
+ return false;
+ }
+
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
@@ -2531,8 +2542,8 @@
}
if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
- LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
- return false;
+ LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
+ return false;
}
// Set the current codecs to be used for the new channel.
@@ -2604,6 +2615,9 @@
receive_channels_.insert(std::make_pair(
default_receive_ssrc_,
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport)));
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
return SetPlayout(voe_channel(), playout_);
}
@@ -2691,6 +2705,11 @@
return false;
}
+ // Set up channel to be able to forward incoming packets to video engine BWE.
+ if (!SetupSharedBweOnChannel(channel)) {
+ return false;
+ }
+
return SetPlayout(channel, playout_);
}
@@ -3060,7 +3079,8 @@
engine()->voe()->network()->ReceivedRTPPacket(
which_channel,
packet->data(),
- static_cast<unsigned int>(packet->length()));
+ static_cast<unsigned int>(packet->length()),
+ webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
@@ -3454,6 +3474,23 @@
return -1;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBandwidthEstimation(
+ webrtc::VideoEngine* vie, int vie_channel) {
+ shared_bwe_vie_ = vie;
+ shared_bwe_vie_channel_ = vie_channel;
+
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
+ for (ChannelMap::iterator it = receive_channels_.begin();
+ it != receive_channels_.end(); ++it) {
+ if (!SetupSharedBweOnChannel(it->second->channel())) {
+ return false;
+ }
+ }
+ return true;
+}
+
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
// Get the RED encodings from the parameter with no name. This may
@@ -3603,6 +3640,25 @@
return true;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBweOnChannel(int voe_channel) {
+ webrtc::ViENetwork* vie_network = NULL;
+ int vie_channel = -1;
+ if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false) &&
+ shared_bwe_vie_ != NULL && shared_bwe_vie_channel_ != -1) {
+ vie_network = webrtc::ViENetwork::GetInterface(shared_bwe_vie_);
+ vie_channel = shared_bwe_vie_channel_;
+ }
+ if (engine()->voe()->rtp()->SetVideoEngineBWETarget(voe_channel, vie_network,
+ vie_channel) == -1) {
+ LOG_RTCERR3(SetVideoEngineBWETarget, voe_channel, vie_network, vie_channel);
+ if (vie_network != NULL) {
+ // Don't fail if we're tearing down.
+ return false;
+ }
+ }
+ return true;
+}
+
int WebRtcSoundclipStream::Read(void *buf, int len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 69705cc..bc2a4e5 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -54,6 +54,10 @@
#error "Bogus include."
#endif
+namespace webrtc {
+class VideoEngine;
+}
+
namespace cricket {
// WebRtcSoundclipStream is an adapter object that allows a memory stream to be
@@ -377,6 +381,8 @@
int GetReceiveChannelNum(uint32 ssrc);
int GetSendChannelNum(uint32 ssrc);
+ bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
+ int vie_channel);
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
@@ -419,6 +425,7 @@
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
+ bool SetupSharedBweOnChannel(int voe_channel);
bool SetChannelRecvRtpHeaderExtensions(
int channel_id,
@@ -442,6 +449,11 @@
bool typing_noise_detected_;
SendFlags desired_send_;
SendFlags send_;
+ // shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
+ // VideoEngine channel that this voice channel should forward incoming packets
+ // to for Bandwidth Estimation purposes.
+ webrtc::VideoEngine* shared_bwe_vie_;
+ int shared_bwe_vie_channel_;
// send_channels_ contains the channels which are being used for sending.
// When the default channel (voe_channel) is used for sending, it is
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 89d4c4d..f5b8260 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -38,6 +38,7 @@
#include "talk/media/base/fakenetworkinterface.h"
#include "talk/media/base/fakertp.h"
#include "talk/media/webrtc/fakewebrtcvoiceengine.h"
+#include "talk/media/webrtc/webrtcvie.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "talk/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
@@ -3160,3 +3161,113 @@
CoUninitialize();
}
#endif
+
+TEST_F(WebRtcVoiceEngineTestFake, ChangeCombinedAudioVideoBweOption) {
+ // Test that changing the combined_audio_video_bwe option results in the
+ // expected state changes in VoiceEngine.
+ cricket::ViEWrapper vie;
+ const int kVieCh = 667;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
+ kVieCh));
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(2)));
+ int recv_ch = voe_.GetLastChannel();
+
+ // Combined BWE should not be set up yet.
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ // Enable combined BWE option - now it should be set up.
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
+
+ // Disable combined BWE option - should be disabled again.
+ options.combined_audio_video_bwe.Set(false);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, SetupSharedBandwidthEstimation) {
+ // Test that calling SetupSharedBandwidthEstimation() on the voice media
+ // channel results in the expected state changes in VoiceEngine.
+ cricket::ViEWrapper vie1;
+ cricket::ViEWrapper vie2;
+ const int kVieCh1 = 667;
+ const int kVieCh2 = 70;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(2)));
+ int recv_ch = voe_.GetLastChannel();
+
+ // Combined BWE should not be set up yet.
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ // Register - should be enabled.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie1.engine(),
+ kVieCh1));
+ EXPECT_EQ(vie1.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh1, voe_.GetVideoChannel(recv_ch));
+
+ // Re-register - should still be enabled.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie2.engine(),
+ kVieCh2));
+ EXPECT_EQ(vie2.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh2, voe_.GetVideoChannel(recv_ch));
+
+ // Unregister - should be disabled again.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, ConfigureCombinedBweForNewRecvStreams) {
+ // Test that adding receive streams after enabling combined bandwidth
+ // estimation will correctly configure each channel.
+ cricket::ViEWrapper vie;
+ const int kVieCh = 667;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
+ kVieCh));
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+
+ static const uint32 kSsrcs[] = {1, 2, 3, 4};
+ int voe_channels[ARRAY_SIZE(kSsrcs)] = {0};
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs[i])));
+ int recv_ch = media_channel->GetReceiveChannelNum(kSsrcs[i]);
+ EXPECT_NE(-1, recv_ch);
+ voe_channels[i] = recv_ch;
+ EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
+ }
+
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+
+ for (unsigned int i = 0; i < ARRAY_SIZE(voe_channels); ++i) {
+ EXPECT_EQ(NULL, voe_.GetViENetwork(voe_channels[i]));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(voe_channels[i]));
+ }
+}
