Adding API for setting bandwidth estimation configurations.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5773 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/experiments.h b/webrtc/experiments.h
index 4b5661b..b03d248 100644
--- a/webrtc/experiments.h
+++ b/webrtc/experiments.h
@@ -39,5 +39,14 @@
const bool enabled;
};
+
+struct AimdRemoteRateControl {
+ AimdRemoteRateControl() : enabled(false) {}
+ explicit AimdRemoteRateControl(bool set_enabled)
+ : enabled(set_enabled) {}
+ virtual ~AimdRemoteRateControl() {}
+
+ const bool enabled;
+};
} // namespace webrtc
#endif // WEBRTC_EXPERIMENTS_H_
diff --git a/webrtc/video_engine/include/vie_network.h b/webrtc/video_engine/include/vie_network.h
index 0f2a10c..bb36818 100644
--- a/webrtc/video_engine/include/vie_network.h
+++ b/webrtc/video_engine/include/vie_network.h
@@ -86,6 +86,13 @@
return 0;
}
+ // TODO(holmer): Remove the default implementation when this has been fixed
+ // in fakewebrtcvideoengine.cc.
+ virtual bool SetBandwidthEstimationConfig(int video_channel,
+ const webrtc::Config& config) {
+ return false;
+ }
+
protected:
ViENetwork() {}
virtual ~ViENetwork() {}
diff --git a/webrtc/video_engine/vie_channel_group.cc b/webrtc/video_engine/vie_channel_group.cc
index f4055e4..9ce52a2 100644
--- a/webrtc/video_engine/vie_channel_group.cc
+++ b/webrtc/video_engine/vie_channel_group.cc
@@ -17,6 +17,7 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/encoder_state_feedback.h"
@@ -40,8 +41,10 @@
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
engine_id_(engine_id),
min_bitrate_bps_(config.Get<RemoteBitrateEstimatorMinRate>().min_rate),
- rbe_(RemoteBitrateEstimatorFactory().Create(observer_, clock_,
- kMimdControl,
+ rate_control_type_(kMimdControl),
+ rbe_(RemoteBitrateEstimatorFactory().Create(observer_,
+ clock_,
+ rate_control_type_,
min_bitrate_bps_)),
using_absolute_send_time_(false),
packets_since_absolute_send_time_(0) {
@@ -56,7 +59,7 @@
int payload_size,
const RTPHeader& header) {
CriticalSectionScoped cs(crit_sect_.get());
- PickEstimator(header);
+ PickEstimatorFromHeader(header);
rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
}
@@ -91,19 +94,27 @@
return rbe_->GetStats(output);
}
+ void SetConfig(const webrtc::Config& config) {
+ CriticalSectionScoped cs(crit_sect_.get());
+ RateControlType new_control_type =
+ config.Get<AimdRemoteRateControl>().enabled ? kAimdControl :
+ kMimdControl;
+ if (new_control_type != rate_control_type_) {
+ rate_control_type_ = new_control_type;
+ PickEstimator();
+ }
+ }
+
private:
- // Instantiate RBE for Time Offset or Absolute Send Time extensions.
- void PickEstimator(const RTPHeader& header) {
+ void PickEstimatorFromHeader(const RTPHeader& header)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
if (header.extension.hasAbsoluteSendTime) {
// If we see AST in header, switch RBE strategy immediately.
if (!using_absolute_send_time_) {
- process_thread_->DeRegisterModule(rbe_.get());
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_),
"WrappingBitrateEstimator: Switching to absolute send time RBE.");
- rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
- observer_, clock_, kMimdControl, min_bitrate_bps_));
- process_thread_->RegisterModule(rbe_.get());
using_absolute_send_time_ = true;
+ PickEstimator();
}
packets_since_absolute_send_time_ = 0;
} else {
@@ -111,25 +122,36 @@
if (using_absolute_send_time_) {
++packets_since_absolute_send_time_;
if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
- process_thread_->DeRegisterModule(rbe_.get());
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_),
"WrappingBitrateEstimator: Switching to transmission time offset "
"RBE.");
- rbe_.reset(RemoteBitrateEstimatorFactory().Create(observer_, clock_,
- kMimdControl, min_bitrate_bps_));
- process_thread_->RegisterModule(rbe_.get());
using_absolute_send_time_ = false;
+ PickEstimator();
}
}
}
}
+ // Instantiate RBE for Time Offset or Absolute Send Time extensions.
+ void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
+ process_thread_->DeRegisterModule(rbe_.get());
+ if (using_absolute_send_time_) {
+ rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
+ observer_, clock_, rate_control_type_, min_bitrate_bps_));
+ } else {
+ rbe_.reset(RemoteBitrateEstimatorFactory().Create(
+ observer_, clock_, rate_control_type_, min_bitrate_bps_));
+ }
+ process_thread_->RegisterModule(rbe_.get());
+ }
+
RemoteBitrateObserver* observer_;
Clock* clock_;
ProcessThread* process_thread_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
const int engine_id_;
const uint32_t min_bitrate_bps_;
+ RateControlType rate_control_type_;
scoped_ptr<RemoteBitrateEstimator> rbe_;
bool using_absolute_send_time_;
uint32_t packets_since_absolute_send_time_;
@@ -224,4 +246,10 @@
}
return true;
}
+
+void ChannelGroup::SetBandwidthEstimationConfig(const webrtc::Config& config) {
+ WrappingBitrateEstimator* estimator =
+ static_cast<WrappingBitrateEstimator*>(remote_bitrate_estimator_.get());
+ estimator->SetConfig(config);
+}
} // namespace webrtc
diff --git a/webrtc/video_engine/vie_channel_group.h b/webrtc/video_engine/vie_channel_group.h
index 036967f..40ce71e 100644
--- a/webrtc/video_engine/vie_channel_group.h
+++ b/webrtc/video_engine/vie_channel_group.h
@@ -42,6 +42,7 @@
bool SetChannelRembStatus(int channel_id, bool sender, bool receiver,
ViEChannel* channel);
+ void SetBandwidthEstimationConfig(const webrtc::Config& config);
BitrateController* GetBitrateController();
CallStats* GetCallStats();
diff --git a/webrtc/video_engine/vie_channel_manager.cc b/webrtc/video_engine/vie_channel_manager.cc
index 37e4692..746cb25 100644
--- a/webrtc/video_engine/vie_channel_manager.cc
+++ b/webrtc/video_engine/vie_channel_manager.cc
@@ -389,6 +389,17 @@
}
}
+bool ViEChannelManager::SetBandwidthEstimationConfig(
+ int channel_id, const webrtc::Config& config) {
+ CriticalSectionScoped cs(channel_id_critsect_);
+ ChannelGroup* group = FindGroup(channel_id);
+ if (!group) {
+ return false;
+ }
+ group->SetBandwidthEstimationConfig(config);
+ return true;
+}
+
bool ViEChannelManager::CreateChannelObject(
int channel_id,
ViEEncoder* vie_encoder,
diff --git a/webrtc/video_engine/vie_channel_manager.h b/webrtc/video_engine/vie_channel_manager.h
index 8325098..a979ab6 100644
--- a/webrtc/video_engine/vie_channel_manager.h
+++ b/webrtc/video_engine/vie_channel_manager.h
@@ -79,6 +79,10 @@
// it will simply be ignored and no error is returned.
void UpdateSsrcs(int channel_id, const std::list<unsigned int>& ssrcs);
+ // Sets bandwidth estimation related configurations.
+ bool SetBandwidthEstimationConfig(int channel_id,
+ const webrtc::Config& config);
+
private:
// Creates a channel object connected to |vie_encoder|. Assumed to be called
// protected.
diff --git a/webrtc/video_engine/vie_network_impl.cc b/webrtc/video_engine/vie_network_impl.cc
index 8c3e1eb0..2b80159 100644
--- a/webrtc/video_engine/vie_network_impl.cc
+++ b/webrtc/video_engine/vie_network_impl.cc
@@ -217,4 +217,13 @@
vie_channel->ReceivedBWEPacket(arrival_time_ms, payload_size, header);
return 0;
}
+
+bool ViENetworkImpl::SetBandwidthEstimationConfig(
+ int video_channel, const webrtc::Config& config) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ return shared_data_->channel_manager()->SetBandwidthEstimationConfig(
+ video_channel, config);
+}
} // namespace webrtc
diff --git a/webrtc/video_engine/vie_network_impl.h b/webrtc/video_engine/vie_network_impl.h
index 26d8a9f..03547d8 100644
--- a/webrtc/video_engine/vie_network_impl.h
+++ b/webrtc/video_engine/vie_network_impl.h
@@ -38,8 +38,13 @@
const void* data,
const int length);
virtual int SetMTU(int video_channel, unsigned int mtu);
+
virtual int ReceivedBWEPacket(const int video_channel,
int64_t arrival_time_ms, int payload_size, const RTPHeader& header);
+
+ virtual bool SetBandwidthEstimationConfig(int video_channel,
+ const webrtc::Config& config);
+
protected:
explicit ViENetworkImpl(ViESharedData* shared_data);
virtual ~ViENetworkImpl();
