Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 235ca84..34f4d3f 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -68,6 +68,7 @@
RemoteBitrateEstimator* remote_bitrate_estimator;
PacedSender* paced_sender;
BitrateStatisticsObserver* send_bitrate_observer;
+ FrameCountObserver* send_frame_count_observer;
};
/*
@@ -340,10 +341,6 @@
virtual int TimeToSendPadding(int bytes) = 0;
- virtual void RegisterSendFrameCountObserver(
- FrameCountObserver* observer) = 0;
- virtual FrameCountObserver* GetSendFrameCountObserver() const = 0;
-
virtual bool GetSendSideDelay(int* avg_send_delay_ms,
int* max_send_delay_ms) const = 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 855d51b..a7f81c6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -38,7 +38,8 @@
audio_messages(NullObjectRtpAudioFeedback()),
remote_bitrate_estimator(NULL),
paced_sender(NULL),
- send_bitrate_observer(NULL) {
+ send_bitrate_observer(NULL),
+ send_frame_count_observer(NULL) {
}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
@@ -62,7 +63,8 @@
configuration.outgoing_transport,
configuration.audio_messages,
configuration.paced_sender,
- configuration.send_bitrate_observer),
+ configuration.send_bitrate_observer,
+ configuration.send_frame_count_observer),
rtcp_sender_(configuration.id,
configuration.audio,
configuration.clock,
@@ -1350,15 +1352,6 @@
return rtp_sender_.GetRtpStatisticsCallback();
}
-void ModuleRtpRtcpImpl::RegisterSendFrameCountObserver(
- FrameCountObserver* observer) {
- rtp_sender_.RegisterFrameCountObserver(observer);
-}
-
-FrameCountObserver* ModuleRtpRtcpImpl::GetSendFrameCountObserver() const {
- return rtp_sender_.GetFrameCountObserver();
-}
-
bool ModuleRtpRtcpImpl::IsDefaultModule() const {
CriticalSectionScoped cs(critical_section_module_ptrs_.get());
return !child_modules_.empty();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index b65131f..37a0fa4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -373,10 +373,6 @@
void OnRequestSendReport();
- virtual void RegisterSendFrameCountObserver(
- FrameCountObserver* observer) OVERRIDE;
- virtual FrameCountObserver* GetSendFrameCountObserver() const OVERRIDE;
-
protected:
void RegisterChildModule(RtpRtcp* module);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 858fc42..c98d498 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -46,7 +46,8 @@
Transport* transport,
RtpAudioFeedback* audio_feedback,
PacedSender* paced_sender,
- BitrateStatisticsObserver* bitrate_callback)
+ BitrateStatisticsObserver* bitrate_callback,
+ FrameCountObserver* frame_count_observer)
: clock_(clock),
bitrate_sent_(clock, this),
id_(id),
@@ -71,9 +72,9 @@
packet_history_(clock),
// Statistics
statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
- frame_count_observer_(NULL),
rtp_stats_callback_(NULL),
bitrate_callback_(bitrate_callback),
+ frame_count_observer_(frame_count_observer),
// RTP variables
start_timestamp_forced_(false),
start_timestamp_(0),
@@ -1664,16 +1665,6 @@
*length += 2;
}
-void RTPSender::RegisterFrameCountObserver(FrameCountObserver* observer) {
- CriticalSectionScoped cs(statistics_crit_.get());
- frame_count_observer_ = observer;
-}
-
-FrameCountObserver* RTPSender::GetFrameCountObserver() const {
- CriticalSectionScoped cs(statistics_crit_.get());
- return frame_count_observer_;
-}
-
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
CriticalSectionScoped cs(statistics_crit_.get());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 0cc35cf..265e69c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -70,7 +70,8 @@
RTPSender(const int32_t id, const bool audio, Clock *clock,
Transport *transport, RtpAudioFeedback *audio_feedback,
PacedSender *paced_sender,
- BitrateStatisticsObserver* bitrate_callback);
+ BitrateStatisticsObserver* bitrate_callback,
+ FrameCountObserver* frame_count_observer);
virtual ~RTPSender();
void ProcessBitrate();
@@ -265,9 +266,6 @@
int32_t SetFecParameters(const FecProtectionParams *delta_params,
const FecProtectionParams *key_params);
- virtual void RegisterFrameCountObserver(FrameCountObserver* observer);
- virtual FrameCountObserver* GetFrameCountObserver() const;
-
int SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms,
int32_t bytes, StorageType store,
bool force_full_size_packets, bool only_pad_after_markerbit);
@@ -373,11 +371,11 @@
scoped_ptr<CriticalSectionWrapper> statistics_crit_;
SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_);
- FrameCountObserver* frame_count_observer_ GUARDED_BY(statistics_crit_);
StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
BitrateStatisticsObserver* const bitrate_callback_;
+ FrameCountObserver* const frame_count_observer_;
// RTP variables
bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index e08aa20..42ee023 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -94,7 +94,7 @@
virtual void SetUp() {
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, NULL));
+ &mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@@ -672,7 +672,7 @@
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
- &mock_paced_sender_, NULL));
+ &mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
// Make all packets go through the pacer.
EXPECT_CALL(mock_paced_sender_,
@@ -817,6 +817,9 @@
uint32_t delta_frames_;
} callback;
+ rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
+ &mock_paced_sender_, NULL, &callback));
+
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
@@ -825,8 +828,6 @@
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
- rtp_sender_->RegisterFrameCountObserver(&callback);
-
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
4321, payload, sizeof(payload),
NULL));
@@ -845,7 +846,7 @@
EXPECT_EQ(1U, callback.key_frames_);
EXPECT_EQ(1U, callback.delta_frames_);
- rtp_sender_->RegisterFrameCountObserver(NULL);
+ rtp_sender_.reset();
}
TEST_F(RtpSenderTest, BitrateCallbacks) {
@@ -866,7 +867,7 @@
BitrateStatistics bitrate_;
} callback;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, &callback));
+ &mock_paced_sender_, &callback, NULL));
// Simulate kNumPackets sent with kPacketInterval ms intervals.
const uint32_t kNumPackets = 15;
@@ -922,7 +923,7 @@
virtual void SetUp() {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, NULL));
+ &mock_paced_sender_, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 4662251..64c2692 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -117,6 +117,7 @@
configuration.paced_sender = paced_sender;
configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
configuration.send_bitrate_observer = &send_bitrate_observer_;
+ configuration.send_frame_count_observer = &send_frame_count_observer_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
@@ -298,7 +299,6 @@
module_process_thread_.DeRegisterModule(rtp_rtcp);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
- rtp_rtcp->RegisterSendFrameCountObserver(NULL);
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
simulcast_rtp_rtcp_.pop_back();
@@ -346,8 +346,6 @@
rtp_rtcp->DeregisterSendRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime);
}
- rtp_rtcp->RegisterSendFrameCountObserver(
- rtp_rtcp_->GetSendFrameCountObserver());
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(
rtp_rtcp_->GetSendChannelRtcpStatisticsCallback());
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
@@ -362,7 +360,6 @@
module_process_thread_.DeRegisterModule(rtp_rtcp);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
- rtp_rtcp->RegisterSendFrameCountObserver(NULL);
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
simulcast_rtp_rtcp_.pop_back();
@@ -1524,7 +1521,6 @@
RtpRtcp* rtp_rtcp = CreateRtpRtcpModule();
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
- rtp_rtcp->RegisterSendFrameCountObserver(NULL);
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
removed_rtp_rtcp_.push_back(rtp_rtcp);
@@ -1714,13 +1710,7 @@
void ViEChannel::RegisterSendFrameCountObserver(
FrameCountObserver* observer) {
- rtp_rtcp_->RegisterSendFrameCountObserver(observer);
- CriticalSectionScoped cs(rtp_rtcp_cs_.get());
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- (*it)->RegisterSendFrameCountObserver(observer);
- }
+ send_frame_count_observer_.Set(observer);
}
void ViEChannel::ReceivedBWEPacket(int64_t arrival_time_ms,
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 39f9b75..50ec8ea 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -411,7 +411,8 @@
DISALLOW_COPY_AND_ASSIGN(RegisterableCallback);
};
- class : public RegisterableCallback<BitrateStatisticsObserver> {
+ class RegisterableBitrateStatisticsObserver:
+ public RegisterableCallback<BitrateStatisticsObserver> {
virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) {
CriticalSectionScoped cs(critsect_.get());
if (callback_)
@@ -420,6 +421,17 @@
}
send_bitrate_observer_;
+ class RegisterableFrameCountObserver
+ : public RegisterableCallback<FrameCountObserver> {
+ virtual void FrameCountUpdated(FrameType frame_type,
+ uint32_t frame_count,
+ const unsigned int ssrc) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (callback_)
+ callback_->FrameCountUpdated(frame_type, frame_count, ssrc);
+ }
+ } send_frame_count_observer_;
+
int32_t channel_id_;
int32_t engine_id_;
uint32_t number_of_cores_;
