Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver.

For use in AudiReceiveStream, introduce a new method GetSyncInfo. This change is analogous to https://webrtc-review.googlesource.com/91123, doing the same for RtpVideoStreamReceiver. It's a preparation for bypassing the RtpReceiver class. Bug: webrtc:7135 Change-Id: I87c1c6f0a1f28b0baebe07c4181f6f0427afa314 Reviewed-on: https://webrtc-review.googlesource.com/93022 Reviewed-by: Oskar Sundbom <[email protected]> Commit-Queue: Niels Moller <[email protected]> Cr-Commit-Position: refs/heads/master@{#24228}
longv2go · Aug 8, 2018 · 848d6d3 · 848d6d3
1 parent 43fa99c
commit 848d6d3
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Showing 8 changed files with 43 additions and 40 deletions.
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
@@ -257,26 +257,12 @@ int AudioReceiveStream::id() const {
 
 absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
   RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
-  Syncable::Info info;
+  absl::optional<Syncable::Info> info = channel_proxy_->GetSyncInfo();
 
-  RtpRtcp* rtp_rtcp = nullptr;
-  RtpReceiver* rtp_receiver = nullptr;
-  channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
-  RTC_DCHECK(rtp_rtcp);
-  RTC_DCHECK(rtp_receiver);
-
-  if (!rtp_receiver->GetLatestTimestamps(
-          &info.latest_received_capture_timestamp,
-          &info.latest_receive_time_ms)) {
-    return absl::nullopt;
-  }
-  if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
-                          &info.capture_time_ntp_frac, nullptr, nullptr,
-                          &info.capture_time_source_clock) != 0) {
+  if (!info)
     return absl::nullopt;
-  }
 
-  info.current_delay_ms = channel_proxy_->GetDelayEstimate();
+  info->current_delay_ms = channel_proxy_->GetDelayEstimate();
   return info;
 }
 

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
@@ -141,8 +141,7 @@ AudioSendStream::AudioSendStream(
 
   channel_proxy_->SetRtcEventLog(event_log_);
   channel_proxy_->SetRTCPStatus(true);
-  RtpReceiver* rtpReceiver = nullptr;  // Unused, but required for call.
-  channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
+  rtp_rtcp_module_ = channel_proxy_->GetRtpRtcp();
   RTC_DCHECK(rtp_rtcp_module_);
 
   ConfigureStream(this, config, true);

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
@@ -189,12 +189,9 @@ struct ConfigHelper {
   void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
     EXPECT_TRUE(channel_proxy_ == nullptr);
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
-    EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
-        .WillRepeatedly(Invoke(
-            [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
-              *rtp_rtcp_module = &this->rtp_rtcp_;
-              *rtp_receiver = nullptr;  // Not deemed necessary for tests yet.
-            }));
+    EXPECT_CALL(*channel_proxy_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
+      return &this->rtp_rtcp_;
+    }));
     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);

diff --git a/audio/channel.cc b/audio/channel.cc
@@ -1342,11 +1342,24 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
   return 0;
 }
 
-int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
-                        RtpReceiver** rtp_receiver) const {
-  *rtpRtcpModule = _rtpRtcpModule.get();
-  *rtp_receiver = rtp_receiver_.get();
-  return 0;
+RtpRtcp* Channel::GetRtpRtcp() const {
+  return _rtpRtcpModule.get();
+}
+
+absl::optional<Syncable::Info> Channel::GetSyncInfo() const {
+  Syncable::Info info;
+  if (!rtp_receiver_->GetLatestTimestamps(
+          &info.latest_received_capture_timestamp,
+          &info.latest_receive_time_ms)) {
+    return absl::nullopt;
+  }
+  if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs,
+                                &info.capture_time_ntp_frac, nullptr, nullptr,
+                                &info.capture_time_source_clock) != 0) {
+    return absl::nullopt;
+  }
+
+  return info;
 }
 
 void Channel::UpdatePlayoutTimestamp(bool rtcp) {

diff --git a/audio/channel.h b/audio/channel.h
@@ -22,6 +22,7 @@
 #include "api/call/audio_sink.h"
 #include "api/call/transport.h"
 #include "audio/audio_level.h"
+#include "call/syncable.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_processing/rms_level.h"
@@ -209,8 +210,12 @@ class Channel
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);  // NOLINT
-  int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
+  // Used by AudioSendStream.
+  RtpRtcp* GetRtpRtcp() const;
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  absl::optional<Syncable::Info> GetSyncInfo() const;
   // DTMF.
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);

diff --git a/audio/channel_proxy.cc b/audio/channel_proxy.cc
@@ -265,13 +265,14 @@ void ChannelProxy::DisassociateSendChannel() {
   channel_->SetAssociatedSendChannel(nullptr);
 }
 
-void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
-                              RtpReceiver** rtp_receiver) const {
+RtpRtcp* ChannelProxy::GetRtpRtcp() const {
   RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
-  RTC_DCHECK(rtp_rtcp);
-  RTC_DCHECK(rtp_receiver);
-  int error = channel_->GetRtpRtcp(rtp_rtcp, rtp_receiver);
-  RTC_DCHECK_EQ(0, error);
+  return channel_->GetRtpRtcp();
+}
+
+absl::optional<Syncable::Info> ChannelProxy::GetSyncInfo() const {
+  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
+  return channel_->GetSyncInfo();
 }
 
 uint32_t ChannelProxy::GetPlayoutTimestamp() const {

diff --git a/audio/channel_proxy.h b/audio/channel_proxy.h
@@ -108,7 +108,10 @@ class ChannelProxy : public RtpPacketSinkInterface {
   virtual void SetTransportOverhead(int transport_overhead_per_packet);
   virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
   virtual void DisassociateSendChannel();
-  virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp, RtpReceiver** rtp_receiver) const;
+  virtual RtpRtcp* GetRtpRtcp() const;
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  absl::optional<Syncable::Info> GetSyncInfo() const;
   virtual uint32_t GetPlayoutTimestamp() const;
   virtual void SetMinimumPlayoutDelay(int delay_ms);
   virtual bool GetRecCodec(CodecInst* codec_inst) const;

diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
@@ -83,8 +83,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_METHOD1(AssociateSendChannel,
                void(const ChannelProxy& send_channel_proxy));
   MOCK_METHOD0(DisassociateSendChannel, void());
-  MOCK_CONST_METHOD2(GetRtpRtcp,
-                     void(RtpRtcp** rtp_rtcp, RtpReceiver** rtp_receiver));
+  MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
   MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));