Add Rtcp parameters for PeerConnection senders

Bug: webrtc:7580
Change-Id: Ibcf5e849a1f11f21fa75f6d006fecf1cd54f8552
Reviewed-on: https://webrtc-review.googlesource.com/78063
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23407}

api/ortc/rtptransportinterface.h

@@ -17,42 +17,13 @@
 #include "api/ortc/packettransportinterface.h"
 #include "api/rtcerror.h"
 #include "api/rtp_headers.h"
+#include "api/rtpparameters.h"
 #include "common_types.h"  // NOLINT(build/include)
 
 namespace webrtc {
 
 class RtpTransportAdapter;
 
-struct RtcpParameters final {
-  // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
-  // will be chosen by the implementation.
-  // TODO(deadbeef): Not implemented.
-  rtc::Optional<uint32_t> ssrc;
-
-  // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
-  //
-  // If empty in the construction of the RtpTransport, one will be generated by
-  // the implementation, and returned in GetRtcpParameters. Multiple
-  // RtpTransports created by the same OrtcFactory will use the same generated
-  // CNAME.
-  //
-  // If empty when passed into SetParameters, the CNAME simply won't be
-  // modified.
-  std::string cname;
-
-  // Send reduced-size RTCP?
-  bool reduced_size = false;
-
-  // Send RTCP multiplexed on the RTP transport?
-  bool mux = true;
-
-  bool operator==(const RtcpParameters& o) const {
-    return ssrc == o.ssrc && cname == o.cname &&
-           reduced_size == o.reduced_size && mux == o.mux;
-  }
-  bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
-};
-
 struct RtpTransportParameters final {
   RtcpParameters rtcp;
 

api/rtpparameters.cc

@@ -65,6 +65,9 @@ RtpCodecParameters::~RtpCodecParameters() {}
 RtpCapabilities::RtpCapabilities() {}
 RtpCapabilities::~RtpCapabilities() {}
 
+RtcpParameters::RtcpParameters() {}
+RtcpParameters::~RtcpParameters() {}
+
 RtpParameters::RtpParameters() {}
 RtpParameters::~RtpParameters() {}
 

api/rtpparameters.h

@@ -547,9 +547,40 @@ struct RtpCapabilities {
   bool operator!=(const RtpCapabilities& o) const { return !(*this == o); }
 };
 
-// Note that unlike in ORTC, an RtcpParameters structure is not included in
-// RtpParameters, because our API includes an additional "RtpTransport"
-// abstraction on which RTCP parameters are set.
+struct RtcpParameters final {
+  RtcpParameters();
+  ~RtcpParameters();
+
+  // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
+  // will be chosen by the implementation.
+  // TODO(deadbeef): Not implemented.
+  rtc::Optional<uint32_t> ssrc;
+
+  // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
+  //
+  // If empty in the construction of the RtpTransport, one will be generated by
+  // the implementation, and returned in GetRtcpParameters. Multiple
+  // RtpTransports created by the same OrtcFactory will use the same generated
+  // CNAME.
+  //
+  // If empty when passed into SetParameters, the CNAME simply won't be
+  // modified.
+  std::string cname;
+
+  // Send reduced-size RTCP?
+  bool reduced_size = false;
+
+  // Send RTCP multiplexed on the RTP transport?
+  // Not used with PeerConnection senders/receivers
+  bool mux = true;
+
+  bool operator==(const RtcpParameters& o) const {
+    return ssrc == o.ssrc && cname == o.cname &&
+           reduced_size == o.reduced_size && mux == o.mux;
+  }
+  bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
+};
+
 struct RtpParameters {
   RtpParameters();
   ~RtpParameters();
@@ -571,6 +602,11 @@ struct RtpParameters {
 
   std::vector<RtpEncodingParameters> encodings;
 
+  // Only available with a Peerconnection RtpSender.
+  // In ORTC, our API includes an additional "RtpTransport"
+  // abstraction on which RTCP parameters are set.
+  RtcpParameters rtcp;
+
   // TODO(deadbeef): Not implemented.
   DegradationPreference degradation_preference =
       DegradationPreference::BALANCED;
@@ -578,7 +614,7 @@ struct RtpParameters {
   bool operator==(const RtpParameters& o) const {
     return mid == o.mid && codecs == o.codecs &&
            header_extensions == o.header_extensions &&
-           encodings == o.encodings &&
+           encodings == o.encodings && rtcp == o.rtcp &&
            degradation_preference == o.degradation_preference;
   }
   bool operator!=(const RtpParameters& o) const { return !(*this == o); }

media/base/mediaengine.cc

@@ -33,6 +33,7 @@ webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
   }
   webrtc::RtpParameters parameters;
   parameters.encodings = encodings;
+  parameters.rtcp.cname = sp.cname;
   return parameters;
 }
 

media/engine/webrtcvideoengine.cc

@@ -1642,6 +1642,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
                                          : webrtc::RtcpMode::kCompound;
   parameters_.config.rtp.mid = send_params.mid;
 
+  rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size;
+
   if (codec_settings) {
     SetCodec(*codec_settings);
   }
@@ -1761,6 +1763,8 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
   bool recreate_stream = false;
   if (params.rtcp_mode) {
     parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
+    rtp_parameters_.rtcp.reduced_size =
+        parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
     recreate_stream = true;
   }
   if (params.rtp_header_extensions) {
@@ -1847,6 +1851,11 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters(
         RTCErrorType::INVALID_MODIFICATION,
         "Attempted to set RtpParameters with different encoding count");
   }
+  if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_MODIFICATION,
+        "Attempted to set RtpParameters with modified RTCP parameters");
+  }
   if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
                          "Attempted to set RtpParameters with modified SSRC");

media/engine/webrtcvideoengine_unittest.cc

@@ -4439,12 +4439,17 @@ TEST_F(WebRtcVideoChannelTest, TestSetSendRtcpReducedSize) {
   // Create stream, expecting that default mode is "compound".
   FakeVideoSendStream* stream1 = AddSendStream();
   EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpSendParameters(last_ssrc_);
+  EXPECT_FALSE(rtp_parameters.rtcp.reduced_size);
 
   // Now enable reduced size mode.
   send_parameters_.rtcp.reduced_size = true;
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
   stream1 = fake_call_->GetVideoSendStreams()[0];
   EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
+  rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
+  EXPECT_TRUE(rtp_parameters.rtcp.reduced_size);
 
   // Create a new stream and ensure it picks up the reduced size mode.
   FakeVideoSendStream* stream2 = AddSendStream();
@@ -5543,6 +5548,16 @@ TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersCodecs) {
             rtp_parameters.codecs[1]);
 }
 
+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
+TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) {
+  StreamParams params = StreamParams::CreateLegacy(kSsrc);
+  params.cname = "rtcpcname";
+  AddSendStream(params);
+
+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc);
+  EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
+}
+
 // Test that RtpParameters for send stream has one encoding and it has
 // the correct SSRC.
 TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {

media/engine/webrtcvoiceengine.cc

@@ -776,6 +776,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     config_.codec_pair_id = codec_pair_id;
     config_.track_id = track_id;
     rtp_parameters_.encodings[0].ssrc = ssrc;
+    rtp_parameters_.rtcp.cname = c_name;
 
     if (send_codec_spec) {
       UpdateSendCodecSpec(*send_codec_spec);
@@ -945,6 +946,11 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
           RTCErrorType::INVALID_MODIFICATION,
           "Attempted to set RtpParameters with different encoding count");
     }
+    if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
+      LOG_AND_RETURN_ERROR(
+          RTCErrorType::INVALID_MODIFICATION,
+          "Attempted to set RtpParameters with modified RTCP parameters");
+    }
     if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
                            "Attempted to set RtpParameters with modified SSRC");
@@ -992,6 +998,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     if (reconfigure_send_stream) {
       ReconfigureAudioSendStream();
     }
+
+    rtp_parameters_.rtcp.cname = config_.rtp.c_name;
+    rtp_parameters_.rtcp.reduced_size = false;
+
     // parameters.encodings[0].active could have changed.
     UpdateSendState();
     return webrtc::RTCError::OK();

media/engine/webrtcvoiceengine_unittest.cc

@@ -217,10 +217,14 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
   }
 
   bool SetupSendStream() {
+    return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX));
+  }
+
+  bool SetupSendStream(const cricket::StreamParams& sp) {
     if (!SetupChannel()) {
       return false;
     }
-    if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) {
+    if (!channel_->AddSendStream(sp)) {
       return false;
     }
     EXPECT_CALL(*apm_, set_output_will_be_muted(false));
@@ -1131,6 +1135,16 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
   EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
 }
 
+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
+  cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX);
+  params.cname = "rtcpcname";
+  EXPECT_TRUE(SetupSendStream(params));
+
+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
+  EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
+}
+
 // Test that GetRtpSendParameters returns an SSRC.
 TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
   EXPECT_TRUE(SetupSendStream());

sdk/BUILD.gn

@@ -683,6 +683,8 @@ if (is_ios || is_mac) {
         "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm",
         "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions+Private.h",
         "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm",
+        "objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h",
+        "objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm",
         "objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h",
         "objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm",
         "objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h",
@@ -718,6 +720,7 @@ if (is_ios || is_mac) {
         "objc/Framework/Headers/WebRTC/RTCPeerConnection.h",
         "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h",
         "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h",
+        "objc/Framework/Headers/WebRTC/RTCRtcpParameters.h",
         "objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h",
         "objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h",
         "objc/Framework/Headers/WebRTC/RTCRtpParameters.h",

sdk/android/api/org/webrtc/RtpParameters.java

@@ -118,8 +118,33 @@ public class RtpParameters {
     }
   }
 
+  public static class Rtcp {
+    /** The Canonical Name used by RTCP */
+    private final String cname;
+    /** Whether reduced size RTCP is configured or compound RTCP */
+    private final boolean reducedSize;
+
+    @CalledByNative("Rtcp")
+    Rtcp(String cname, boolean reducedSize) {
+      this.cname = cname;
+      this.reducedSize = reducedSize;
+    }
+
+    @CalledByNative("Rtcp")
+    public String getCname() {
+      return cname;
+    }
+
+    @CalledByNative("Rtcp")
+    public boolean getReducedSize() {
+      return reducedSize;
+    }
+  }
+
   public final String transactionId;
 
+  private final Rtcp rtcp;
+
   public final List<Encoding> encodings;
   // Codec parameters can't currently be changed between getParameters and
   // setParameters. Though in the future it will be possible to reorder them or
@@ -127,8 +152,9 @@ public class RtpParameters {
   public final List<Codec> codecs;
 
   @CalledByNative
-  RtpParameters(String transactionId, List<Encoding> encodings, List<Codec> codecs) {
+  RtpParameters(String transactionId, Rtcp rtcp, List<Encoding> encodings, List<Codec> codecs) {
     this.transactionId = transactionId;
+    this.rtcp = rtcp;
     this.encodings = encodings;
     this.codecs = codecs;
   }
@@ -138,6 +164,11 @@ public class RtpParameters {
     return transactionId;
   }
 
+  @CalledByNative
+  public Rtcp getRtcp() {
+    return rtcp;
+  }
+
   @CalledByNative
   List<Encoding> getEncodings() {
     return encodings;

sdk/android/src/jni/pc/rtpparameters.cc

@@ -39,6 +39,13 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpCodecParameter(
                                 NativeToJavaStringMap(env, codec.parameters));
 }
 
+ScopedJavaLocalRef<jobject> NativeToJavaRtpRtcpParameters(
+    JNIEnv* env,
+    const RtcpParameters& rtcp) {
+  return Java_Rtcp_Constructor(env, NativeToJavaString(env, rtcp.cname),
+                               rtcp.reduced_size);
+}
+
 }  // namespace
 
 RtpEncodingParameters JavaToNativeRtpEncodingParameters(
@@ -64,6 +71,13 @@ RtpParameters JavaToNativeRtpParameters(JNIEnv* jni,
       Java_RtpParameters_getTransactionId(jni, j_parameters);
   parameters.transaction_id = JavaToNativeString(jni, j_transaction_id);
 
+  ScopedJavaLocalRef<jobject> j_rtcp =
+      Java_RtpParameters_getRtcp(jni, j_parameters);
+  ScopedJavaLocalRef<jstring> j_rtcp_cname = Java_Rtcp_getCname(jni, j_rtcp);
+  jboolean j_rtcp_reduced_size = Java_Rtcp_getReducedSize(jni, j_rtcp);
+  parameters.rtcp.cname = JavaToNativeString(jni, j_rtcp_cname);
+  parameters.rtcp.reduced_size = j_rtcp_reduced_size;
+
   // Convert encodings.
   ScopedJavaLocalRef<jobject> j_encodings =
       Java_RtpParameters_getEncodings(jni, j_parameters);
@@ -99,6 +113,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpParameters(
     const RtpParameters& parameters) {
   return Java_RtpParameters_Constructor(
       env, NativeToJavaString(env, parameters.transaction_id),
+      NativeToJavaRtpRtcpParameters(env, parameters.rtcp),
       NativeToJavaList(env, parameters.encodings,
                        &NativeToJavaRtpEncodingParameter),
       NativeToJavaList(env, parameters.codecs, &NativeToJavaRtpCodecParameter));

sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtcpParameters.h"
+
+#include "api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtcpParameters ()
+
+/** Returns the equivalent native RtcpParameters structure. */
+@property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
+
+/** Initialize the object with a native RtcpParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm

@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtcpParameters+Private.h"
+
+#import "NSString+StdString.h"
+
+@implementation RTCRtcpParameters
+
+@synthesize cname = _cname;
+@synthesize isReducedSize = _isReducedSize;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
+  if (self = [self init]) {
+    _cname = [NSString stringForStdString:nativeParameters.cname];
+    _isReducedSize = nativeParameters.reduced_size;
+  }
+  return self;
+}
+
+- (webrtc::RtcpParameters)nativeParameters {
+  webrtc::RtcpParameters parameters;
+  parameters.cname = [NSString stdStringForString:_cname];
+  parameters.reduced_size = _isReducedSize;
+  return parameters;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm

@@ -11,12 +11,14 @@
 #import "RTCRtpParameters+Private.h"
 
 #import "NSString+StdString.h"
+#import "RTCRtcpParameters+Private.h"
 #import "RTCRtpCodecParameters+Private.h"
 #import "RTCRtpEncodingParameters+Private.h"
 
 @implementation RTCRtpParameters
 
 @synthesize transactionId = _transactionId;
+@synthesize rtcp = _rtcp;
 @synthesize encodings = _encodings;
 @synthesize codecs = _codecs;
 
@@ -28,6 +30,7 @@
     (const webrtc::RtpParameters &)nativeParameters {
   if (self = [self init]) {
     _transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
+    _rtcp = [[RTCRtcpParameters alloc] initWithNativeParameters:nativeParameters.rtcp];
     NSMutableArray *encodings = [[NSMutableArray alloc] init];
     for (const auto &encoding : nativeParameters.encodings) {
       [encodings addObject:[[RTCRtpEncodingParameters alloc]
@@ -48,6 +51,7 @@
 - (webrtc::RtpParameters)nativeParameters {
   webrtc::RtpParameters parameters;
   parameters.transaction_id = [NSString stdStringForString:_transactionId];
+  parameters.rtcp = [_rtcp nativeParameters];
   for (RTCRtpEncodingParameters *encoding in _encodings) {
     parameters.encodings.push_back(encoding.nativeParameters);
   }

sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCRtcpParameters : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h

@@ -11,6 +11,7 @@
 #import <Foundation/Foundation.h>
 
 #import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCRtcpParameters.h>
 #import <WebRTC/RTCRtpCodecParameters.h>
 #import <WebRTC/RTCRtpEncodingParameters.h>
 
@@ -22,6 +23,9 @@ RTC_EXPORT
 /** A unique identifier for the last set of parameters applied. */
 @property(nonatomic, copy) NSString *transactionId;
 
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTCRtcpParameters *rtcp;
+
 /** The currently active encodings in the order of preference. */
 @property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;