Return a copy of the supported RTP header extensions instead of a reference.

This also renames the method to better reflect what it does.

BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1486123002 .

Cr-Commit-Position: refs/heads/master@{#10910}

8 files changed, 69 insertions, 74 deletions

diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index bc454e5b9f..ab0c5eb1ff 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -694,12 +694,10 @@ class FakeBaseEngine {
         fail_create_channel_(false) {}
   void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
 
-  const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
-    return rtp_header_extensions_;
-  }
+  RtpCapabilities GetCapabilities() const { return capabilities_; }
   void set_rtp_header_extensions(
       const std::vector<RtpHeaderExtension>& extensions) {
-    rtp_header_extensions_ = extensions;
+    capabilities_.header_extensions = extensions;
   }
 
  protected:
@@ -708,7 +706,7 @@ class FakeBaseEngine {
   // TODO(thaloun): Replace with explicit checks of before & after values.
   bool options_changed_;
   bool fail_create_channel_;
-  std::vector<RtpHeaderExtension> rtp_header_extensions_;
+  RtpCapabilities capabilities_;
 };
 
 class FakeVoiceEngine : public FakeBaseEngine {
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index d39f2b45e6..c31fd72c84 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -57,6 +57,10 @@ namespace cricket {
 
 class VideoCapturer;
 
+struct RtpCapabilities {
+  std::vector<RtpHeaderExtension> header_extensions;
+};
+
 // MediaEngineInterface is an abstraction of a media engine which can be
 // subclassed to support different media componentry backends.
 // It supports voice and video operations in the same class to facilitate
@@ -108,11 +112,9 @@ class MediaEngineInterface {
   virtual int GetInputLevel() = 0;
 
   virtual const std::vector<AudioCodec>& audio_codecs() = 0;
-  virtual const std::vector<RtpHeaderExtension>&
-      audio_rtp_header_extensions() = 0;
+  virtual RtpCapabilities GetAudioCapabilities() = 0;
   virtual const std::vector<VideoCodec>& video_codecs() = 0;
-  virtual const std::vector<RtpHeaderExtension>&
-      video_rtp_header_extensions() = 0;
+  virtual RtpCapabilities GetVideoCapabilities() = 0;
 
   // Starts AEC dump using existing file.
   virtual bool StartAecDump(rtc::PlatformFile file) = 0;
@@ -202,14 +204,14 @@ class CompositeMediaEngine : public MediaEngineInterface {
   virtual const std::vector<AudioCodec>& audio_codecs() {
     return voice_.codecs();
   }
-  virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
-    return voice_.rtp_header_extensions();
+  virtual RtpCapabilities GetAudioCapabilities() {
+    return voice_.GetCapabilities();
   }
   virtual const std::vector<VideoCodec>& video_codecs() {
     return video_.codecs();
   }
-  virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
-    return video_.rtp_header_extensions();
+  virtual RtpCapabilities GetVideoCapabilities() {
+    return video_.GetCapabilities();
   }
 
   virtual bool StartAecDump(rtc::PlatformFile file) {
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 844b118527..e84cdab60d 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -488,20 +488,6 @@ WebRtcVideoEngine2::WebRtcVideoEngine2()
       external_encoder_factory_(NULL) {
   LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
   video_codecs_ = GetSupportedCodecs();
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
-                         kRtpTimestampOffsetHeaderExtensionDefaultId));
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
-                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
-                         kRtpVideoRotationHeaderExtensionDefaultId));
-  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
-    rtp_header_extensions_.push_back(RtpHeaderExtension(
-        kRtpTransportSequenceNumberHeaderExtension,
-        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
-  }
 }
 
 WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@@ -549,9 +535,23 @@ const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
   return video_codecs_;
 }
 
-const std::vector<RtpHeaderExtension>&
-WebRtcVideoEngine2::rtp_header_extensions() const {
-  return rtp_header_extensions_;
+RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
+  RtpCapabilities capabilities;
+  capabilities.header_extensions.push_back(
+      RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
+                         kRtpTimestampOffsetHeaderExtensionDefaultId));
+  capabilities.header_extensions.push_back(
+      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+  capabilities.header_extensions.push_back(
+      RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
+                         kRtpVideoRotationHeaderExtensionDefaultId));
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
+    capabilities.header_extensions.push_back(RtpHeaderExtension(
+        kRtpTransportSequenceNumberHeaderExtension,
+        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
+  }
+  return capabilities;
 }
 
 void WebRtcVideoEngine2::SetExternalDecoderFactory(
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index c5dc88c034..e6e26e7123 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -118,7 +118,7 @@ class WebRtcVideoEngine2 {
                                      const VideoOptions& options);
 
   const std::vector<VideoCodec>& codecs() const;
-  const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
+  RtpCapabilities GetCapabilities() const;
 
   // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
   // not take the ownership of |decoder_factory|. The caller needs to make sure
@@ -143,7 +143,6 @@ class WebRtcVideoEngine2 {
   std::vector<VideoCodec> GetSupportedCodecs() const;
 
   std::vector<VideoCodec> video_codecs_;
-  std::vector<RtpHeaderExtension> rtp_header_extensions_;
 
   bool initialized_;
 
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 832c4d0623..1de27546be 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -242,11 +242,11 @@ TEST_F(WebRtcVideoEngine2Test, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
 }
 
 TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
-  std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
-  ASSERT_FALSE(extensions.empty());
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    if (extensions[i].uri == kRtpTimestampOffsetHeaderExtension) {
-      EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extensions[i].id);
+  RtpCapabilities capabilities = engine_.GetCapabilities();
+  ASSERT_FALSE(capabilities.header_extensions.empty());
+  for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+    if (extension.uri == kRtpTimestampOffsetHeaderExtension) {
+      EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extension.id);
       return;
     }
   }
@@ -254,12 +254,11 @@ TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
 }
 
 TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
-  std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
-  ASSERT_FALSE(extensions.empty());
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
-      EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
-                extensions[i].id);
+  RtpCapabilities capabilities = engine_.GetCapabilities();
+  ASSERT_FALSE(capabilities.header_extensions.empty());
+  for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+    if (extension.uri == kRtpAbsoluteSenderTimeHeaderExtension) {
+      EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId, extension.id);
       return;
     }
   }
@@ -274,12 +273,12 @@ class WebRtcVideoEngine2WithSendSideBweTest : public WebRtcVideoEngine2Test {
 
 TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
        SupportsTransportSequenceNumberHeaderExtension) {
-  std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
-  ASSERT_FALSE(extensions.empty());
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    if (extensions[i].uri == kRtpTransportSequenceNumberHeaderExtension) {
+  RtpCapabilities capabilities = engine_.GetCapabilities();
+  ASSERT_FALSE(capabilities.header_extensions.empty());
+  for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+    if (extension.uri == kRtpTransportSequenceNumberHeaderExtension) {
       EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtensionDefaultId,
-                extensions[i].id);
+                extension.id);
       return;
     }
   }
@@ -287,11 +286,11 @@ TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
 }
 
 TEST_F(WebRtcVideoEngine2Test, SupportsVideoRotationHeaderExtension) {
-  std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
-  ASSERT_FALSE(extensions.empty());
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    if (extensions[i].uri == kRtpVideoRotationHeaderExtension) {
-      EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extensions[i].id);
+  RtpCapabilities capabilities = engine_.GetCapabilities();
+  ASSERT_FALSE(capabilities.header_extensions.empty());
+  for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+    if (extension.uri == kRtpVideoRotationHeaderExtension) {
+      EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extension.id);
       return;
     }
   }
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 1ffc66b8fa..fd0fc4be17 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -517,18 +517,6 @@ void WebRtcVoiceEngine::Construct() {
   // Load our audio codec list.
   codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
 
-  // Load our RTP Header extensions.
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
-                         kRtpAudioLevelHeaderExtensionDefaultId));
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
-                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
-  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
-    rtp_header_extensions_.push_back(RtpHeaderExtension(
-        kRtpTransportSequenceNumberHeaderExtension,
-        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
-  }
   options_ = GetDefaultEngineOptions();
   voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
 }
@@ -1075,10 +1063,20 @@ const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
   return codecs_;
 }
 
-const std::vector<RtpHeaderExtension>&
-WebRtcVoiceEngine::rtp_header_extensions() const {
+RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return rtp_header_extensions_;
+  RtpCapabilities capabilities;
+  capabilities.header_extensions.push_back(RtpHeaderExtension(
+      kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
+  capabilities.header_extensions.push_back(
+      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
+    capabilities.header_extensions.push_back(RtpHeaderExtension(
+        kRtpTransportSequenceNumberHeaderExtension,
+        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
+  }
+  return capabilities;
 }
 
 int WebRtcVoiceEngine::GetLastEngineError() {
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 89cf25c25f..3222861b21 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -80,7 +80,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   int GetInputLevel();
 
   const std::vector<AudioCodec>& codecs();
-  const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
+  RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
@@ -140,7 +140,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   webrtc::AudioDeviceModule* adm_ = nullptr;
   bool is_dumping_aec_ = false;
   std::vector<AudioCodec> codecs_;
-  std::vector<RtpHeaderExtension> rtp_header_extensions_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::AgcConfig default_agc_config_;
 
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 1a27a4c5cc..f639e74b5c 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -3109,17 +3109,17 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
   }
 
   // Set up receive extensions.
-  const auto& e_exts = engine_.rtp_header_extensions();
+  cricket::RtpCapabilities capabilities = engine_.GetCapabilities();
   cricket::AudioRecvParameters recv_parameters;
-  recv_parameters.extensions = e_exts;
+  recv_parameters.extensions = capabilities.header_extensions;
   channel_->SetRecvParameters(recv_parameters);
   EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
   for (uint32_t ssrc : ssrcs) {
     const auto* s = call_.GetAudioReceiveStream(ssrc);
     EXPECT_NE(nullptr, s);
     const auto& s_exts = s->GetConfig().rtp.extensions;
-    EXPECT_EQ(e_exts.size(), s_exts.size());
-    for (const auto& e_ext : e_exts) {
+    EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size());
+    for (const auto& e_ext : capabilities.header_extensions) {
       for (const auto& s_ext : s_exts) {
         if (e_ext.id == s_ext.id) {
           EXPECT_EQ(e_ext.uri, s_ext.name);