Replace uses of WebRTCRtpSource by RTCRtpSource

BUG=787254, 939192
R=guidou@chromium.org, haraken@chromium.org

Change-Id: Ic4d7594070d71ca8ca9153ca197a8d9b4ade053f
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1932089
Commit-Queue: Antonio Gomes <tonikitoo@igalia.com>
Reviewed-by: Kentaro Hara <haraken@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#719103}

third_party/blink/public/BUILD.gn

@@ -268,7 +268,6 @@ source_set("blink_headers") {
     "platform/web_rtc_peer_connection_handler.h",
     "platform/web_rtc_peer_connection_handler_client.h",
     "platform/web_rtc_rtp_receiver.h",
-    "platform/web_rtc_rtp_source.h",
     "platform/web_rtc_rtp_transceiver.h",
     "platform/web_rtc_stats.h",
     "platform/web_runtime_features.h",

third_party/blink/public/platform/web_rtc_rtp_receiver.h

@@ -18,8 +18,8 @@
 
 namespace blink {
 
+class RTCRtpSource;
 class WebMediaStreamTrack;
-class WebRTCRtpSource;
 
 // Implementations of this interface keep the corresponding WebRTC-layer
 // receiver alive through reference counting. Multiple |WebRTCRtpReceiver|s
@@ -40,7 +40,7 @@ class BLINK_PLATFORM_EXPORT WebRTCRtpReceiver {
   virtual webrtc::DtlsTransportInformation DtlsTransportInformation() = 0;
   virtual const WebMediaStreamTrack& Track() const = 0;
   virtual WebVector<WebString> StreamIds() const = 0;
-  virtual WebVector<std::unique_ptr<WebRTCRtpSource>> GetSources() = 0;
+  virtual WebVector<std::unique_ptr<RTCRtpSource>> GetSources() = 0;
   virtual void GetStats(blink::WebRTCStatsReportCallback,
                         const WebVector<webrtc::NonStandardGroupId>&) = 0;
   virtual std::unique_ptr<webrtc::RtpParameters> GetParameters() const = 0;

third_party/blink/public/platform/web_rtc_rtp_source.h

-// Copyright 2017 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#ifndef THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_RTP_SOURCE_H_
-#define THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_RTP_SOURCE_H_
-
-#include <memory>
-
-#include "base/optional.h"
-#include "third_party/blink/public/platform/web_common.h"
-
-namespace base {
-class TimeTicks;
-}
-
-namespace webrtc {
-class RtpSource;
-}
-
-namespace blink {
-
-// Represents both SSRCs and CSRCs.
-// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsynchronizationsource
-// https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource
-class BLINK_PLATFORM_EXPORT WebRTCRtpSource {
- public:
-  enum class Type {
-    kSSRC,
-    kCSRC,
-  };
-
-  virtual ~WebRTCRtpSource();
-
-  virtual Type SourceType() const = 0;
-  virtual base::TimeTicks Timestamp() const = 0;
-  virtual uint32_t Source() const = 0;
-  virtual base::Optional<double> AudioLevel() const = 0;
-  virtual uint32_t RtpTimestamp() const = 0;
-};
-
-BLINK_PLATFORM_EXPORT std::unique_ptr<WebRTCRtpSource> CreateRTCRtpSource(
-    const webrtc::RtpSource& source);
-
-}  // namespace blink
-
-#endif  // THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_RTP_SOURCE_H_

third_party/blink/renderer/modules/peerconnection/fake_rtc_rtp_transceiver_impl.cc

@@ -168,7 +168,7 @@ blink::WebVector<blink::WebString> FakeRTCRtpReceiverImpl::StreamIds() const {
   return web_stream_ids;
 }
 
-blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>>
+blink::WebVector<std::unique_ptr<RTCRtpSource>>
 FakeRTCRtpReceiverImpl::GetSources() {
   NOTIMPLEMENTED();
   return {};

third_party/blink/renderer/modules/peerconnection/fake_rtc_rtp_transceiver_impl.h

@@ -13,11 +13,11 @@
 #include "third_party/blink/public/platform/web_media_stream_source.h"
 #include "third_party/blink/public/platform/web_media_stream_track.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_receiver.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_transceiver.h"
 #include "third_party/blink/renderer/platform/mediastream/media_stream_audio_source.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_dtmf_sender_handler.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_sender_platform.h"
+#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
 
 namespace blink {
 
@@ -76,8 +76,7 @@ class FakeRTCRtpReceiverImpl : public blink::WebRTCRtpReceiver {
   webrtc::DtlsTransportInformation DtlsTransportInformation() override;
   const blink::WebMediaStreamTrack& Track() const override;
   blink::WebVector<blink::WebString> StreamIds() const override;
-  blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>> GetSources()
-      override;
+  blink::WebVector<std::unique_ptr<RTCRtpSource>> GetSources() override;
   void GetStats(blink::WebRTCStatsReportCallback,
                 const blink::WebVector<webrtc::NonStandardGroupId>&) override;
   std::unique_ptr<webrtc::RtpParameters> GetParameters() const override;

third_party/blink/renderer/modules/peerconnection/mock_web_rtc_peer_connection_handler.cc

@@ -9,13 +9,13 @@
 #include "third_party/blink/public/platform/web_media_stream.h"
 #include "third_party/blink/public/platform/web_media_stream_source.h"
 #include "third_party/blink/public/platform/web_media_stream_track.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_transceiver.h"
 #include "third_party/blink/public/platform/web_rtc_stats.h"
 #include "third_party/blink/public/platform/web_vector.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_dtmf_sender_handler.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_ice_candidate_platform.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_sender_platform.h"
+#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_session_description_platform.h"
 #include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
 #include "third_party/webrtc/api/stats/rtc_stats.h"
@@ -136,8 +136,8 @@ class DummyWebRTCRtpReceiver : public WebRTCRtpReceiver {
   WebVector<WebString> StreamIds() const override {
     return WebVector<WebString>();
   }
-  WebVector<std::unique_ptr<WebRTCRtpSource>> GetSources() override {
-    return WebVector<std::unique_ptr<WebRTCRtpSource>>();
+  WebVector<std::unique_ptr<RTCRtpSource>> GetSources() override {
+    return WebVector<std::unique_ptr<RTCRtpSource>>();
   }
   void GetStats(WebRTCStatsReportCallback,
                 const WebVector<webrtc::NonStandardGroupId>&) override {}

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc

@@ -6,7 +6,6 @@
 
 #include "third_party/blink/public/platform/web_media_stream.h"
 #include "third_party/blink/public/platform/web_media_stream_track.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/blink/renderer/core/loader/document_loader.h"
 #include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.h"
@@ -76,7 +75,7 @@ RTCRtpReceiver::getSynchronizationSources() {
 
   HeapVector<Member<RTCRtpSynchronizationSource>> synchronization_sources;
   for (const auto& web_source : web_sources_) {
-    if (web_source->SourceType() != WebRTCRtpSource::Type::kSSRC)
+    if (web_source->SourceType() != RTCRtpSource::Type::kSSRC)
       continue;
     RTCRtpSynchronizationSource* synchronization_source =
         MakeGarbageCollected<RTCRtpSynchronizationSource>();
@@ -103,7 +102,7 @@ RTCRtpReceiver::getContributingSources() {
 
   HeapVector<Member<RTCRtpContributingSource>> contributing_sources;
   for (const auto& web_source : web_sources_) {
-    if (web_source->SourceType() != WebRTCRtpSource::Type::kCSRC)
+    if (web_source->SourceType() != RTCRtpSource::Type::kCSRC)
       continue;
     RTCRtpContributingSource* contributing_source =
         MakeGarbageCollected<RTCRtpContributingSource>();

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.h

@@ -8,7 +8,6 @@
 #include "base/optional.h"
 #include "third_party/blink/public/platform/platform.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_receiver.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/blink/public/platform/web_vector.h"
 #include "third_party/blink/renderer/modules/mediastream/media_stream.h"
 #include "third_party/blink/renderer/modules/mediastream/media_stream_track.h"
@@ -20,6 +19,7 @@
 #include "third_party/blink/renderer/platform/heap/garbage_collected.h"
 #include "third_party/blink/renderer/platform/heap/member.h"
 #include "third_party/blink/renderer/platform/heap/visitor.h"
+#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
 
 namespace blink {
 class RTCDtlsTransport;
@@ -70,7 +70,7 @@ class RTCRtpReceiver final : public ScriptWrappable {
 
   // The current SSRCs and CSRCs. getSynchronizationSources() returns the SSRCs
   // and getContributingSources() returns the CSRCs.
-  WebVector<std::unique_ptr<WebRTCRtpSource>> web_sources_;
+  WebVector<std::unique_ptr<RTCRtpSource>> web_sources_;
   bool web_sources_needs_updating_ = true;
   Member<RTCRtpTransceiver> transceiver_;
 

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver_impl.cc

@@ -6,9 +6,9 @@
 
 #include "base/bind.h"
 #include "base/logging.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/blink/public/platform/web_rtc_stats.h"
 #include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_sender_platform.h"
+#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
 #include "third_party/blink/renderer/platform/peerconnection/webrtc_util.h"
 #include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
 #include "third_party/webrtc/api/scoped_refptr.h"
@@ -159,14 +159,14 @@ class RTCRtpReceiverImpl::RTCRtpReceiverInternal
     state_ = std::move(state);
   }
 
-  blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>> GetSources() {
+  blink::WebVector<std::unique_ptr<RTCRtpSource>> GetSources() {
     // The webrtc_recever_ is a proxy, so this is a blocking call to the webrtc
     // signalling thread.
     auto webrtc_sources = webrtc_receiver_->GetSources();
-    blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>> sources(
+    blink::WebVector<std::unique_ptr<RTCRtpSource>> sources(
         webrtc_sources.size());
     for (size_t i = 0; i < webrtc_sources.size(); ++i) {
-      sources[i] = blink::CreateRTCRtpSource(webrtc_sources[i]);
+      sources[i] = std::make_unique<RTCRtpSource>(webrtc_sources[i]);
     }
     return sources;
   }
@@ -296,7 +296,7 @@ blink::WebVector<blink::WebString> RTCRtpReceiverImpl::StreamIds() const {
   return web_stream_ids;
 }
 
-blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>>
+blink::WebVector<std::unique_ptr<RTCRtpSource>>
 RTCRtpReceiverImpl::GetSources() {
   return internal_->GetSources();
 }

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver_impl.h

@@ -130,8 +130,7 @@ class MODULES_EXPORT RTCRtpReceiverImpl : public blink::WebRTCRtpReceiver {
 
   const blink::WebMediaStreamTrack& Track() const override;
   blink::WebVector<blink::WebString> StreamIds() const override;
-  blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>> GetSources()
-      override;
+  blink::WebVector<std::unique_ptr<RTCRtpSource>> GetSources() override;
   void GetStats(blink::WebRTCStatsReportCallback,
                 const blink::WebVector<webrtc::NonStandardGroupId>&) override;
   std::unique_ptr<webrtc::RtpParameters> GetParameters() const override;

third_party/blink/renderer/platform/BUILD.gn

@@ -531,7 +531,6 @@ jumbo_component("platform") {
     "exported/web_resource_timing_info.cc",
     "exported/web_rtc_peer_connection_handler_client.cc",
     "exported/web_rtc_rtp_receiver.cc",
-    "exported/web_rtc_rtp_source.cc",
     "exported/web_rtc_rtp_transceiver.cc",
     "exported/web_rtc_stats.cc",
     "exported/web_runtime_features.cc",

third_party/blink/renderer/platform/exported/web_rtc_rtp_source.cc

-// Copyright 2017 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
-
-namespace blink {
-
-WebRTCRtpSource::~WebRTCRtpSource() = default;
-
-}  // namespace blink

third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.cc

@@ -12,24 +12,19 @@
 
 namespace blink {
 
-std::unique_ptr<WebRTCRtpSource> CreateRTCRtpSource(
-    const webrtc::RtpSource& source) {
-  return std::make_unique<RTCRtpSource>(source);
-}
-
 RTCRtpSource::RTCRtpSource(const webrtc::RtpSource& source) : source_(source) {}
 
 RTCRtpSource::~RTCRtpSource() {}
 
-WebRTCRtpSource::Type RTCRtpSource::SourceType() const {
+RTCRtpSource::Type RTCRtpSource::SourceType() const {
   switch (source_.source_type()) {
     case webrtc::RtpSourceType::SSRC:
-      return WebRTCRtpSource::Type::kSSRC;
+      return RTCRtpSource::Type::kSSRC;
     case webrtc::RtpSourceType::CSRC:
-      return WebRTCRtpSource::Type::kCSRC;
+      return RTCRtpSource::Type::kCSRC;
     default:
       NOTREACHED();
-      return WebRTCRtpSource::Type::kSSRC;
+      return RTCRtpSource::Type::kSSRC;
   }
 }
 

third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h

@@ -6,22 +6,35 @@
 #define THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_SOURCE_H_
 
 #include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
+#include "base/optional.h"
+#include "third_party/blink/renderer/platform/platform_export.h"
 #include "third_party/webrtc/api/rtp_receiver_interface.h"
 
+namespace base {
+class TimeTicks;
+}
+
+namespace webrtc {
+class RtpSource;
+}
+
 namespace blink {
 
-class RTCRtpSource : public WebRTCRtpSource {
+class PLATFORM_EXPORT RTCRtpSource {
  public:
+  enum class Type {
+    kSSRC,
+    kCSRC,
+  };
+
   explicit RTCRtpSource(const webrtc::RtpSource& source);
-  ~RTCRtpSource() override;
+  ~RTCRtpSource();
 
-  WebRTCRtpSource::Type SourceType() const override;
-  base::TimeTicks Timestamp() const override;
-  uint32_t Source() const override;
-  base::Optional<double> AudioLevel() const override;
-  uint32_t RtpTimestamp() const override;
+  Type SourceType() const;
+  base::TimeTicks Timestamp() const;
+  uint32_t Source() const;
+  base::Optional<double> AudioLevel() const;
+  uint32_t RtpTimestamp() const;
 
  private:
   const webrtc::RtpSource source_;