Onion soup RtcRtpSource

This is phase 1.4 on the design document [1].

[1] https://docs.google.com/document/d/1AJKVA5U4nDkyDB9p4ROrggWXadCxyy-grKaE9KS5vOU/

rtc_rtp_receiver.cc|h are moved to renderer/platform/peerconnection,
and web_rtc_rtp_source.h gets a factory method added so
content can still access its functionality.

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

Change-Id: I5591f8aa79eea7f08c5c3460fc79212ab5fb4805
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1770240
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@{#690732}

content/renderer/BUILD.gn

@@ -216,8 +216,6 @@ target(link_target_type, "renderer") {
     "media/webrtc/rtc_rtp_receiver.h",
     "media/webrtc/rtc_rtp_sender.cc",
     "media/webrtc/rtc_rtp_sender.h",
-    "media/webrtc/rtc_rtp_source.cc",
-    "media/webrtc/rtc_rtp_source.h",
     "media/webrtc/rtc_rtp_transceiver.cc",
     "media/webrtc/rtc_rtp_transceiver.h",
     "media/webrtc/rtc_stats.cc",

content/renderer/media/webrtc/rtc_rtp_receiver.cc

@@ -6,9 +6,9 @@
 
 #include "base/bind.h"
 #include "base/logging.h"
-#include "content/renderer/media/webrtc/rtc_rtp_source.h"
 #include "content/renderer/media/webrtc/rtc_stats.h"
 #include "content/renderer/media/webrtc/webrtc_util.h"
+#include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/webrtc/api/scoped_refptr.h"
 
 namespace content {
@@ -163,7 +163,7 @@ class RTCRtpReceiver::RTCRtpReceiverInternal
     blink::WebVector<std::unique_ptr<blink::WebRTCRtpSource>> sources(
         webrtc_sources.size());
     for (size_t i = 0; i < webrtc_sources.size(); ++i) {
-      sources[i] = std::make_unique<RTCRtpSource>(webrtc_sources[i]);
+      sources[i] = blink::CreateRTCRtpSource(webrtc_sources[i]);
     }
     return sources;
   }

third_party/blink/public/platform/web_rtc_rtp_source.h

@@ -12,6 +12,10 @@ namespace base {
 class TimeTicks;
 }
 
+namespace webrtc {
+class RtpSource;
+}
+
 namespace blink {
 
 // Represents both SSRCs and CSRCs.
@@ -33,6 +37,9 @@ class BLINK_PLATFORM_EXPORT WebRTCRtpSource {
   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/platform/BUILD.gn

@@ -1231,6 +1231,8 @@ jumbo_component("platform") {
     "peerconnection/rtc_dtmf_sender_handler.cc",
     "peerconnection/rtc_dtmf_sender_handler.h",
     "peerconnection/rtc_offer_options_platform.h",
+    "peerconnection/rtc_rtp_source.cc",
+    "peerconnection/rtc_rtp_source.h",
     "peerconnection/rtc_session_description_request.h",
     "peerconnection/rtc_stats_request.h",
     "peerconnection/rtc_stats_response_base.h",

content/renderer/media/webrtc/rtc_rtp_source.cc → third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.cc

@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "content/renderer/media/webrtc/rtc_rtp_source.h"
+#include "third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h"
 
 #include <cmath>
 
@@ -10,21 +10,26 @@
 #include "base/time/time.h"
 #include "third_party/webrtc/api/scoped_refptr.h"
 
-namespace content {
+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() {}
 
-blink::WebRTCRtpSource::Type RTCRtpSource::SourceType() const {
+WebRTCRtpSource::Type RTCRtpSource::SourceType() const {
   switch (source_.source_type()) {
     case webrtc::RtpSourceType::SSRC:
-      return blink::WebRTCRtpSource::Type::kSSRC;
+      return WebRTCRtpSource::Type::kSSRC;
     case webrtc::RtpSourceType::CSRC:
-      return blink::WebRTCRtpSource::Type::kCSRC;
+      return WebRTCRtpSource::Type::kCSRC;
     default:
       NOTREACHED();
-      return blink::WebRTCRtpSource::Type::kSSRC;
+      return WebRTCRtpSource::Type::kSSRC;
   }
 }
 
@@ -54,4 +59,4 @@ uint32_t RTCRtpSource::RtpTimestamp() const {
   return source_.rtp_timestamp();
 }
 
-}  // namespace content
+}  // namespace blink

content/renderer/media/webrtc/rtc_rtp_source.h → third_party/blink/renderer/platform/peerconnection/rtc_rtp_source.h

@@ -2,23 +2,22 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_RTC_RTP_SOURCE_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_RTC_RTP_SOURCE_H_
+#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_SOURCE_H_
+#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_SOURCE_H_
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
-#include "content/common/content_export.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_source.h"
 #include "third_party/webrtc/api/rtp_receiver_interface.h"
 
-namespace content {
+namespace blink {
 
-class CONTENT_EXPORT RTCRtpSource : public blink::WebRTCRtpSource {
+class RTCRtpSource : public WebRTCRtpSource {
  public:
   explicit RTCRtpSource(const webrtc::RtpSource& source);
   ~RTCRtpSource() override;
 
-  blink::WebRTCRtpSource::Type SourceType() const override;
+  WebRTCRtpSource::Type SourceType() const override;
   base::TimeTicks Timestamp() const override;
   uint32_t Source() const override;
   base::Optional<double> AudioLevel() const override;
@@ -30,6 +29,6 @@ class CONTENT_EXPORT RTCRtpSource : public blink::WebRTCRtpSource {
   DISALLOW_COPY_AND_ASSIGN(RTCRtpSource);
 };
 
-}  // namespace content
+}  // namespace blink
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_RTC_RTP_SOURCE_H_
+#endif  // THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_SOURCE_H_