From e2173d9f0d2d991d576ffa71a18761af69b72ede Mon Sep 17 00:00:00 2001
From: eladalon <eladalon@webrtc.org>
Date: Fri, 28 Jul 2017 10:05:45 -0700
Subject: [PATCH] Only one implementation of MockRtpPacketSink once

MockRtpPacketSink has three identical implementations now, so time to move it to its own file.

BUG=None

Review-Url: https://codereview.webrtc.org/2988853002
Cr-Commit-Position: refs/heads/master@{#19183}
---
 webrtc/call/BUILD.gn                          | 17 +++++++++++-
 webrtc/call/rtp_demuxer_unittest.cc           |  7 +----
 webrtc/call/rtx_receive_stream_unittest.cc    |  6 +----
 .../test/mock_rtp_packet_sink_interface.h     | 26 +++++++++++++++++++
 webrtc/video/BUILD.gn                         |  1 +
 5 files changed, 45 insertions(+), 12 deletions(-)
 create mode 100644 webrtc/call/test/mock_rtp_packet_sink_interface.h

diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index c40b557d4c..62b8ac465f 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -34,7 +34,7 @@ rtc_source_set("call_interfaces") {
 }
 
 # TODO(nisse): These RTP targets should be moved elsewhere
-# when interfaces have stabilized.
+# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
 rtc_source_set("rtp_interfaces") {
   sources = [
     "rtcp_packet_sink_interface.h",
@@ -144,6 +144,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":call",
+      ":mock_rtp_interfaces",
       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",
@@ -213,4 +214,18 @@ if (rtc_include_tests) {
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
+
+  # TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
+  rtc_source_set("mock_rtp_interfaces") {
+    testonly = true
+
+    sources = [
+      "test/mock_rtp_packet_sink_interface.h",
+    ]
+    deps = [
+      ":rtp_interfaces",
+      "../test:test_support",
+      "//testing/gmock",
+    ]
+  }
 }
diff --git a/webrtc/call/rtp_demuxer_unittest.cc b/webrtc/call/rtp_demuxer_unittest.cc
index 8c2eab11bf..a2b4578a71 100644
--- a/webrtc/call/rtp_demuxer_unittest.cc
+++ b/webrtc/call/rtp_demuxer_unittest.cc
@@ -14,7 +14,7 @@
 #include <string>
 
 #include "webrtc/call/rsid_resolution_observer.h"
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "webrtc/call/test/mock_rtp_packet_sink_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -35,11 +35,6 @@ using ::testing::AtLeast;
 using ::testing::InSequence;
 using ::testing::NiceMock;
 
-class MockRtpPacketSink : public RtpPacketSinkInterface {
- public:
-  MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
-};
-
 class MockRsidResolutionObserver : public RsidResolutionObserver {
  public:
   MOCK_METHOD2(OnRsidResolved, void(const std::string& rsid, uint32_t ssrc));
diff --git a/webrtc/call/rtx_receive_stream_unittest.cc b/webrtc/call/rtx_receive_stream_unittest.cc
index 91ed2ca2c7..e9c8210521 100644
--- a/webrtc/call/rtx_receive_stream_unittest.cc
+++ b/webrtc/call/rtx_receive_stream_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/call/test/mock_rtp_packet_sink_interface.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -22,11 +23,6 @@ namespace {
 using ::testing::_;
 using ::testing::StrictMock;
 
-class MockRtpPacketSink : public RtpPacketSinkInterface {
- public:
-  MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
-};
-
 constexpr int kMediaPayloadType = 100;
 constexpr int kRtxPayloadType = 98;
 constexpr uint32_t kMediaSSRC = 0x3333333;
diff --git a/webrtc/call/test/mock_rtp_packet_sink_interface.h b/webrtc/call/test/mock_rtp_packet_sink_interface.h
new file mode 100644
index 0000000000..a13900436a
--- /dev/null
+++ b/webrtc/call/test/mock_rtp_packet_sink_interface.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+#ifndef WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
+#define WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
+
+#include "webrtc/call/rtp_packet_sink_interface.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockRtpPacketSink : public RtpPacketSinkInterface {
+ public:
+  MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 77ba8e92e1..f9c3453d59 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -265,6 +265,7 @@ if (rtc_include_tests) {
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
+      "../call:mock_rtp_interfaces",
       "../call:rtp_receiver",
       "../common_video",
       "../logging:rtc_event_log_api",