Adding MockVoipEngine for downstream project's tests

Bug: webrtc:11989
Change-Id: Ie9cfe11a0c2b041457de66c3e3a6cdcd6179e4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201900
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33093}

BUILD.gn

@@ -539,6 +539,7 @@ if (rtc_include_tests) {
       "api/transport:stun_unittest",
       "api/video/test:rtc_api_video_unittests",
       "api/video_codecs/test:video_codecs_api_unittests",
+      "api/voip:compile_all_headers",
       "call:fake_network_pipe_unittests",
       "p2p:libstunprober_unittests",
       "p2p:rtc_p2p_unittests",
@@ -698,6 +699,7 @@ if (rtc_include_tests) {
   rtc_test("voip_unittests") {
     testonly = true
     deps = [
+      "api/voip:compile_all_headers",
       "api/voip:voip_engine_factory_unittests",
       "audio/voip/test:audio_channel_unittests",
       "audio/voip/test:audio_egress_unittests",

api/voip/BUILD.gn

@@ -49,9 +49,21 @@ rtc_library("voip_engine_factory") {
 }
 
 if (rtc_include_tests) {
+  rtc_source_set("mock_voip_engine") {
+    testonly = true
+    visibility = [ "*" ]
+    sources = [ "test/mock_voip_engine.h" ]
+    deps = [
+      ":voip_api",
+      "..:array_view",
+      "../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
   rtc_library("voip_engine_factory_unittests") {
     testonly = true
-    sources = [ "voip_engine_factory_unittest.cc" ]
+    sources = [ "test/voip_engine_factory_unittest.cc" ]
     deps = [
       ":voip_engine_factory",
       "../../modules/audio_device:mock_audio_device",
@@ -61,4 +73,13 @@ if (rtc_include_tests) {
       "../task_queue:default_task_queue_factory",
     ]
   }
+
+  rtc_library("compile_all_headers") {
+    testonly = true
+    sources = [ "test/compile_all_headers.cc" ]
+    deps = [
+      ":mock_voip_engine",
+      "../../test:test_support",
+    ]
+  }
 }

api/voip/test/compile_all_headers.cc

@@ -0,0 +1,14 @@
+/*
+ *  Copyright (c) 2021 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.
+ */
+
+// This file verifies that all include files in this directory can be
+// compiled without errors or other required includes.
+
+#include "api/voip/test/mock_voip_engine.h"

api/voip/test/mock_voip_engine.h

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2021 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 API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
+#define API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
+
+#include <map>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_codec.h"
+#include "api/voip/voip_dtmf.h"
+#include "api/voip/voip_engine.h"
+#include "api/voip/voip_network.h"
+#include "api/voip/voip_statistics.h"
+#include "api/voip/voip_volume_control.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockVoipBase : public VoipBase {
+ public:
+  MOCK_METHOD(ChannelId,
+              CreateChannel,
+              (Transport*, absl::optional<uint32_t>),
+              (override));
+  MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StopSend, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StartPlayout, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StopPlayout, (ChannelId), (override));
+};
+
+class MockVoipCodec : public VoipCodec {
+ public:
+  MOCK_METHOD(VoipResult,
+              SetSendCodec,
+              (ChannelId, int, const SdpAudioFormat&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              SetReceiveCodecs,
+              (ChannelId, (const std::map<int, SdpAudioFormat>&)),
+              (override));
+};
+
+class MockVoipDtmf : public VoipDtmf {
+ public:
+  MOCK_METHOD(VoipResult,
+              RegisterTelephoneEventType,
+              (ChannelId, int, int),
+              (override));
+  MOCK_METHOD(VoipResult,
+              SendDtmfEvent,
+              (ChannelId, DtmfEvent, int),
+              (override));
+};
+
+class MockVoipNetwork : public VoipNetwork {
+ public:
+  MOCK_METHOD(VoipResult,
+              ReceivedRTPPacket,
+              (ChannelId channel_id, rtc::ArrayView<const uint8_t> rtp_packet),
+              (override));
+  MOCK_METHOD(VoipResult,
+              ReceivedRTCPPacket,
+              (ChannelId channel_id, rtc::ArrayView<const uint8_t> rtcp_packet),
+              (override));
+};
+
+class MockVoipStatistics : public VoipStatistics {
+ public:
+  MOCK_METHOD(VoipResult,
+              GetIngressStatistics,
+              (ChannelId, IngressStatistics&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              GetChannelStatistics,
+              (ChannelId channel_id, ChannelStatistics&),
+              (override));
+};
+
+class MockVoipVolumeControl : public VoipVolumeControl {
+ public:
+  MOCK_METHOD(VoipResult, SetInputMuted, (ChannelId, bool), (override));
+
+  MOCK_METHOD(VoipResult,
+              GetInputVolumeInfo,
+              (ChannelId, VolumeInfo&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              GetOutputVolumeInfo,
+              (ChannelId, VolumeInfo&),
+              (override));
+};
+
+class MockVoipEngine : public VoipEngine {
+ public:
+  VoipBase& Base() override { return base_; }
+  VoipNetwork& Network() override { return network_; }
+  VoipCodec& Codec() override { return codec_; }
+  VoipDtmf& Dtmf() override { return dtmf_; }
+  VoipStatistics& Statistics() override { return statistics_; }
+  VoipVolumeControl& VolumeControl() override { return volume_; }
+
+  // Direct access to underlying members are required for testing.
+  MockVoipBase base_;
+  MockVoipNetwork network_;
+  MockVoipCodec codec_;
+  MockVoipDtmf dtmf_;
+  MockVoipStatistics statistics_;
+  MockVoipVolumeControl volume_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VOIP_TEST_MOCK_VOIP_ENGINE_H_