Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(26)

Side by Side Diff: webrtc/media/engine/webrtcvoiceengine_unittest.cc

Issue 2516993002: Pass SdpAudioFormat through Channel, without converting to CodecInst (Closed)
Patch Set: Fix SetRecvCodecsAfterAddingStreams Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 89 matching lines...) Expand 10 before | Expand all | Expand 10 after
100 void SetSink(Sink* sink) override {} 100 void SetSink(Sink* sink) override {}
101 }; 101 };
102 102
103 class WebRtcVoiceEngineTestFake : public testing::Test { 103 class WebRtcVoiceEngineTestFake : public testing::Test {
104 public: 104 public:
105 WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {} 105 WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
106 106
107 explicit WebRtcVoiceEngineTestFake(const char* field_trials) 107 explicit WebRtcVoiceEngineTestFake(const char* field_trials)
108 : call_(webrtc::Call::Config(&event_log_)), voe_(&apm_), 108 : call_(webrtc::Call::Config(&event_log_)), voe_(&apm_),
109 override_field_trials_(field_trials) { 109 override_field_trials_(field_trials) {
110 auto factory = webrtc::MockAudioDecoderFactory::CreateUnusedFactory(); 110 auto factory =
111 webrtc::MockAudioDecoderFactory::CreateUnusedPotemkinFactory();
the sun 2016/12/14 08:58:24 I guess that will be explained later, but I'm not
ossu 2016/12/14 10:21:36 I had to google it. That said, it's pretty witty.
kwiberg-webrtc 2016/12/14 13:09:36 "I had to google it" and "pretty witty"---what mor
ossu 2016/12/14 13:53:35 I think I'd just set up the mock manually in this
kwiberg-webrtc 2016/12/15 14:30:50 OK, using the normal factory and adding a TODO sou
111 EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0)); 112 EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0));
112 EXPECT_CALL(adm_, Release()).WillOnce(Return(0)); 113 EXPECT_CALL(adm_, Release()).WillOnce(Return(0));
113 EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false)); 114 EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false));
114 EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false)); 115 EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false));
115 EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false)); 116 EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false));
116 EXPECT_CALL(apm_, ApplyConfig(testing::_)); 117 EXPECT_CALL(apm_, ApplyConfig(testing::_));
117 EXPECT_CALL(apm_, SetExtraOptions(testing::_)); 118 EXPECT_CALL(apm_, SetExtraOptions(testing::_));
118 EXPECT_CALL(apm_, Initialize()).WillOnce(Return(0)); 119 EXPECT_CALL(apm_, Initialize()).WillOnce(Return(0));
119 engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, factory, 120 engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, factory,
120 new FakeVoEWrapper(&voe_))); 121 new FakeVoEWrapper(&voe_)));
(...skipping 742 matching lines...) Expand 10 before | Expand all | Expand 10 after
863 EXPECT_STREQ("telephone-event", gcodec.plname); 864 EXPECT_STREQ("telephone-event", gcodec.plname);
864 } 865 }
865 866
866 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { 867 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
867 EXPECT_TRUE(SetupRecvStream()); 868 EXPECT_TRUE(SetupRecvStream());
868 cricket::AudioRecvParameters parameters; 869 cricket::AudioRecvParameters parameters;
869 parameters.codecs.push_back(kIsacCodec); 870 parameters.codecs.push_back(kIsacCodec);
870 parameters.codecs[0].id = 106; // collide with existing CN 32k 871 parameters.codecs[0].id = 106; // collide with existing CN 32k
871 EXPECT_TRUE(channel_->SetRecvParameters(parameters)); 872 EXPECT_TRUE(channel_->SetRecvParameters(parameters));
872 873
873 int channel_num2 = voe_.GetLastChannel(); 874 const auto& dm = GetRecvStreamConfig(kSsrc1).decoder_map;
874 webrtc::CodecInst gcodec; 875 ASSERT_EQ(1, dm.count(106));
875 rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC"); 876 EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106));
the sun 2016/12/14 08:58:25 Any reason "ISAC" changed to "isac"?
kwiberg-webrtc 2016/12/14 13:09:36 I re-wrote it by hand rather than copy+pasting, an
876 gcodec.plfreq = 16000;
877 gcodec.channels = 1;
878 EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
879 EXPECT_EQ(106, gcodec.pltype);
880 EXPECT_STREQ("ISAC", gcodec.plname);
881 } 877 }
882 878
883 // Test that we can apply the same set of codecs again while playing. 879 // Test that we can apply the same set of codecs again while playing.
884 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { 880 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
885 EXPECT_TRUE(SetupRecvStream()); 881 EXPECT_TRUE(SetupRecvStream());
886 cricket::AudioRecvParameters parameters; 882 cricket::AudioRecvParameters parameters;
887 parameters.codecs.push_back(kIsacCodec); 883 parameters.codecs.push_back(kIsacCodec);
888 parameters.codecs.push_back(kCn16000Codec); 884 parameters.codecs.push_back(kCn16000Codec);
889 EXPECT_TRUE(channel_->SetRecvParameters(parameters)); 885 EXPECT_TRUE(channel_->SetRecvParameters(parameters));
890 channel_->SetPlayout(true); 886 channel_->SetPlayout(true);
(...skipping 2715 matching lines...) Expand 10 before | Expand all | Expand 10 after
3606 nullptr, webrtc::CreateBuiltinAudioDecoderFactory()); 3602 nullptr, webrtc::CreateBuiltinAudioDecoderFactory());
3607 webrtc::RtcEventLogNullImpl event_log; 3603 webrtc::RtcEventLogNullImpl event_log;
3608 std::unique_ptr<webrtc::Call> call( 3604 std::unique_ptr<webrtc::Call> call(
3609 webrtc::Call::Create(webrtc::Call::Config(&event_log))); 3605 webrtc::Call::Create(webrtc::Call::Config(&event_log)));
3610 cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(), 3606 cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
3611 cricket::AudioOptions(), call.get()); 3607 cricket::AudioOptions(), call.get());
3612 cricket::AudioRecvParameters parameters; 3608 cricket::AudioRecvParameters parameters;
3613 parameters.codecs = engine.recv_codecs(); 3609 parameters.codecs = engine.recv_codecs();
3614 EXPECT_TRUE(channel.SetRecvParameters(parameters)); 3610 EXPECT_TRUE(channel.SetRecvParameters(parameters));
3615 } 3611 }
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698