- Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.
BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1319163002 .
Cr-Commit-Position: refs/heads/master@{#9817}
diff --git a/talk/media/base/videoengine_unittest.h b/talk/media/base/videoengine_unittest.h
index 8fc3729..5f00a0e 100644
--- a/talk/media/base/videoengine_unittest.h
+++ b/talk/media/base/videoengine_unittest.h
@@ -40,10 +40,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/timeutils.h"
-#ifdef WIN32
-#include <objbase.h> // NOLINT
-#endif
-
#define EXPECT_FRAME_WAIT(c, w, h, t) \
EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
EXPECT_EQ((w), renderer_.width()); \
@@ -139,28 +135,6 @@
engine_.Terminate();
}
-#ifdef WIN32
- // Tests that the COM reference count is not munged by the engine.
- // Test to make sure LMI does not munge the CoInitialize reference count.
- void CheckCoInitialize() {
- // Initial refcount should be 0.
- EXPECT_EQ(S_OK, CoInitializeEx(NULL, COINIT_MULTITHREADED));
-
- // Engine should start even with COM already inited.
- EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
- engine_.Terminate();
- // Refcount after terminate should be 1; this tests if it is nonzero.
- EXPECT_EQ(S_FALSE, CoInitializeEx(NULL, COINIT_MULTITHREADED));
- // Decrement refcount to (hopefully) 0.
- CoUninitialize();
- CoUninitialize();
-
- // Ensure refcount is 0.
- EXPECT_EQ(S_OK, CoInitializeEx(NULL, COINIT_MULTITHREADED));
- CoUninitialize();
- }
-#endif
-
void ConstrainNewCodecBody() {
cricket::VideoCodec empty, in, out;
cricket::VideoCodec max_settings(engine_.codecs()[0].id,
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 545520f..257c548 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -753,16 +753,6 @@
#define WEBRTC_DISABLED_BASE_TEST(test) \
TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_##test) { Base::test(); }
-// TODO(pbos): Fix WebRtcVideoEngine2BaseTest, where we want CheckCoInitialize.
-#if 0
-// TODO(juberti): Figure out why ViE is munging the COM refcount.
-#ifdef WIN32
-WEBRTC_DISABLED_BASE_TEST(CheckCoInitialize) {
- Base::CheckCoInitialize();
-}
-#endif
-#endif
-
WEBRTC_BASE_TEST(SetSend);
WEBRTC_BASE_TEST(SetSendWithoutCodecs);
WEBRTC_BASE_TEST(SetSendSetsTransportBufferSizes);
@@ -2785,6 +2775,151 @@
WebRtcVideoEngine2 engine_;
};
+// Test that if we add a stream with RTX SSRC's, SSRC's get set correctly.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestStreamWithRtx) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that if we get too few ssrcs are given in AddSendStream(),
+// only supported sub-streams will be added.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TooFewSimulcastSsrcs) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that even more than enough ssrcs are given in AddSendStream(),
+// only supported sub-streams will be added.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_MoreThanEnoughSimulcastSscrs) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that SetSendStreamFormat works well with simulcast.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_SetSendStreamFormatWithSimulcast) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that simulcast send codec is reset on new video frame size.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_ResetSimulcastSendCodecOnNewFrameSize) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that simulcast send codec is reset on new portait mode video frame.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_ResetSimulcastSendCodecOnNewPortaitFrame) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_SetBandwidthInConferenceWithSimulcast) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that sending screencast frames in conference mode changes
+// bitrate.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_SetBandwidthScreencastInConference) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test AddSendStream with simulcast rejects bad StreamParams.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_AddSendStreamWithBadStreamParams) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test AddSendStream with simulcast sets ssrc and cname correctly.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_AddSendStreamWithSimulcast) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test RemoveSendStream with simulcast.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_RemoveSendStreamWithSimulcast) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test AddSendStream after send codec has already been set will reset
+// send codec with simulcast settings.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_AddSimulcastStreamAfterSetSendCodec) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_GetStatsWithMultipleSsrcs) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test receiving channel(s) local ssrc is set to the same as the first
+// simulcast sending ssrc.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_AddSimulcastStreamAfterCreatingRecvChannels) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test 1:1 call never turn on simulcast.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_NoSimulcastWith1on1) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test SetOptions with OPT_CONFERENCE flag.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_SetOptionsWithConferenceMode) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that two different streams can have different formats.
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_MultipleSendStreamsDifferentFormats) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestAdaptToOutputFormat) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_TestAdaptWithCpuOveruseObserver) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+// Test that codec is not reset for every frame sent in non-conference and
+// non-screencast mode.
+TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_DontResetCodecOnSendFrame) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_UseSimulcastAdapterOnVp8OnlyFactory) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
+TEST_F(WebRtcVideoEngine2SimulcastTest,
+ DISABLED_DontUseSimulcastAdapterOnNonVp8Factory) {
+ // TODO(pbos): Implement.
+ FAIL() << "Not implemented.";
+}
+
class WebRtcVideoChannel2SimulcastTest : public WebRtcVideoEngine2SimulcastTest,
public WebRtcCallFactory {
public:
@@ -2968,151 +3103,6 @@
VerifySimulcastSettings(codec, VideoOptions::NORMAL, 2, 2, SBM_NORMAL);
}
-// Test that if we add a stream with RTX SSRC's, SSRC's get set correctly.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestStreamWithRtx) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that if we get too few ssrcs are given in AddSendStream(),
-// only supported sub-streams will be added.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TooFewSimulcastSsrcs) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that even more than enough ssrcs are given in AddSendStream(),
-// only supported sub-streams will be added.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_MoreThanEnoughSimulcastSscrs) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that SetSendStreamFormat works well with simulcast.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_SetSendStreamFormatWithSimulcast) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that simulcast send codec is reset on new video frame size.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_ResetSimulcastSendCodecOnNewFrameSize) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that simulcast send codec is reset on new portait mode video frame.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_ResetSimulcastSendCodecOnNewPortaitFrame) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_SetBandwidthInConferenceWithSimulcast) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that sending screencast frames in conference mode changes
-// bitrate.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_SetBandwidthScreencastInConference) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test AddSendStream with simulcast rejects bad StreamParams.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_AddSendStreamWithBadStreamParams) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test AddSendStream with simulcast sets ssrc and cname correctly.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_AddSendStreamWithSimulcast) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test RemoveSendStream with simulcast.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_RemoveSendStreamWithSimulcast) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test AddSendStream after send codec has already been set will reset
-// send codec with simulcast settings.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_AddSimulcastStreamAfterSetSendCodec) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_GetStatsWithMultipleSsrcs) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test receiving channel(s) local ssrc is set to the same as the first
-// simulcast sending ssrc.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_AddSimulcastStreamAfterCreatingRecvChannels) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test 1:1 call never turn on simulcast.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_NoSimulcastWith1on1) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test SetOptions with OPT_CONFERENCE flag.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_SetOptionsWithConferenceMode) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that two different streams can have different formats.
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_MultipleSendStreamsDifferentFormats) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestAdaptToOutputFormat) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_TestAdaptWithCpuOveruseObserver) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-// Test that codec is not reset for every frame sent in non-conference and
-// non-screencast mode.
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_DontResetCodecOnSendFrame) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_UseSimulcastAdapterOnVp8OnlyFactory) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest,
- DISABLED_DontUseSimulcastAdapterOnNonVp8Factory) {
- // TODO(pbos): Implement.
- FAIL() << "Not implemented.";
-}
-
TEST_F(WebRtcVideoChannel2SimulcastTest, DISABLED_SimulcastSend_1280x800) {
// TODO(pbos): Implement.
FAIL() << "Not implemented.";
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 391052b..f8c3ac3 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -54,10 +54,6 @@
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#ifdef WIN32
-#include <objbase.h> // NOLINT
-#endif
-
namespace cricket {
static const int kMaxNumPacketSize = 6;
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index ef87c74..1daf58d 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -25,11 +25,6 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifdef WIN32
-#include "webrtc/base/win32.h"
-#include <objbase.h>
-#endif
-
#include "webrtc/base/byteorder.h"
#include "webrtc/base/gunit.h"
#include "talk/media/base/constants.h"
@@ -3135,16 +3130,6 @@
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
-TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
- cricket::WebRtcVoiceEngine engine;
- cricket::AudioOptions options = engine.GetOptions();
- // The default options should have at least a few things set. We purposefully
- // don't check the option values here, though.
- EXPECT_TRUE(options.echo_cancellation.IsSet());
- EXPECT_TRUE(options.auto_gain_control.IsSet());
- EXPECT_TRUE(options.noise_suppression.IsSet());
-}
-
// Test that GetReceiveChannelNum returns the default channel for the first
// recv stream in 1-1 calls.
TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelNumIn1To1Calls) {
@@ -3194,165 +3179,6 @@
EXPECT_DOUBLE_EQ(1, right);
}
-// Tests for the actual WebRtc VoE library.
-
-// Tests that the library initializes and shuts down properly.
-TEST(WebRtcVoiceEngineTest, StartupShutdown) {
- cricket::WebRtcVoiceEngine engine;
- EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
- cricket::VoiceMediaChannel* channel =
- engine.CreateChannel(cricket::AudioOptions());
- EXPECT_TRUE(channel != nullptr);
- delete channel;
- engine.Terminate();
-
- // Reinit to catch regression where VoiceEngineObserver reference is lost
- EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
- engine.Terminate();
-}
-
-// Tests that the library is configured with the codecs we want.
-TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
- cricket::WebRtcVoiceEngine engine;
- // Check codecs by name.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "OPUS", 48000, 0, 2, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ISAC", 16000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ISAC", 32000, 0, 1, 0)));
- // Check that name matching is case-insensitive.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ILBC", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "iLBC", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "PCMU", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "G722", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 32000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 16000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "telephone-event", 8000, 0, 1, 0)));
- // Check codecs with an id by id.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(0, "", 8000, 0, 1, 0))); // PCMU
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(8, "", 8000, 0, 1, 0))); // PCMA
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(9, "", 8000, 0, 1, 0))); // G722
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(13, "", 8000, 0, 1, 0))); // CN
- // Check sample/bitrate matching.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 0)));
- // Check that bad codecs fail.
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(99, "ABCD", 0, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(88, "", 0, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 0, 2, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 5000, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 5000, 1, 0)));
- // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
- for (std::vector<cricket::AudioCodec>::const_iterator it =
- engine.codecs().begin(); it != engine.codecs().end(); ++it) {
- if (it->name == "CN" && it->clockrate == 16000) {
- EXPECT_EQ(105, it->id);
- } else if (it->name == "CN" && it->clockrate == 32000) {
- EXPECT_EQ(106, it->id);
- } else if (it->name == "ISAC" && it->clockrate == 16000) {
- EXPECT_EQ(103, it->id);
- } else if (it->name == "ISAC" && it->clockrate == 32000) {
- EXPECT_EQ(104, it->id);
- } else if (it->name == "G722" && it->clockrate == 8000) {
- EXPECT_EQ(9, it->id);
- } else if (it->name == "telephone-event") {
- EXPECT_EQ(126, it->id);
- } else if (it->name == "red") {
- EXPECT_EQ(127, it->id);
- } else if (it->name == "opus") {
- EXPECT_EQ(111, it->id);
- ASSERT_TRUE(it->params.find("minptime") != it->params.end());
- EXPECT_EQ("10", it->params.find("minptime")->second);
- ASSERT_TRUE(it->params.find("maxptime") != it->params.end());
- EXPECT_EQ("60", it->params.find("maxptime")->second);
- ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
- EXPECT_EQ("1", it->params.find("useinbandfec")->second);
- }
- }
-
- engine.Terminate();
-}
-
-// Tests that VoE supports at least 32 channels
-TEST(WebRtcVoiceEngineTest, Has32Channels) {
- cricket::WebRtcVoiceEngine engine;
- EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
-
- cricket::VoiceMediaChannel* channels[32];
- int num_channels = 0;
-
- while (num_channels < ARRAY_SIZE(channels)) {
- cricket::VoiceMediaChannel* channel =
- engine.CreateChannel(cricket::AudioOptions());
- if (!channel)
- break;
-
- channels[num_channels++] = channel;
- }
-
- int expected = ARRAY_SIZE(channels);
- EXPECT_EQ(expected, num_channels);
-
- while (num_channels > 0) {
- delete channels[--num_channels];
- }
-
- engine.Terminate();
-}
-
-// Test that we set our preferred codecs properly.
-TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
- cricket::WebRtcVoiceEngine engine;
- EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
- cricket::WebRtcVoiceMediaChannel channel(&engine);
- EXPECT_TRUE(channel.SetRecvCodecs(engine.codecs()));
-}
-
-#ifdef WIN32
-// Test our workarounds to WebRtc VoE' munging of the coinit count
-TEST(WebRtcVoiceEngineTest, CoInitialize) {
- cricket::WebRtcVoiceEngine* engine = new cricket::WebRtcVoiceEngine();
-
- // Initial refcount should be 0.
- EXPECT_EQ(S_OK, CoInitializeEx(NULL, COINIT_MULTITHREADED));
-
- // Engine should start even with COM already inited.
- EXPECT_TRUE(engine->Init(rtc::Thread::Current()));
- engine->Terminate();
- EXPECT_TRUE(engine->Init(rtc::Thread::Current()));
- engine->Terminate();
-
- // Refcount after terminate should be 1 (in reality 3); test if it is nonzero.
- EXPECT_EQ(S_FALSE, CoInitializeEx(NULL, COINIT_MULTITHREADED));
- // Decrement refcount to (hopefully) 0.
- CoUninitialize();
- CoUninitialize();
- delete engine;
-
- // Ensure refcount is 0.
- EXPECT_EQ(S_OK, CoInitializeEx(NULL, COINIT_MULTITHREADED));
- CoUninitialize();
-}
-#endif
-
TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) {
cricket::FakeCall call((webrtc::Call::Config()));
const uint32 kAudioSsrc = 123;
@@ -3663,3 +3489,145 @@
EXPECT_TRUE(channel_->RemoveSendStream(2));
EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), -1);
}
+
+// Tests for the actual WebRtc VoE library.
+
+TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
+ cricket::WebRtcVoiceEngine engine;
+ cricket::AudioOptions options = engine.GetOptions();
+ // The default options should have at least a few things set. We purposefully
+ // don't check the option values here, though.
+ EXPECT_TRUE(options.echo_cancellation.IsSet());
+ EXPECT_TRUE(options.auto_gain_control.IsSet());
+ EXPECT_TRUE(options.noise_suppression.IsSet());
+}
+
+// Tests that the library initializes and shuts down properly.
+TEST(WebRtcVoiceEngineTest, StartupShutdown) {
+ cricket::WebRtcVoiceEngine engine;
+ EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
+ cricket::VoiceMediaChannel* channel =
+ engine.CreateChannel(cricket::AudioOptions());
+ EXPECT_TRUE(channel != nullptr);
+ delete channel;
+ engine.Terminate();
+
+ // Reinit to catch regression where VoiceEngineObserver reference is lost
+ EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
+ engine.Terminate();
+}
+
+// Tests that the library is configured with the codecs we want.
+TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
+ cricket::WebRtcVoiceEngine engine;
+ // Check codecs by name.
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "OPUS", 48000, 0, 2, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "ISAC", 16000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "ISAC", 32000, 0, 1, 0)));
+ // Check that name matching is case-insensitive.
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "ILBC", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "iLBC", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "PCMU", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "G722", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "CN", 32000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "CN", 16000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "CN", 8000, 0, 1, 0)));
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(96, "telephone-event", 8000, 0, 1, 0)));
+ // Check codecs with an id by id.
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(0, "", 8000, 0, 1, 0))); // PCMU
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(8, "", 8000, 0, 1, 0))); // PCMA
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(9, "", 8000, 0, 1, 0))); // G722
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(13, "", 8000, 0, 1, 0))); // CN
+ // Check sample/bitrate matching.
+ EXPECT_TRUE(engine.FindCodec(
+ cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 0)));
+ // Check that bad codecs fail.
+ EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(99, "ABCD", 0, 0, 1, 0)));
+ EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(88, "", 0, 0, 1, 0)));
+ EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 0, 2, 0)));
+ EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 5000, 0, 1, 0)));
+ EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 5000, 1, 0)));
+ // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
+ for (std::vector<cricket::AudioCodec>::const_iterator it =
+ engine.codecs().begin(); it != engine.codecs().end(); ++it) {
+ if (it->name == "CN" && it->clockrate == 16000) {
+ EXPECT_EQ(105, it->id);
+ } else if (it->name == "CN" && it->clockrate == 32000) {
+ EXPECT_EQ(106, it->id);
+ } else if (it->name == "ISAC" && it->clockrate == 16000) {
+ EXPECT_EQ(103, it->id);
+ } else if (it->name == "ISAC" && it->clockrate == 32000) {
+ EXPECT_EQ(104, it->id);
+ } else if (it->name == "G722" && it->clockrate == 8000) {
+ EXPECT_EQ(9, it->id);
+ } else if (it->name == "telephone-event") {
+ EXPECT_EQ(126, it->id);
+ } else if (it->name == "red") {
+ EXPECT_EQ(127, it->id);
+ } else if (it->name == "opus") {
+ EXPECT_EQ(111, it->id);
+ ASSERT_TRUE(it->params.find("minptime") != it->params.end());
+ EXPECT_EQ("10", it->params.find("minptime")->second);
+ ASSERT_TRUE(it->params.find("maxptime") != it->params.end());
+ EXPECT_EQ("60", it->params.find("maxptime")->second);
+ ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
+ EXPECT_EQ("1", it->params.find("useinbandfec")->second);
+ }
+ }
+
+ engine.Terminate();
+}
+
+// Tests that VoE supports at least 32 channels
+TEST(WebRtcVoiceEngineTest, Has32Channels) {
+ cricket::WebRtcVoiceEngine engine;
+ EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
+
+ cricket::VoiceMediaChannel* channels[32];
+ int num_channels = 0;
+
+ while (num_channels < ARRAY_SIZE(channels)) {
+ cricket::VoiceMediaChannel* channel =
+ engine.CreateChannel(cricket::AudioOptions());
+ if (!channel)
+ break;
+
+ channels[num_channels++] = channel;
+ }
+
+ int expected = ARRAY_SIZE(channels);
+ EXPECT_EQ(expected, num_channels);
+
+ while (num_channels > 0) {
+ delete channels[--num_channels];
+ }
+
+ engine.Terminate();
+}
+
+// Test that we set our preferred codecs properly.
+TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
+ cricket::WebRtcVoiceEngine engine;
+ EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
+ cricket::WebRtcVoiceMediaChannel channel(&engine);
+ EXPECT_TRUE(channel.SetRecvCodecs(engine.codecs()));
+}
