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- Move test cases for more natural ordering.

Browse Source 
- 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}
This commit is contained in:
Fredrik Solenberg
2015-08-31 11:13:51 +02:00
parent 3c4ef29140
commit c20a5dc930
4 changed files with 287 additions and 359 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
talk/media/base/videoengine_unittest.h
View File

@ -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 @@ class VideoEngineTest : public testing::Test {
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,

300
talk/media/webrtc/webrtcvideoengine2_unittest.cc
View File

@ -753,16 +753,6 @@ class WebRtcVideoChannel2BaseTest
#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 @@ class WebRtcVideoEngine2SimulcastTest : public testing::Test {
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 @@ TEST_F(WebRtcVideoChannel2SimulcastTest, SetSendCodecsWithOddSizeInSimulcast) {
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.";

4
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -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;

316
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -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 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOutputScaling) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, AssociateChannelResetUponDeleteChannnel) {
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()));
}