// Tests for the WebRtcVoiceEngine/VoiceChannel code.
+using cricket::kRtpAudioLevelHeaderExtension;
+using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
+
static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
static const cricket::AudioCodec kCeltCodec(110, "CELT", 32000, 64000, 2, 0);
options_conference_.conference_mode.Set(true);
options_adjust_agc_.adjust_agc_delta.Set(-10);
}
- bool SetupEngine() {
- bool result = engine_.Init(talk_base::Thread::Current());
- if (result) {
- channel_ = engine_.CreateChannel();
- result = (channel_ != NULL);
+ bool SetupEngineWithoutStream() {
+ if (!engine_.Init(talk_base::Thread::Current())) {
+ return false;
}
- if (result) {
- result = channel_->AddSendStream(
- cricket::StreamParams::CreateLegacy(kSsrc1));
+ channel_ = engine_.CreateChannel();
+ return (channel_ != NULL);
+ }
+ bool SetupEngine() {
+ if (!SetupEngineWithoutStream()) {
+ return false;
}
- return result;
+ return channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrc1));
}
void SetupForMultiSendStream() {
EXPECT_TRUE(SetupEngine());
EXPECT_EQ(expected_bitrate, temp_codec.rate);
}
-
- void TestSetSendRtpHeaderExtensions(int channel_id) {
- std::vector<cricket::RtpHeaderExtension> extensions;
+ void TestSetSendRtpHeaderExtensions(const std::string& ext) {
+ EXPECT_TRUE(SetupEngineWithoutStream());
+ int channel_num = voe_.GetLastChannel();
// Ensure extensions are off by default.
- EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
+ std::vector<cricket::RtpHeaderExtension> extensions;
// Ensure unknown extensions won't cause an error.
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
// Ensure extensions stay off with an empty list of headers.
extensions.clear();
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
- // Ensure audio levels are enabled if the audio-level header is specified
- // (but AST is still off).
- extensions.push_back(cricket::RtpHeaderExtension(
- "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
+ // Ensure extension is set properly.
+ const int id = 1;
+ extensions.push_back(cricket::RtpHeaderExtension(ext, id));
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(id, voe_.GetSendRtpExtensionId(channel_num, ext));
-#ifdef USE_WEBRTC_DEV_BRANCH
- // Ensure audio level and AST are enabled if the extensions are specified.
- extensions.push_back(cricket::RtpHeaderExtension(
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 12));
- EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
- EXPECT_EQ(12, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ // Ensure extension is set properly on new channel.
+ // The first stream to occupy the default channel.
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(123)));
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(234)));
+ int new_channel_num = voe_.GetLastChannel();
+ EXPECT_NE(channel_num, new_channel_num);
+ EXPECT_EQ(id, voe_.GetSendRtpExtensionId(new_channel_num, ext));
// Ensure all extensions go back off with an empty list.
extensions.clear();
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(new_channel_num, ext));
}
- void TestSetRecvRtpHeaderExtensions(int channel_id) {
- std::vector<cricket::RtpHeaderExtension> extensions;
+ void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
+ EXPECT_TRUE(SetupEngineWithoutStream());
+ int channel_num = voe_.GetLastChannel();
-#ifdef USE_WEBRTC_DEV_BRANCH
// Ensure extensions are off by default.
- EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ std::vector<cricket::RtpHeaderExtension> extensions;
// Ensure unknown extensions won't cause an error.
extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
- // An empty list shouldn't cause any headers to be enabled.
+ // Ensure extensions stay off with an empty list of headers.
extensions.clear();
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
-#ifdef USE_WEBRTC_DEV_BRANCH
- // Nor should indicating we can receive the absolute sender time header.
- extensions.push_back(cricket::RtpHeaderExtension(
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 11));
+ // Ensure extension is set properly.
+ const int id = 2;
+ extensions.push_back(cricket::RtpHeaderExtension(ext, id));
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
- EXPECT_EQ(11, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(channel_num, ext));
- // Resetting to an empty list shouldn't cause any headers to be enabled.
+ // Ensure extension is set properly on new channel.
+ // The first stream to occupy the default channel.
+ EXPECT_TRUE(channel_->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(345)));
+ EXPECT_TRUE(channel_->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(456)));
+ int new_channel_num = voe_.GetLastChannel();
+ EXPECT_NE(channel_num, new_channel_num);
+ EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
+
+ // Ensure all extensions go back off with an empty list.
extensions.clear();
EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-#ifdef USE_WEBRTC_DEV_BRANCH
- EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
-#endif
+ EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
}
protected:
codecs[0].id = 96;
codecs[0].bitrate = 48000;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(1, voe_.GetNumSetSendCodecs());
webrtc::CodecInst gcodec;
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(96, gcodec.pltype);
EXPECT_EQ(106, voe_.GetSendTelephoneEventPayloadType(channel_num));
}
+// Test that VoE Channel doesn't call SetSendCodec again if same codec is tried
+// to apply.
+TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) {
+ EXPECT_TRUE(SetupEngine());
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kIsacCodec);
+ codecs.push_back(kPcmuCodec);
+ codecs.push_back(kRedCodec);
+ codecs[0].id = 96;
+ codecs[0].bitrate = 48000;
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(1, voe_.GetNumSetSendCodecs());
+ // Calling SetSendCodec again with same codec which is already set.
+ // In this case media channel shouldn't send codec to VoE.
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(1, voe_.GetNumSetSendCodecs());
+}
+
// Test that if clockrate is not 48000 for opus, we fail.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
EXPECT_TRUE(SetupEngine());
EXPECT_FALSE(voe_.GetFEC(channel_num));
}
-// Test that we support setting certain send header extensions.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendRtpHeaderExtensions) {
- EXPECT_TRUE(SetupEngine());
- TestSetSendRtpHeaderExtensions(voe_.GetLastChannel());
+// Test support for audio level header extension.
+TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
}
+#ifdef USE_WEBRTC_DEV_BRANCH
+TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
+}
+#endif // USE_WEBRTC_DEV_BRANCH
-// Test that we support setting recv header extensions.
-TEST_F(WebRtcVoiceEngineTestFake, SetRecvRtpHeaderExtensions) {
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- TestSetRecvRtpHeaderExtensions(voe_.GetLastChannel());
+// Test support for absolute send time header extension.
+TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
+}
+TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension);
}
// Test that we can create a channel and start sending/playing out on it.
EXPECT_EQ(1u, info.receivers.size());
}
-// Test that we support setting header extensions on multiple send streams.
-TEST_F(WebRtcVoiceEngineTestFake,
- SetSendRtpHeaderExtensionsWithMultipleSendStreams) {
- SetupForMultiSendStream();
-
- static const uint32 kSsrcs[] = {1, 2, 3, 4};
- for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
- EXPECT_TRUE(channel_->AddSendStream(
- cricket::StreamParams::CreateLegacy(kSsrcs[i])));
- }
-
- for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
- int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs[i]);
- TestSetSendRtpHeaderExtensions(channel_num);
- }
-}
-
-// Test that we support setting header extensions on multiple receive streams.
-TEST_F(WebRtcVoiceEngineTestFake,
- SetRecvRtpHeaderExtensionsWithMultipleRecvStreams) {
- EXPECT_TRUE(SetupEngine());
-
- static const uint32 kSsrcs[] = {1, 2, 3, 4};
- int channel_ids[ARRAY_SIZE(kSsrcs)] = {0};
- for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
- EXPECT_TRUE(channel_->AddRecvStream(
- cricket::StreamParams::CreateLegacy(kSsrcs[i])));
- channel_ids[i] = voe_.GetLastChannel();
- }
-
- for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
- TestSetRecvRtpHeaderExtensions(channel_ids[i]);
- }
-}
-
// Test that we can add and remove receive streams, and do proper send/playout.
// We can receive on multiple streams while sending one stream.
TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
EXPECT_EQ(set_config.limiterEnable, config.limiterEnable);
}
-
TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_TRUE(SetupEngine());
talk_base::scoped_ptr<cricket::VoiceMediaChannel> channel1(
EXPECT_DOUBLE_EQ(1, right);
}
-
// Tests for the actual WebRtc VoE library.
// Tests that the library initializes and shuts down properly.
}
#endif
-
-TEST_F(WebRtcVoiceEngineTestFake, SetExperimentalAcm) {
- EXPECT_TRUE(SetupEngine());
-
- // By default experimental ACM should not be used.
- int media_channel = engine_.CreateMediaVoiceChannel();
- ASSERT_GE(media_channel, 0);
- EXPECT_FALSE(voe_.IsUsingExperimentalAcm(media_channel));
-
- int soundclip_channel = engine_.CreateSoundclipVoiceChannel();
- ASSERT_GE(soundclip_channel, 0);
- EXPECT_FALSE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-
- // Set options to use experimental ACM.
- cricket::AudioOptions options;
- options.experimental_acm.Set(true);
- ASSERT_TRUE(engine_.SetOptions(options));
- media_channel = engine_.CreateMediaVoiceChannel();
- ASSERT_GE(media_channel, 0);
- EXPECT_TRUE(voe_.IsUsingExperimentalAcm(media_channel));
-
- soundclip_channel = engine_.CreateSoundclipVoiceChannel();
- ASSERT_GE(soundclip_channel, 0);
- EXPECT_TRUE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-
- // Set option to use legacy ACM.
- options.experimental_acm.Set(false);
- ASSERT_TRUE(engine_.SetOptions(options));
- media_channel = engine_.CreateMediaVoiceChannel();
- ASSERT_GE(media_channel, 0);
- EXPECT_FALSE(voe_.IsUsingExperimentalAcm(media_channel));
-
- soundclip_channel = engine_.CreateSoundclipVoiceChannel();
- ASSERT_GE(soundclip_channel, 0);
- EXPECT_FALSE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-}
projects / platform / framework / web / crosswalk.git / blobdiff