Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1362913004
Cr-Commit-Position: refs/heads/master@{#10133}
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index 4e39a2a..13ab8fa 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -240,7 +240,6 @@
explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
const AudioOptions& options)
: engine_(engine),
- fail_set_send_(false),
time_since_last_typing_(-1) {
output_scalings_[0] = OutputScaling(); // For default channel.
SetOptions(options);
@@ -270,9 +269,6 @@
return true;
}
virtual bool SetSend(SendFlags flag) {
- if (fail_set_send_) {
- return false;
- }
return set_sending(flag != SEND_NOTHING);
}
virtual bool SetAudioSend(uint32 ssrc, bool enable,
@@ -370,17 +366,6 @@
}
virtual bool GetStats(VoiceMediaInfo* info) { return false; }
- virtual void GetLastMediaError(uint32* ssrc,
- VoiceMediaChannel::Error* error) {
- *ssrc = 0;
- *error = fail_set_send_ ? VoiceMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED
- : VoiceMediaChannel::ERROR_NONE;
- }
-
- void set_fail_set_send(bool fail) { fail_set_send_ = fail; }
- void TriggerError(uint32 ssrc, VoiceMediaChannel::Error error) {
- VoiceMediaChannel::SignalMediaError(ssrc, error);
- }
private:
struct OutputScaling {
@@ -458,7 +443,6 @@
std::vector<AudioCodec> send_codecs_;
std::map<uint32, OutputScaling> output_scalings_;
std::vector<DtmfInfo> dtmf_info_queue_;
- bool fail_set_send_;
int time_since_last_typing_;
AudioOptions options_;
std::map<uint32, VoiceChannelAudioSink*> local_renderers_;
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 9ed662d..dd46a2f 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -1097,16 +1097,6 @@
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
- // Gets last reported error for this media channel.
- virtual void GetLastMediaError(uint32* ssrc,
- VoiceMediaChannel::Error* error) {
- ASSERT(error != NULL);
- *error = ERROR_NONE;
- }
-
- // Signal errors from MediaChannel. Arguments are:
- // ssrc(uint32), and error(VoiceMediaChannel::Error).
- sigslot::signal2<uint32, VoiceMediaChannel::Error> SignalMediaError;
};
struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
@@ -1160,10 +1150,6 @@
virtual bool RequestIntraFrame() = 0;
virtual void UpdateAspectRatio(int ratio_w, int ratio_h) = 0;
- // Signal errors from MediaChannel. Arguments are:
- // ssrc(uint32), and error(VideoMediaChannel::Error).
- sigslot::signal2<uint32, Error> SignalMediaError;
-
protected:
VideoRenderer *renderer_;
};
@@ -1286,9 +1272,6 @@
sigslot::signal3<const ReceiveDataParams&,
const char*,
size_t> SignalDataReceived;
- // Signal errors from MediaChannel. Arguments are:
- // ssrc(uint32), and error(DataMediaChannel::Error).
- sigslot::signal2<uint32, DataMediaChannel::Error> SignalMediaError;
// Signal when the media channel is ready to send the stream. Arguments are:
// writable(bool)
sigslot::signal1<bool> SignalReadyToSend;
diff --git a/talk/media/base/videoengine_unittest.h b/talk/media/base/videoengine_unittest.h
index 3f02055..74f6294 100644
--- a/talk/media/base/videoengine_unittest.h
+++ b/talk/media/base/videoengine_unittest.h
@@ -465,7 +465,6 @@
channel_.reset(
engine_.CreateChannel(call_.get(), cricket::VideoOptions()));
EXPECT_TRUE(channel_.get() != NULL);
- ConnectVideoChannelError();
network_interface_.SetDestination(channel_.get());
channel_->SetInterface(&network_interface_);
media_error_ = cricket::VideoMediaChannel::ERROR_NONE;
@@ -520,10 +519,6 @@
virtual void TearDown() {
channel_.reset();
}
- void ConnectVideoChannelError() {
- channel_->SignalMediaError.connect(this,
- &VideoMediaChannelTest<E, C>::OnVideoChannelError);
- }
bool SetDefaultCodec() {
return SetOneCodec(DefaultCodec());
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index ac37425..40d8442 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2970,14 +2970,6 @@
return true;
}
-void WebRtcVoiceMediaChannel::GetLastMediaError(
- uint32* ssrc, VoiceMediaChannel::Error* error) {
- RTC_DCHECK(ssrc != NULL);
- RTC_DCHECK(error != NULL);
- FindSsrc(voe_channel(), ssrc);
- *error = WebRtcErrorToChannelError(GetLastEngineError());
-}
-
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
rtc::CritScope lock(&receive_channels_cs_);
RTC_DCHECK(ssrc != NULL);
@@ -3019,7 +3011,6 @@
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
typing_noise_detected_ = false;
}
- SignalMediaError(ssrc, WebRtcErrorToChannelError(error));
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 8f8f695..7950024 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -221,10 +221,6 @@
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override {}
bool GetStats(VoiceMediaInfo* info) override;
- // Gets last reported error from WebRtc voice engine. This should be only
- // called in response a failure.
- void GetLastMediaError(uint32* ssrc,
- VoiceMediaChannel::Error* error) override;
// implements Transport interface
bool SendRtp(const uint8_t* data, size_t len) override {
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 791e7cf..0a72783 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -89,35 +89,6 @@
class WebRtcVoiceEngineTestFake : public testing::Test {
public:
- class ChannelErrorListener : public sigslot::has_slots<> {
- public:
- explicit ChannelErrorListener(cricket::VoiceMediaChannel* channel)
- : ssrc_(0), error_(cricket::VoiceMediaChannel::ERROR_NONE) {
- RTC_DCHECK(channel != NULL);
- channel->SignalMediaError.connect(
- this, &ChannelErrorListener::OnVoiceChannelError);
- }
- void OnVoiceChannelError(uint32 ssrc,
- cricket::VoiceMediaChannel::Error error) {
- ssrc_ = ssrc;
- error_ = error;
- }
- void Reset() {
- ssrc_ = 0;
- error_ = cricket::VoiceMediaChannel::ERROR_NONE;
- }
- uint32 ssrc() const {
- return ssrc_;
- }
- cricket::VoiceMediaChannel::Error error() const {
- return error_;
- }
-
- private:
- uint32 ssrc_;
- cricket::VoiceMediaChannel::Error error_;
- };
-
WebRtcVoiceEngineTestFake()
: call_(webrtc::Call::Config()),
voe_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
@@ -2543,51 +2514,6 @@
TestInsertDtmf(kSsrc1, false);
}
-TEST_F(WebRtcVoiceEngineTestFake, MediaEngineCallbackOnError) {
- rtc::scoped_ptr<ChannelErrorListener> listener;
- cricket::WebRtcVoiceMediaChannel* media_channel;
- unsigned int ssrc = 0;
-
- EXPECT_TRUE(SetupEngine());
- send_parameters_.options = options_conference_;
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
- EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
-
- media_channel = static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
- listener.reset(new ChannelErrorListener(channel_));
-
- // Test on WebRtc VoE channel.
- voe_.TriggerCallbackOnError(media_channel->voe_channel(),
- VE_SATURATION_WARNING);
- EXPECT_EQ(cricket::VoiceMediaChannel::ERROR_REC_DEVICE_SATURATION,
- listener->error());
- EXPECT_NE(-1, voe_.GetLocalSSRC(voe_.GetLastChannel(), ssrc));
- EXPECT_EQ(ssrc, listener->ssrc());
-
- listener->Reset();
- voe_.TriggerCallbackOnError(-1, VE_TYPING_NOISE_WARNING);
- EXPECT_EQ(cricket::VoiceMediaChannel::ERROR_REC_TYPING_NOISE_DETECTED,
- listener->error());
- EXPECT_EQ(0U, listener->ssrc());
-
- // Add another stream and test on that.
- ++ssrc;
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(
- ssrc)));
- listener->Reset();
- voe_.TriggerCallbackOnError(voe_.GetLastChannel(),
- VE_SATURATION_WARNING);
- EXPECT_EQ(cricket::VoiceMediaChannel::ERROR_REC_DEVICE_SATURATION,
- listener->error());
- EXPECT_EQ(ssrc, listener->ssrc());
-
- // Testing a non-existing channel.
- listener->Reset();
- voe_.TriggerCallbackOnError(voe_.GetLastChannel() + 2,
- VE_SATURATION_WARNING);
- EXPECT_EQ(0, listener->error());
-}
-
TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
EXPECT_TRUE(SetupEngine());
send_parameters_.options = options_conference_;
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 6eb9b58..fc998c2 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -1301,10 +1301,6 @@
if (!BaseChannel::Init()) {
return false;
}
- media_channel()->SignalMediaError.connect(
- this, &VoiceChannel::OnVoiceChannelError);
- srtp_filter()->SignalSrtpError.connect(
- this, &VoiceChannel::OnSrtpError);
return true;
}
@@ -1430,9 +1426,7 @@
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
bool recv = IsReadyToReceive();
- if (!media_channel()->SetPlayout(recv)) {
- SendLastMediaError();
- }
+ media_channel()->SetPlayout(recv);
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
@@ -1440,7 +1434,6 @@
SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
if (!media_channel()->SetSend(send_flag)) {
LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
- SendLastMediaError();
}
LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
@@ -1571,7 +1564,6 @@
case MSG_CHANNEL_ERROR: {
VoiceChannelErrorMessageData* data =
static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
- SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
@@ -1597,36 +1589,6 @@
SignalAudioMonitor(this, info);
}
-void VoiceChannel::OnVoiceChannelError(
- uint32 ssrc, VoiceMediaChannel::Error err) {
- VoiceChannelErrorMessageData* data = new VoiceChannelErrorMessageData(
- ssrc, err);
- signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
-}
-
-void VoiceChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
- SrtpFilter::Error error) {
- switch (error) {
- case SrtpFilter::ERROR_FAIL:
- OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- VoiceMediaChannel::ERROR_REC_SRTP_ERROR :
- VoiceMediaChannel::ERROR_PLAY_SRTP_ERROR);
- break;
- case SrtpFilter::ERROR_AUTH:
- OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- VoiceMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
- VoiceMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
- break;
- case SrtpFilter::ERROR_REPLAY:
- // Only receving channel should have this error.
- ASSERT(mode == SrtpFilter::UNPROTECT);
- OnVoiceChannelError(ssrc, VoiceMediaChannel::ERROR_PLAY_SRTP_REPLAY);
- break;
- default:
- break;
- }
-}
-
void VoiceChannel::GetSrtpCryptoSuiteNames(
std::vector<std::string>* ciphers) const {
GetSupportedAudioCryptoSuites(ciphers);
@@ -1649,20 +1611,9 @@
if (!BaseChannel::Init()) {
return false;
}
- media_channel()->SignalMediaError.connect(
- this, &VideoChannel::OnVideoChannelError);
- srtp_filter()->SignalSrtpError.connect(
- this, &VideoChannel::OnSrtpError);
return true;
}
-void VoiceChannel::SendLastMediaError() {
- uint32 ssrc;
- VoiceMediaChannel::Error error;
- media_channel()->GetLastMediaError(&ssrc, &error);
- SignalMediaError(this, ssrc, error);
-}
-
VideoChannel::~VideoChannel() {
std::vector<uint32> screencast_ssrcs;
ScreencastMap::iterator iter;
@@ -1964,7 +1915,6 @@
case MSG_CHANNEL_ERROR: {
const VideoChannelErrorMessageData* data =
static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
- SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
@@ -2029,38 +1979,6 @@
return false;
}
-void VideoChannel::OnVideoChannelError(uint32 ssrc,
- VideoMediaChannel::Error error) {
- VideoChannelErrorMessageData* data = new VideoChannelErrorMessageData(
- ssrc, error);
- signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
-}
-
-void VideoChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
- SrtpFilter::Error error) {
- switch (error) {
- case SrtpFilter::ERROR_FAIL:
- OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- VideoMediaChannel::ERROR_REC_SRTP_ERROR :
- VideoMediaChannel::ERROR_PLAY_SRTP_ERROR);
- break;
- case SrtpFilter::ERROR_AUTH:
- OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- VideoMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
- VideoMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
- break;
- case SrtpFilter::ERROR_REPLAY:
- // Only receving channel should have this error.
- ASSERT(mode == SrtpFilter::UNPROTECT);
- // TODO(gangji): Turn on the signaling of replay error once we have
- // switched to the new mechanism for doing video retransmissions.
- // OnVideoChannelError(ssrc, VideoMediaChannel::ERROR_PLAY_SRTP_REPLAY);
- break;
- default:
- break;
- }
-}
-
void VideoChannel::GetSrtpCryptoSuiteNames(
std::vector<std::string>* ciphers) const {
GetSupportedVideoCryptoSuites(ciphers);
@@ -2093,14 +2011,10 @@
}
media_channel()->SignalDataReceived.connect(
this, &DataChannel::OnDataReceived);
- media_channel()->SignalMediaError.connect(
- this, &DataChannel::OnDataChannelError);
media_channel()->SignalReadyToSend.connect(
this, &DataChannel::OnDataChannelReadyToSend);
media_channel()->SignalStreamClosedRemotely.connect(
this, &DataChannel::OnStreamClosedRemotely);
- srtp_filter()->SignalSrtpError.connect(
- this, &DataChannel::OnSrtpError);
return true;
}
@@ -2308,7 +2222,6 @@
case MSG_CHANNEL_ERROR: {
const DataChannelErrorMessageData* data =
static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
- SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
@@ -2374,29 +2287,6 @@
new DataChannelReadyToSendMessageData(writable));
}
-void DataChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
- SrtpFilter::Error error) {
- switch (error) {
- case SrtpFilter::ERROR_FAIL:
- OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- DataMediaChannel::ERROR_SEND_SRTP_ERROR :
- DataMediaChannel::ERROR_RECV_SRTP_ERROR);
- break;
- case SrtpFilter::ERROR_AUTH:
- OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
- DataMediaChannel::ERROR_SEND_SRTP_AUTH_FAILED :
- DataMediaChannel::ERROR_RECV_SRTP_AUTH_FAILED);
- break;
- case SrtpFilter::ERROR_REPLAY:
- // Only receving channel should have this error.
- ASSERT(mode == SrtpFilter::UNPROTECT);
- OnDataChannelError(ssrc, DataMediaChannel::ERROR_RECV_SRTP_REPLAY);
- break;
- default:
- break;
- }
-}
-
void DataChannel::GetSrtpCryptoSuiteNames(
std::vector<std::string>* ciphers) const {
GetSupportedDataCryptoSuites(ciphers);
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index ac8736e..969f907 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -144,10 +144,6 @@
// For ConnectionStatsGetter, used by ConnectionMonitor
bool GetConnectionStats(ConnectionInfos* infos) override;
- void set_srtp_signal_silent_time(uint32 silent_time) {
- srtp_filter_.set_signal_silent_time(silent_time);
- }
-
BundleFilter* bundle_filter() { return &bundle_filter_; }
const std::vector<StreamParams>& local_streams() const {
@@ -170,6 +166,8 @@
// Only public for unit tests. Otherwise, consider protected.
virtual int SetOption(SocketType type, rtc::Socket::Option o, int val);
+ SrtpFilter* srtp_filter() { return &srtp_filter_; }
+
protected:
virtual MediaChannel* media_channel() const { return media_channel_; }
// Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
@@ -192,7 +190,6 @@
rtc::Thread* signaling_thread() {
return transport_controller_->signaling_thread();
}
- SrtpFilter* srtp_filter() { return &srtp_filter_; }
bool rtcp_transport_enabled() const { return rtcp_transport_enabled_; }
void ConnectToTransportChannel(TransportChannel* tc);
@@ -389,11 +386,6 @@
int GetOutputLevel_w();
void GetActiveStreams_w(AudioInfo::StreamList* actives);
- // Signal errors from VoiceMediaChannel. Arguments are:
- // ssrc(uint32), and error(VoiceMediaChannel::Error).
- sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error>
- SignalMediaError;
-
private:
// overrides from BaseChannel
virtual void OnChannelRead(TransportChannel* channel,
@@ -420,9 +412,6 @@
virtual void OnMediaMonitorUpdate(
VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
- void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error);
- void SendLastMediaError();
- void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
static const int kEarlyMediaTimeout = 1000;
MediaEngineInterface* media_engine_;
@@ -481,8 +470,6 @@
bool SendIntraFrame();
bool RequestIntraFrame();
- sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error>
- SignalMediaError;
// Configure sending media on the stream with SSRC |ssrc|
// If there is only one sending stream SSRC 0 can be used.
@@ -521,9 +508,6 @@
virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
bool GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc);
- void OnVideoChannelError(uint32 ssrc, VideoMediaChannel::Error error);
- void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
-
VideoRenderer* renderer_;
ScreencastMap screencast_capturers_;
rtc::scoped_ptr<VideoMediaMonitor> media_monitor_;
@@ -564,8 +548,6 @@
sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
SignalConnectionMonitor;
- sigslot::signal3<DataChannel*, uint32, DataMediaChannel::Error>
- SignalMediaError;
sigslot::signal3<DataChannel*,
const ReceiveDataParams&,
const rtc::Buffer&>
@@ -646,7 +628,6 @@
const ReceiveDataParams& params, const char* data, size_t len);
void OnDataChannelError(uint32 ssrc, DataMediaChannel::Error error);
void OnDataChannelReadyToSend(bool writable);
- void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
void OnStreamClosedRemotely(uint32 sid);
rtc::scoped_ptr<DataMediaMonitor> media_monitor_;
diff --git a/talk/session/media/channel_unittest.cc b/talk/session/media/channel_unittest.cc
index 784d53c..7f76e16 100644
--- a/talk/session/media/channel_unittest.cc
+++ b/talk/session/media/channel_unittest.cc
@@ -139,9 +139,7 @@
rtp_packet_(reinterpret_cast<const char*>(rtp_data), rtp_len),
rtcp_packet_(reinterpret_cast<const char*>(rtcp_data), rtcp_len),
media_info_callbacks1_(),
- media_info_callbacks2_(),
- ssrc_(0),
- error_(T::MediaChannel::ERROR_NONE) {}
+ media_info_callbacks2_() {}
void CreateChannels(int flags1, int flags2) {
CreateChannels(new typename T::MediaChannel(NULL, typename T::Options()),
@@ -163,10 +161,6 @@
this, &ChannelTest<T>::OnMediaMonitor);
channel2_->SignalMediaMonitor.connect(
this, &ChannelTest<T>::OnMediaMonitor);
- channel1_->SignalMediaError.connect(
- this, &ChannelTest<T>::OnMediaChannelError);
- channel2_->SignalMediaError.connect(
- this, &ChannelTest<T>::OnMediaChannelError);
if ((flags1 & DTLS) && (flags2 & DTLS)) {
flags1 = (flags1 & ~SECURE);
flags2 = (flags2 & ~SECURE);
@@ -472,13 +466,6 @@
}
}
- void OnMediaChannelError(typename T::Channel* channel,
- uint32 ssrc,
- typename T::MediaChannel::Error error) {
- ssrc_ = ssrc;
- error_ = error;
- }
-
void AddLegacyStreamInContent(uint32 ssrc, int flags,
typename T::Content* content) {
// Base implementation.
@@ -1677,16 +1664,20 @@
EXPECT_TRUE(CheckRtcp2());
}
- void TestChangeStateError() {
- CreateChannels(RTCP, RTCP);
- EXPECT_TRUE(SendInitiate());
- media_channel2_->set_fail_set_send(true);
- EXPECT_TRUE(channel2_->Enable(true));
- EXPECT_EQ(cricket::VoiceMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED,
- error_);
- }
-
void TestSrtpError(int pl_type) {
+ struct SrtpErrorHandler : public sigslot::has_slots<> {
+ SrtpErrorHandler() :
+ mode_(cricket::SrtpFilter::UNPROTECT),
+ error_(cricket::SrtpFilter::ERROR_NONE) {}
+ void OnSrtpError(uint32 ssrc, cricket::SrtpFilter::Mode mode,
+ cricket::SrtpFilter::Error error) {
+ mode_ = mode;
+ error_ = error;
+ }
+ cricket::SrtpFilter::Mode mode_;
+ cricket::SrtpFilter::Error error_;
+ } error_handler;
+
// For Audio, only pl_type 0 is added to the bundle filter.
// For Video, only pl_type 97 is added to the bundle filter.
// So we need to pass in pl_type so that the packet can pass through
@@ -1711,30 +1702,36 @@
EXPECT_TRUE(SendAccept());
EXPECT_TRUE(channel1_->secure());
EXPECT_TRUE(channel2_->secure());
- channel2_->set_srtp_signal_silent_time(200);
+
+ channel2_->srtp_filter()->SignalSrtpError.connect(
+ &error_handler, &SrtpErrorHandler::OnSrtpError);
// Testing failures in sending packets.
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
// The first failure will trigger an error.
- EXPECT_EQ_WAIT(T::MediaChannel::ERROR_REC_SRTP_ERROR, error_, 500);
- error_ = T::MediaChannel::ERROR_NONE;
+ EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+ EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);
+ error_handler.error_ = cricket::SrtpFilter::ERROR_NONE;
// The next 1 sec failures will not trigger an error.
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
// Wait for a while to ensure no message comes in.
rtc::Thread::Current()->ProcessMessages(210);
- EXPECT_EQ(T::MediaChannel::ERROR_NONE, error_);
+ EXPECT_EQ(cricket::SrtpFilter::ERROR_NONE, error_handler.error_);
// The error will be triggered again.
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
- EXPECT_EQ_WAIT(T::MediaChannel::ERROR_REC_SRTP_ERROR, error_, 500);
+ EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+ EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);
// Testing failures in receiving packets.
- error_ = T::MediaChannel::ERROR_NONE;
+ error_handler.error_ = cricket::SrtpFilter::ERROR_NONE;
+
cricket::TransportChannel* transport_channel =
channel2_->transport_channel();
transport_channel->SignalReadPacket(
transport_channel, reinterpret_cast<const char*>(kBadPacket),
sizeof(kBadPacket), rtc::PacketTime(), 0);
- EXPECT_EQ_WAIT(T::MediaChannel::ERROR_PLAY_SRTP_ERROR, error_, 500);
+ EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+ EXPECT_EQ(cricket::SrtpFilter::UNPROTECT, error_handler.mode_);
}
void TestOnReadyToSend() {
@@ -1802,9 +1799,6 @@
std::string rtcp_packet_;
int media_info_callbacks1_;
int media_info_callbacks2_;
-
- uint32 ssrc_;
- typename T::MediaChannel::Error error_;
};
template<>
@@ -2168,10 +2162,6 @@
Base::TestFlushRtcp();
}
-TEST_F(VoiceChannelTest, TestChangeStateError) {
- Base::TestChangeStateError();
-}
-
TEST_F(VoiceChannelTest, TestSrtpError) {
Base::TestSrtpError(kAudioPts[0]);
}
@@ -2524,8 +2514,6 @@
kVideoPts, ARRAY_SIZE(kVideoPts), true, true);
}
-// TODO(gangji): Add VideoChannelTest.TestChangeStateError.
-
TEST_F(VideoChannelTest, TestSrtpError) {
Base::TestSrtpError(kVideoPts[0]);
}
