diff options
| -rw-r--r-- | engine_configurations.h | 4 | ||||
| -rw-r--r-- | voice_engine/channel.cc | 88 | ||||
| -rw-r--r-- | voice_engine/channel.h | 9 | ||||
| -rw-r--r-- | voice_engine/include/voe_call_report.h | 87 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc | 2 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/fixtures/before_initialization_fixture.h | 2 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/standard/call_report_test.cc | 66 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/voe_standard_test.cc | 14 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/voe_standard_test.h | 14 | ||||
| -rw-r--r-- | voice_engine/test/auto_test/voe_test_defines.h | 4 | ||||
| -rw-r--r-- | voice_engine/voe_call_report_impl.cc | 373 | ||||
| -rw-r--r-- | voice_engine/voe_call_report_impl.h | 57 | ||||
| -rw-r--r-- | voice_engine/voice_engine.gyp | 4 | ||||
| -rw-r--r-- | voice_engine/voice_engine_impl.h | 9 |
14 files changed, 1 insertions, 732 deletions
diff --git a/engine_configurations.h b/engine_configurations.h index 0fb09496..45536bfd 100644 --- a/engine_configurations.h +++ b/engine_configurations.h @@ -84,10 +84,6 @@ #define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API #define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API -#ifndef WEBRTC_CHROMIUM_BUILD -#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API -#endif - // ============================================================================ // VideoEngine // ============================================================================ diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc index 2c75a790..4f310d36 100644 --- a/voice_engine/channel.cc +++ b/voice_engine/channel.cc @@ -502,8 +502,6 @@ Channel::OnPeriodicDeadOrAlive(int32_t id, isAlive = (_outputSpeechType != AudioFrame::kPLCCNG); } - UpdateDeadOrAliveCounters(isAlive); - // Send callback to the registered observer if (_connectionObserver) { @@ -908,8 +906,6 @@ Channel::Channel(int32_t channelId, _rtpTimeOutSeconds(0), _connectionObserver(false), _connectionObserverPtr(NULL), - _countAliveDetections(0), - _countDeadDetections(0), _outputSpeechType(AudioFrame::kNormalSpeech), vie_network_(NULL), video_channel_(-1), @@ -4354,68 +4350,6 @@ int Channel::SetExternalMixing(bool enabled) { } int -Channel::ResetRTCPStatistics() -{ - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::ResetRTCPStatistics()"); - uint32_t remoteSSRC(0); - remoteSSRC = rtp_receiver_->SSRC(); - return _rtpRtcpModule->ResetRTT(remoteSSRC); -} - -int -Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const -{ - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::GetRoundTripTimeSummary()"); - // Override default module outputs for the case when RTCP is disabled. - // This is done to ensure that we are backward compatible with the - // VoiceEngine where we did not use RTP/RTCP module. - if (!_rtpRtcpModule->RTCP()) - { - delaysMs.min = -1; - delaysMs.max = -1; - delaysMs.average = -1; - WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::GetRoundTripTimeSummary() RTCP is disabled =>" - " valid RTT measurements cannot be retrieved"); - return 0; - } - - uint32_t remoteSSRC; - uint16_t RTT; - uint16_t avgRTT; - uint16_t maxRTT; - uint16_t minRTT; - // The remote SSRC will be zero if no RTP packet has been received. - remoteSSRC = rtp_receiver_->SSRC(); - if (remoteSSRC == 0) - { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::GetRoundTripTimeSummary() unable to measure RTT" - " since no RTP packet has been received yet"); - } - - // Retrieve RTT statistics from the RTP/RTCP module for the specified - // channel and SSRC. The SSRC is required to parse out the correct source - // in conference scenarios. - if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0) - { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), - "GetRoundTripTimeSummary unable to retrieve RTT values" - " from the RTCP layer"); - delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1; - } - else - { - delaysMs.min = minRTT; - delaysMs.max = maxRTT; - delaysMs.average = avgRTT; - } - return 0; -} - -int Channel::GetNetworkStatistics(NetworkStatistics& stats) { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), @@ -4810,28 +4744,6 @@ Channel::InsertInbandDtmfTone() return 0; } -void -Channel::ResetDeadOrAliveCounters() -{ - _countDeadDetections = 0; - _countAliveDetections = 0; -} - -void -Channel::UpdateDeadOrAliveCounters(bool alive) -{ - if (alive) - _countAliveDetections++; - else - _countDeadDetections++; -} - -int -Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const -{ - return 0; -} - int32_t Channel::SendPacketRaw(const void *data, int len, bool RTCP) { diff --git a/voice_engine/channel.h b/voice_engine/channel.h index b490f489..7a366924 100644 --- a/voice_engine/channel.h +++ b/voice_engine/channel.h @@ -281,12 +281,6 @@ public: int SetChannelOutputVolumeScaling(float scaling); int GetChannelOutputVolumeScaling(float& scaling) const; - // VoECallReport - void ResetDeadOrAliveCounters(); - int ResetRTCPStatistics(); - int GetRoundTripTimeSummary(StatVal& delaysMs) const; - int GetDeadOrAliveCounters(int& countDead, int& countAlive) const; - // VoENetEqStats int GetNetworkStatistics(NetworkStatistics& stats); void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; @@ -514,7 +508,6 @@ private: int InsertInbandDtmfTone(); int32_t MixOrReplaceAudioWithFile(int mixingFrequency); int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); - void UpdateDeadOrAliveCounters(bool alive); int32_t SendPacketRaw(const void *data, int len, bool RTCP); void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber); @@ -615,8 +608,6 @@ private: uint32_t _rtpTimeOutSeconds; bool _connectionObserver; VoEConnectionObserver* _connectionObserverPtr; - uint32_t _countAliveDetections; - uint32_t _countDeadDetections; AudioFrame::SpeechType _outputSpeechType; ViENetwork* vie_network_; int video_channel_; diff --git a/voice_engine/include/voe_call_report.h b/voice_engine/include/voe_call_report.h deleted file mode 100644 index de2c7781..00000000 --- a/voice_engine/include/voe_call_report.h +++ /dev/null @@ -1,87 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This sub-API supports the following functionalities: -// -// - Long-term speech and noise level metrics. -// - Long-term echo metric statistics. -// - Round Trip Time (RTT) statistics. -// - Dead-or-Alive connection summary. -// - Generation of call reports to text files. -// -// Usage example, omitting error checking: -// -// using namespace webrtc; -// VoiceEngine* voe = VoiceEngine::Create(); -// VoEBase* base = VoEBase::GetInterface(voe); -// VoECallReport report = VoECallReport::GetInterface(voe); -// base->Init(); -// LevelStatistics stats; -// report->GetSpeechAndNoiseSummary(stats); -// ... -// base->Terminate(); -// base->Release(); -// report->Release(); -// VoiceEngine::Delete(voe); -// -#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H -#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H - -#include "webrtc/common_types.h" - -namespace webrtc { - -class VoiceEngine; - -// VoECallReport -class WEBRTC_DLLEXPORT VoECallReport -{ -public: - // Factory for the VoECallReport sub-API. Increases an internal - // reference counter if successful. Returns NULL if the API is not - // supported or if construction fails. - static VoECallReport* GetInterface(VoiceEngine* voiceEngine); - - // Releases the VoECallReport sub-API and decreases an internal - // reference counter. Returns the new reference count. This value should - // be zero for all sub-API:s before the VoiceEngine object can be safely - // deleted. - virtual int Release() = 0; - - // Performs a combined reset of all components involved in generating - // the call report for a specified |channel|. Pass in -1 to reset - // all channels. - virtual int ResetCallReportStatistics(int channel) = 0; - - // Gets minimum, maximum and average levels for long-term echo metrics. - virtual int GetEchoMetricSummary(EchoStatistics& stats) = 0; - - // Gets minimum, maximum and average levels for Round Trip Time (RTT) - // measurements. - virtual int GetRoundTripTimeSummary(int channel, - StatVal& delaysMs) = 0; - - // Gets the total amount of dead and alive connection detections - // during a VoIP session. - virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections, - int& numOfAliveDetections) = 0; - - // Creates a text file in ASCII format, which contains a summary - // of all the statistics that can be obtained by the call report sub-API. - virtual int WriteReportToFile(const char* fileNameUTF8) = 0; - -protected: - VoECallReport() { } - virtual ~VoECallReport() { } -}; - -} // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc index fcace2a0..ebb874e7 100644 --- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc +++ b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc @@ -36,7 +36,6 @@ BeforeInitializationFixture::BeforeInitializationFixture() { voe_vsync_ = webrtc::VoEVideoSync::GetInterface(voice_engine_); voe_hardware_ = webrtc::VoEHardware::GetInterface(voice_engine_); voe_xmedia_ = webrtc::VoEExternalMedia::GetInterface(voice_engine_); - voe_call_report_ = webrtc::VoECallReport::GetInterface(voice_engine_); voe_neteq_stats_ = webrtc::VoENetEqStats::GetInterface(voice_engine_); } @@ -52,7 +51,6 @@ BeforeInitializationFixture::~BeforeInitializationFixture() { voe_vsync_->Release(); voe_hardware_->Release(); voe_xmedia_->Release(); - voe_call_report_->Release(); voe_neteq_stats_->Release(); EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine_)); diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h index e58a10b2..7a3fad83 100644 --- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h +++ b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h @@ -19,7 +19,6 @@ #include "webrtc/test/testsupport/gtest_disable.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/voice_engine/include/voe_call_report.h" #include "webrtc/voice_engine/include/voe_codec.h" #include "webrtc/voice_engine/include/voe_dtmf.h" #include "webrtc/voice_engine/include/voe_errors.h" @@ -68,7 +67,6 @@ class BeforeInitializationFixture : public testing::Test { webrtc::VoEVideoSync* voe_vsync_; webrtc::VoEHardware* voe_hardware_; webrtc::VoEExternalMedia* voe_xmedia_; - webrtc::VoECallReport* voe_call_report_; webrtc::VoENetEqStats* voe_neteq_stats_; webrtc::Config config_; }; diff --git a/voice_engine/test/auto_test/standard/call_report_test.cc b/voice_engine/test/auto_test/standard/call_report_test.cc deleted file mode 100644 index ee98dc1e..00000000 --- a/voice_engine/test/auto_test/standard/call_report_test.cc +++ /dev/null @@ -1,66 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" - -class CallReportTest : public AfterStreamingFixture { -}; - -TEST_F(CallReportTest, ResetCallReportStatisticsFailsForBadInput) { - EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(-2)); - EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(1)); -} - -TEST_F(CallReportTest, ResetCallReportStatisticsSucceedsWithCorrectInput) { - EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(channel_)); - EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(-1)); -} - -TEST_F(CallReportTest, EchoMetricSummarySucceeds) { - EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true)); - Sleep(1000); - - webrtc::EchoStatistics echo_statistics; - EXPECT_EQ(0, voe_call_report_->GetEchoMetricSummary(echo_statistics)); -} - -TEST_F(CallReportTest, GetRoundTripTimeSummaryReturnsAllMinusOnesIfRtcpIsOff) { - voe_rtp_rtcp_->SetRTCPStatus(channel_, false); - - webrtc::StatVal delays; - EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays)); - EXPECT_EQ(-1, delays.average); - EXPECT_EQ(-1, delays.min); - EXPECT_EQ(-1, delays.max); -} - -// Flaky: https://code.google.com/p/webrtc/issues/detail?id=1719 -TEST_F(CallReportTest, DISABLED_GetRoundTripTimesReturnsValuesIfRtcpIsOn) { - voe_rtp_rtcp_->SetRTCPStatus(channel_, true); - Sleep(1000); - - webrtc::StatVal delays; - EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays)); - EXPECT_NE(-1, delays.average); - EXPECT_NE(-1, delays.min); - EXPECT_NE(-1, delays.max); -} - -TEST_F(CallReportTest, WriteReportToFileFailsOnBadInput) { - EXPECT_EQ(-1, voe_call_report_->WriteReportToFile(NULL)); -} - -TEST_F(CallReportTest, WriteReportToFileSucceedsWithCorrectFilename) { - std::string output_path = webrtc::test::OutputPath(); - std::string report_filename = output_path + "call_report.txt"; - - EXPECT_EQ(0, voe_call_report_->WriteReportToFile(report_filename.c_str())); -} diff --git a/voice_engine/test/auto_test/voe_standard_test.cc b/voice_engine/test/auto_test/voe_standard_test.cc index 5d0fdb0e..53614458 100644 --- a/voice_engine/test/auto_test/voe_standard_test.cc +++ b/voice_engine/test/auto_test/voe_standard_test.cc @@ -42,8 +42,6 @@ void SubAPIManager::DisplayStatus() const { TEST_LOG("Supported sub APIs:\n\n"); if (_base) TEST_LOG(" Base\n"); - if (_callReport) - TEST_LOG(" CallReport\n"); if (_codec) TEST_LOG(" Codec\n"); if (_dtmf) @@ -70,8 +68,6 @@ void SubAPIManager::DisplayStatus() const { TEST_LOG("Excluded sub APIs:\n\n"); if (!_base) TEST_LOG(" Base\n"); - if (!_callReport) - TEST_LOG(" CallReport\n"); if (!_codec) TEST_LOG(" Codec\n"); if (!_dtmf) @@ -101,7 +97,6 @@ VoETestManager::VoETestManager() : initialized_(false), voice_engine_(NULL), voe_base_(0), - voe_call_report_(0), voe_codec_(0), voe_dtmf_(0), voe_xmedia_(0), @@ -174,9 +169,6 @@ void VoETestManager::GetInterfaces() { #ifdef _TEST_XMEDIA_ voe_xmedia_ = VoEExternalMedia::GetInterface(voice_engine_); #endif -#ifdef _TEST_CALL_REPORT_ - voe_call_report_ = VoECallReport::GetInterface(voice_engine_); -#endif #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API voe_neteq_stats_ = VoENetEqStats::GetInterface(voice_engine_); #endif @@ -234,12 +226,6 @@ int VoETestManager::ReleaseInterfaces() { voe_xmedia_ = NULL; } #endif -#ifdef _TEST_CALL_REPORT_ - if (voe_call_report_) { - voe_call_report_->Release(); - voe_call_report_ = NULL; - } -#endif #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API if (voe_neteq_stats_) { voe_neteq_stats_->Release(); diff --git a/voice_engine/test/auto_test/voe_standard_test.h b/voice_engine/test/auto_test/voe_standard_test.h index 15dde0cd..566014db 100644 --- a/voice_engine/test/auto_test/voe_standard_test.h +++ b/voice_engine/test/auto_test/voe_standard_test.h @@ -25,9 +25,6 @@ #include "webrtc/voice_engine/test/auto_test/resource_manager.h" #include "webrtc/voice_engine/test/auto_test/voe_test_common.h" #include "webrtc/voice_engine/test/auto_test/voe_test_interface.h" -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API -#include "webrtc/voice_engine/include/voe_call_report.h" -#endif #ifdef WEBRTC_VOICE_ENGINE_CODEC_API #include "webrtc/voice_engine/include/voe_codec.h" #endif @@ -65,7 +62,6 @@ class SubAPIManager { public: SubAPIManager() : _base(true), - _callReport(false), _codec(false), _dtmf(false), _externalMedia(false), @@ -77,9 +73,6 @@ class SubAPIManager { _videoSync(false), _volumeControl(false), _apm(false) { -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API - _callReport = true; -#endif #ifdef WEBRTC_VOICE_ENGINE_CODEC_API _codec = true; #endif @@ -116,7 +109,7 @@ class SubAPIManager { void DisplayStatus() const; private: - bool _base, _callReport, _codec, _dtmf; + bool _base, _codec, _dtmf; bool _externalMedia, _file, _hardware; bool _netEqStats, _network, _rtp_rtcp, _videoSync, _volumeControl, _apm; }; @@ -182,10 +175,6 @@ class VoETestManager { return voe_xmedia_; } - VoECallReport* CallReportPtr() const { - return voe_call_report_; - } - #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API VoENetEqStats* NetEqStatsPtr() const { return voe_neteq_stats_; @@ -197,7 +186,6 @@ class VoETestManager { VoiceEngine* voice_engine_; VoEBase* voe_base_; - VoECallReport* voe_call_report_; VoECodec* voe_codec_; VoEDtmf* voe_dtmf_; VoEExternalMedia* voe_xmedia_; diff --git a/voice_engine/test/auto_test/voe_test_defines.h b/voice_engine/test/auto_test/voe_test_defines.h index c2fb38e9..10f64922 100644 --- a/voice_engine/test/auto_test/voe_test_defines.h +++ b/voice_engine/test/auto_test/voe_test_defines.h @@ -28,7 +28,6 @@ #define _TEST_AUDIO_PROCESSING_ #define _TEST_FILE_ #define _TEST_NETWORK_ -#define _TEST_CALL_REPORT_ #define _TEST_VIDEO_SYNC_ #define _TEST_NETEQ_STATS_ #define _TEST_XMEDIA_ @@ -77,9 +76,6 @@ #ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API #undef _TEST_NETEQ_STATS_ #endif -#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API -#undef _TEST_CALL_REPORT_ -#endif // Some parts can cause problems while running Insure #ifdef __INSURE__ diff --git a/voice_engine/voe_call_report_impl.cc b/voice_engine/voe_call_report_impl.cc deleted file mode 100644 index 9febc1c3..00000000 --- a/voice_engine/voe_call_report_impl.cc +++ /dev/null @@ -1,373 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/voice_engine/voe_call_report_impl.h" - -#include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/file_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/voice_engine/channel.h" -#include "webrtc/voice_engine/include/voe_errors.h" -#include "webrtc/voice_engine/voice_engine_impl.h" - -namespace webrtc -{ - -VoECallReport* VoECallReport::GetInterface(VoiceEngine* voiceEngine) -{ -#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API - return NULL; -#else - if (NULL == voiceEngine) - { - return NULL; - } - VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine); - s->AddRef(); - return s; -#endif -} - -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API - -VoECallReportImpl::VoECallReportImpl(voe::SharedData* shared) : - _file(*FileWrapper::Create()), _shared(shared) -{ - WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1), - "VoECallReportImpl() - ctor"); -} - -VoECallReportImpl::~VoECallReportImpl() -{ - WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1), - "~VoECallReportImpl() - dtor"); - delete &_file; -} - -int VoECallReportImpl::ResetCallReportStatistics(int channel) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "ResetCallReportStatistics(channel=%d)", channel); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - - assert(_shared->audio_processing() != NULL); - - bool echoMode = - _shared->audio_processing()->echo_cancellation()->are_metrics_enabled(); - - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), - " current AudioProcessingModule echo metric state %d)", - echoMode); - // Reset the APM statistics - if (_shared->audio_processing()->echo_cancellation()->enable_metrics(true) - != 0) - { - _shared->SetLastError(VE_APM_ERROR, kTraceError, - "ResetCallReportStatistics() unable to " - "set the AudioProcessingModule echo metrics state"); - return -1; - } - // Restore metric states - _shared->audio_processing()->echo_cancellation()->enable_metrics(echoMode); - - // Reset channel dependent statistics - if (channel != -1) - { - voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel); - voe::Channel* channelPtr = ch.channel(); - if (channelPtr == NULL) - { - _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, - "ResetCallReportStatistics() failed to locate channel"); - return -1; - } - channelPtr->ResetDeadOrAliveCounters(); - channelPtr->ResetRTCPStatistics(); - } else { - for (voe::ChannelManager::Iterator it(&_shared->channel_manager()); - it.IsValid(); - it.Increment()) { - it.GetChannel()->ResetDeadOrAliveCounters(); - it.GetChannel()->ResetRTCPStatistics(); - } - } - - return 0; -} - -int VoECallReportImpl::GetEchoMetricSummary(EchoStatistics& stats) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "GetEchoMetricSummary()"); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - assert(_shared->audio_processing() != NULL); - - return (GetEchoMetricSummaryInternal(stats)); -} - -int VoECallReportImpl::GetEchoMetricSummaryInternal(EchoStatistics& stats) -{ - // Retrieve echo metrics from the AudioProcessingModule - int ret(0); - bool mode(false); - EchoCancellation::Metrics metrics; - - // Ensure that echo metrics is enabled - - mode = - _shared->audio_processing()->echo_cancellation()->are_metrics_enabled(); - if (mode != false) - { - ret = _shared->audio_processing()->echo_cancellation()-> - GetMetrics(&metrics); - if (ret != 0) - { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, - VoEId(_shared->instance_id(), -1), - " AudioProcessingModule GetMetrics() => error"); - } - } - else - { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, - VoEId(_shared->instance_id(), -1), - " AudioProcessingModule echo metrics is not enabled"); - } - - if ((ret != 0) || (mode == false)) - { - // Mark complete struct as invalid (-100 dB) - WEBRTC_TRACE(kTraceWarning, kTraceVoice, - VoEId(_shared->instance_id(), -1), - " unable to retrieve echo metrics from the AudioProcessingModule"); - stats.erl.min = -100; - stats.erl.max = -100; - stats.erl.average = -100; - stats.erle.min = -100; - stats.erle.max = -100; - stats.erle.average = -100; - stats.rerl.min = -100; - stats.rerl.max = -100; - stats.rerl.average = -100; - stats.a_nlp.min = -100; - stats.a_nlp.max = -100; - stats.a_nlp.average = -100; - } - else - { - - // Deliver output results to user - stats.erl.min = metrics.echo_return_loss.minimum; - stats.erl.max = metrics.echo_return_loss.maximum; - stats.erl.average = metrics.echo_return_loss.average; - WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, - VoEId(_shared->instance_id(), -1), " erl: min=%d, max=%d, avg=%d", - stats.erl.min, stats.erl.max, stats.erl.average); - - stats.erle.min = metrics.echo_return_loss_enhancement.minimum; - stats.erle.max = metrics.echo_return_loss_enhancement.maximum; - stats.erle.average = metrics.echo_return_loss_enhancement.average; - WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, - VoEId(_shared->instance_id(), -1), " erle: min=%d, max=%d, avg=%d", - stats.erle.min, stats.erle.max, stats.erle.average); - - stats.rerl.min = metrics.residual_echo_return_loss.minimum; - stats.rerl.max = metrics.residual_echo_return_loss.maximum; - stats.rerl.average = metrics.residual_echo_return_loss.average; - WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, - VoEId(_shared->instance_id(), -1), " rerl: min=%d, max=%d, avg=%d", - stats.rerl.min, stats.rerl.max, stats.rerl.average); - - stats.a_nlp.min = metrics.a_nlp.minimum; - stats.a_nlp.max = metrics.a_nlp.maximum; - stats.a_nlp.average = metrics.a_nlp.average; - WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, - VoEId(_shared->instance_id(), -1), - " a_nlp: min=%d, max=%d, avg=%d", - stats.a_nlp.min, stats.a_nlp.max, stats.a_nlp.average); - } - return 0; -} - -int VoECallReportImpl::GetRoundTripTimeSummary(int channel, StatVal& delaysMs) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "GetRoundTripTimeSummary()"); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel); - voe::Channel* channelPtr = ch.channel(); - if (channelPtr == NULL) - { - _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, - "GetRoundTripTimeSummary() failed to locate channel"); - return -1; - } - - return channelPtr->GetRoundTripTimeSummary(delaysMs); -} - -int VoECallReportImpl::GetDeadOrAliveSummary(int channel, - int& numOfDeadDetections, - int& numOfAliveDetections) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "GetDeadOrAliveSummary(channel=%d)", channel); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - - return (GetDeadOrAliveSummaryInternal(channel, numOfDeadDetections, - numOfAliveDetections)); -} - -int VoECallReportImpl::GetDeadOrAliveSummaryInternal(int channel, - int& numOfDeadDetections, - int& numOfAliveDetections) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "GetDeadOrAliveSummary(channel=%d)", channel); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel); - voe::Channel* channelPtr = ch.channel(); - if (channelPtr == NULL) - { - _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, - "GetRoundTripTimeSummary() failed to locate channel"); - return -1; - } - - return channelPtr->GetDeadOrAliveCounters(numOfDeadDetections, - numOfAliveDetections); -} - -int VoECallReportImpl::WriteReportToFile(const char* fileNameUTF8) -{ - WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), - "WriteReportToFile(fileNameUTF8=%s)", fileNameUTF8); - - if (!_shared->statistics().Initialized()) - { - _shared->SetLastError(VE_NOT_INITED, kTraceError); - return -1; - } - - if (NULL == fileNameUTF8) - { - _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError, - "WriteReportToFile() invalid filename"); - return -1; - } - - if (_file.Open()) - { - _file.CloseFile(); - } - - // Open text file in write mode - if (_file.OpenFile(fileNameUTF8, false, false, true) != 0) - { - _shared->SetLastError(VE_BAD_FILE, kTraceError, - "WriteReportToFile() unable to open the file"); - return -1; - } - - // Summarize information and add it to the open file - // - _file.WriteText("WebRtc VoiceEngine Call Report\n"); - _file.WriteText("==============================\n"); - _file.WriteText("\nNetwork Packet Round Trip Time (RTT)\n"); - _file.WriteText("------------------------------------\n\n"); - - if (_shared->channel_manager().NumOfChannels() == 0) - return 0; - - for (voe::ChannelManager::Iterator it(&_shared->channel_manager()); - it.IsValid(); - it.Increment()) { - StatVal delaysMs; - _file.WriteText("channel %d:\n", it.GetChannel()->ChannelId()); - it.GetChannel()->GetRoundTripTimeSummary(delaysMs); - _file.WriteText(" min:%5d [ms]\n", delaysMs.min); - _file.WriteText(" max:%5d [ms]\n", delaysMs.max); - _file.WriteText(" avg:%5d [ms]\n", delaysMs.average); - } - - _file.WriteText("\nDead-or-Alive Connection Detections\n"); - _file.WriteText("------------------------------------\n\n"); - - for (voe::ChannelManager::Iterator it(&_shared->channel_manager()); - it.IsValid(); - it.Increment()) { - int dead = 0; - int alive = 0; - _file.WriteText("channel %d:\n", it.GetChannel()->ChannelId()); - GetDeadOrAliveSummary(it.GetChannel()->ChannelId(), dead, alive); - _file.WriteText(" #dead :%6d\n", dead); - _file.WriteText(" #alive:%6d\n", alive); - } - - EchoStatistics echo; - GetEchoMetricSummary(echo); - - _file.WriteText("\nEcho Metrics\n"); - _file.WriteText("------------\n\n"); - - _file.WriteText("erl:\n"); - _file.WriteText(" min:%5d [dB]\n", echo.erl.min); - _file.WriteText(" max:%5d [dB]\n", echo.erl.max); - _file.WriteText(" avg:%5d [dB]\n", echo.erl.average); - _file.WriteText("\nerle:\n"); - _file.WriteText(" min:%5d [dB]\n", echo.erle.min); - _file.WriteText(" max:%5d [dB]\n", echo.erle.max); - _file.WriteText(" avg:%5d [dB]\n", echo.erle.average); - _file.WriteText("rerl:\n"); - _file.WriteText(" min:%5d [dB]\n", echo.rerl.min); - _file.WriteText(" max:%5d [dB]\n", echo.rerl.max); - _file.WriteText(" avg:%5d [dB]\n", echo.rerl.average); - _file.WriteText("a_nlp:\n"); - _file.WriteText(" min:%5d [dB]\n", echo.a_nlp.min); - _file.WriteText(" max:%5d [dB]\n", echo.a_nlp.max); - _file.WriteText(" avg:%5d [dB]\n", echo.a_nlp.average); - - _file.WriteText("\n<END>"); - - _file.Flush(); - _file.CloseFile(); - - return 0; -} - -#endif // WEBRTC_VOICE_ENGINE_CALL_REPORT_API - -} // namespace webrtc diff --git a/voice_engine/voe_call_report_impl.h b/voice_engine/voe_call_report_impl.h deleted file mode 100644 index b9927f56..00000000 --- a/voice_engine/voe_call_report_impl.h +++ /dev/null @@ -1,57 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H -#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H - -#include "webrtc/voice_engine/include/voe_call_report.h" - -#include "webrtc/voice_engine/shared_data.h" - - -namespace webrtc -{ -class FileWrapper; - -class VoECallReportImpl: public VoECallReport -{ -public: - virtual int ResetCallReportStatistics(int channel); - - virtual int GetEchoMetricSummary(EchoStatistics& stats); - - virtual int GetRoundTripTimeSummary(int channel, - StatVal& delaysMs); - - virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections, - int& numOfAliveDetections); - - virtual int WriteReportToFile(const char* fileNameUTF8); - -protected: - VoECallReportImpl(voe::SharedData* shared); - virtual ~VoECallReportImpl(); - -private: - int GetDeadOrAliveSummaryInternal(int channel, - int& numOfDeadDetections, - int& numOfAliveDetections); - - int GetEchoMetricSummaryInternal(EchoStatistics& stats); - - int GetSpeechAndNoiseSummaryInternal(LevelStatistics& stats); - - FileWrapper& _file; - voe::SharedData* _shared; -}; - -} // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H diff --git a/voice_engine/voice_engine.gyp b/voice_engine/voice_engine.gyp index 16629d93..699677cf 100644 --- a/voice_engine/voice_engine.gyp +++ b/voice_engine/voice_engine.gyp @@ -31,7 +31,6 @@ '../typedefs.h', 'include/voe_audio_processing.h', 'include/voe_base.h', - 'include/voe_call_report.h', 'include/voe_codec.h', 'include/voe_dtmf.h', 'include/voe_errors.h', @@ -69,8 +68,6 @@ 'voe_audio_processing_impl.h', 'voe_base_impl.cc', 'voe_base_impl.h', - 'voe_call_report_impl.cc', - 'voe_call_report_impl.h', 'voe_codec_impl.cc', 'voe_codec_impl.h', 'voe_dtmf_impl.cc', @@ -166,7 +163,6 @@ 'test/auto_test/fixtures/before_initialization_fixture.cc', 'test/auto_test/fixtures/before_initialization_fixture.h', 'test/auto_test/standard/audio_processing_test.cc', - 'test/auto_test/standard/call_report_test.cc', 'test/auto_test/standard/codec_before_streaming_test.cc', 'test/auto_test/standard/codec_test.cc', 'test/auto_test/standard/dtmf_test.cc', diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h index ffdf5910..1d2b9c5e 100644 --- a/voice_engine/voice_engine_impl.h +++ b/voice_engine/voice_engine_impl.h @@ -18,9 +18,6 @@ #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API #include "webrtc/voice_engine/voe_audio_processing_impl.h" #endif -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API -#include "webrtc/voice_engine/voe_call_report_impl.h" -#endif #ifdef WEBRTC_VOICE_ENGINE_CODEC_API #include "webrtc/voice_engine/voe_codec_impl.h" #endif @@ -58,9 +55,6 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API public VoEAudioProcessingImpl, #endif -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API - public VoECallReportImpl, -#endif #ifdef WEBRTC_VOICE_ENGINE_CODEC_API public VoECodecImpl, #endif @@ -97,9 +91,6 @@ public: #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API VoEAudioProcessingImpl(this), #endif -#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API - VoECallReportImpl(this), -#endif #ifdef WEBRTC_VOICE_ENGINE_CODEC_API VoECodecImpl(this), #endif |