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Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

Browse Source 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrika@webrtc.org 2014-04-14 14:12:50 +00:00
parent 5692531f18
commit b9309beea4
14 changed files with 1 additions and 732 deletions
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4
webrtc/engine_configurations.h
View File

@ -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
// ============================================================================

88
webrtc/voice_engine/channel.cc
View File

@ -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),
@ -4353,68 +4349,6 @@ int Channel::SetExternalMixing(bool enabled) {
return 0;
}
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)
{
@ -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)
{

9
webrtc/voice_engine/channel.h
View File

@ -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_;

87
webrtc/voice_engine/include/voe_call_report.h
View File

@ -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

2
webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
View File

@ -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_));

2
webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
View File

@ -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_;
};

66
webrtc/voice_engine/test/auto_test/standard/call_report_test.cc
View File

@ -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()));
}

14
webrtc/voice_engine/test/auto_test/voe_standard_test.cc
View File

@ -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();

14
webrtc/voice_engine/test/auto_test/voe_standard_test.h
View File

@ -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_;

4
webrtc/voice_engine/test/auto_test/voe_test_defines.h
View File

@ -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__

373
webrtc/voice_engine/voe_call_report_impl.cc
View File

@ -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

57
webrtc/voice_engine/voe_call_report_impl.h
View File

@ -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

4
webrtc/voice_engine/voice_engine.gyp
View File

@ -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',

9
webrtc/voice_engine/voice_engine_impl.h
View File

@ -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