admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove voe::Statistics.

Browse Source 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3020473002
Cr-Commit-Position: refs/heads/master@{#20042}
This commit is contained in:
solenberg 2017-09-29 06:00:28 -07:00 committed by Commit Bot
parent 2666cf7eba
commit 1c239d476e
20 changed files with 69 additions and 408 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
audio/audio_receive_stream.cc
View File

@ -94,7 +94,7 @@ AudioReceiveStream::AudioReceiveStream(
RTC_CHECK_EQ(config.decoder_factory,
channel_proxy_->GetAudioDecoderFactory());
channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
channel_proxy_->RegisterTransport(config.rtcp_send_transport);
channel_proxy_->SetReceiveCodecs(config.decoder_map);
for (const auto& extension : config.rtp.extensions) {
@ -122,7 +122,7 @@ AudioReceiveStream::~AudioReceiveStream() {
Stop();
}
channel_proxy_->DisassociateSendChannel();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->RegisterTransport(nullptr);
channel_proxy_->ResetReceiverCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
}

5
audio/audio_receive_stream_unittest.cc
View File

@ -101,10 +101,7 @@ struct ConfigHelper {
.Times(1);
EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
.Times(1);
EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
.WillOnce(ReturnRef(decoder_factory_));
testing::Expectation expect_set =

6
audio/audio_send_stream.cc
View File

@ -120,7 +120,7 @@ AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->RegisterTransport(nullptr);
channel_proxy_->ResetSenderCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
channel_proxy_->SetRtcpRttStats(nullptr);
@ -164,7 +164,7 @@ void AudioSendStream::ConfigureStream(
if (first_time ||
new_config.send_transport != old_config.send_transport) {
if (old_config.send_transport) {
channel_proxy->DeRegisterExternalTransport();
channel_proxy->RegisterTransport(nullptr);
}
if (new_config.send_transport) {
stream->timed_send_transport_adapter_.reset(new TimedTransport(
@ -172,7 +172,7 @@ void AudioSendStream::ConfigureStream(
} else {
stream->timed_send_transport_adapter_.reset(nullptr);
}
channel_proxy->RegisterExternalTransport(
channel_proxy->RegisterTransport(
stream->timed_send_transport_adapter_.get());
}

3
audio/audio_send_stream_unittest.cc
View File

@ -235,8 +235,7 @@ struct ConfigHelper {
.Times(1);
EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1); // Destructor resets the event log

3
test/mock_voe_channel_proxy.h
View File

@ -65,8 +65,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
// TODO(solenberg): Talk the compiler into accepting this mock method:
// MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
MOCK_METHOD1(SetInputMute, void(bool muted));
MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
MOCK_METHOD0(DeRegisterExternalTransport, void());
MOCK_METHOD1(RegisterTransport, void(Transport* transport));
MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
MOCK_CONST_METHOD0(GetAudioDecoderFactory,

2
voice_engine/BUILD.gn
View File

@ -20,8 +20,6 @@ rtc_static_library("voice_engine") {
"include/voe_errors.h",
"shared_data.cc",
"shared_data.h",
"statistics.cc",
"statistics.h",
"transmit_mixer.cc",
"transmit_mixer.h",
"transport_feedback_packet_loss_tracker.cc",

119
voice_engine/channel.cc
View File

@ -40,7 +40,6 @@
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
namespace webrtc {
@ -447,9 +446,8 @@ int32_t Channel::SendData(FrameType frameType,
// received from the capture device as
// undefined for voice for now.
-1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"Channel::SendData() failed to send data to RTP/RTCP module");
LOG(LS_ERROR) <<
"Channel::SendData() failed to send data to RTP/RTCP module";
return -1;
}
@ -475,11 +473,7 @@ bool Channel::SendRtp(const uint8_t* data,
size_t bufferLength = len;
if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendPacket() RTP transmission using %s failed",
transport_name.c_str());
LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
return false;
}
return true;
@ -502,11 +496,7 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) {
int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
if (n < 0) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendRtcp() transmission using %s failed",
transport_name.c_str());
LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
return false;
}
return true;
@ -556,7 +546,6 @@ int32_t Channel::OnInitializeDecoder(
"Channel::OnInitializeDecoder() invalid codec ("
"pt=%d, name=%s) received - 1",
payloadType, payloadName);
_engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
return -1;
}
@ -585,9 +574,8 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
"Channel::OnReceivedPayloadData() unable to push data to the ACM");
LOG(LS_ERROR) <<
"Channel::OnReceivedPayloadData() unable to push data to the ACM";
return -1;
}
@ -759,7 +747,6 @@ Channel::Channel(int32_t channelId,
rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_externalTransport(false),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
ntp_estimator_(Clock::GetRealTimeClock()),
@ -769,10 +756,8 @@ Channel::Channel(int32_t channelId,
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_engineStatisticsPtr(NULL),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_callbackCritSectPtr(NULL),
_transportPtr(NULL),
input_mute_(false),
previous_frame_muted_(false),
@ -835,7 +820,7 @@ int32_t Channel::Init() {
// --- Initial sanity
if ((_engineStatisticsPtr == NULL) || (_moduleProcessThreadPtr == NULL)) {
if (_moduleProcessThreadPtr == NULL) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Init() must call SetEngineInformation() first");
return -1;
@ -848,9 +833,7 @@ int32_t Channel::Init() {
// --- ACM initialization
if (audio_coding_->InitializeReceiver() == -1) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"Channel::Init() unable to initialize the ACM - 1");
LOG(LS_ERROR) << "Channel::Init() unable to initialize the ACM - 1";
return -1;
}
@ -866,9 +849,7 @@ int32_t Channel::Init() {
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
// --- Register all permanent callbacks
if (audio_coding_->RegisterTransportCallback(this) == -1) {
_engineStatisticsPtr->SetLastError(
VE_CANNOT_INIT_CHANNEL, kTraceError,
"Channel::Init() callbacks not registered");
LOG(LS_ERROR) << "Channel::Init() callbacks not registered";
return -1;
}
@ -903,19 +884,15 @@ void Channel::Terminate() {
// End of modules shutdown
}
int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
ProcessThread& moduleProcessThread,
int32_t Channel::SetEngineInformation(ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
rtc::CriticalSection* callbackCritSect,
rtc::TaskQueue* encoder_queue) {
RTC_DCHECK(encoder_queue);
RTC_DCHECK(!encoder_queue_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
_callbackCritSectPtr = callbackCritSect;
encoder_queue_ = encoder_queue;
return 0;
}
@ -974,9 +951,7 @@ int32_t Channel::StartSend() {
}
_rtpRtcpModule->SetSendingMediaStatus(true);
if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"StartSend() RTP/RTCP failed to start sending");
LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
_rtpRtcpModule->SetSendingMediaStatus(false);
rtc::CritScope cs(&_callbackCritSect);
channel_state_.SetSending(false);
@ -1023,9 +998,7 @@ void Channel::StopSend() {
// Reset sending SSRC and sequence number and triggers direct transmission
// of RTCP BYE
if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"StartSend() RTP/RTCP failed to stop sending");
LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
}
_rtpRtcpModule->SetSendingMediaStatus(false);
}
@ -1198,39 +1171,9 @@ void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms,
});
}
int32_t Channel::RegisterExternalTransport(Transport* transport) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterExternalTransport()");
void Channel::RegisterTransport(Transport* transport) {
rtc::CritScope cs(&_callbackCritSect);
if (_externalTransport) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceError,
"RegisterExternalTransport() external transport already enabled");
return -1;
}
_externalTransport = true;
_transportPtr = transport;
return 0;
}
int32_t Channel::DeRegisterExternalTransport() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::DeRegisterExternalTransport()");
rtc::CritScope cs(&_callbackCritSect);
if (_transportPtr) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"DeRegisterExternalTransport() all transport is disabled");
} else {
_engineStatisticsPtr->SetLastError(
VE_INVALID_OPERATION, kTraceWarning,
"DeRegisterExternalTransport() external transport already "
"disabled");
}
_externalTransport = false;
_transportPtr = NULL;
return 0;
}
void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
@ -1380,9 +1323,7 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
}
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
_engineStatisticsPtr->SetLastError(
VE_SEND_DTMF_FAILED, kTraceWarning,
"SendTelephoneEventOutband() failed to send event");
LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
return -1;
}
return 0;
@ -1401,10 +1342,8 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type,
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetSendTelephoneEventPayloadType() failed to register send"
"payload type");
LOG(LS_ERROR) << "SetSendTelephoneEventPayloadType() failed to register "
"send payload type";
return -1;
}
}
@ -1415,8 +1354,7 @@ int Channel::SetLocalSSRC(unsigned int ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetLocalSSRC()");
if (channel_state_.Get().sending) {
_engineStatisticsPtr->SetLastError(VE_ALREADY_SENDING, kTraceError,
"SetLocalSSRC() already sending");
LOG(LS_ERROR) << "SetLocalSSRC() already sending";
return -1;
}
_rtpRtcpModule->SetSSRC(ssrc);
@ -1517,9 +1455,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCP_CNAME()");
if (_rtpRtcpModule->SetCNAME(cName) != 0) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"SetRTCP_CNAME() failed to set RTCP CNAME");
LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
return -1;
}
return 0;
@ -1528,9 +1464,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
int Channel::GetRemoteRTCPReportBlocks(
std::vector<ReportBlock>* report_blocks) {
if (report_blocks == NULL) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteRTCPReportBlock()s invalid report_blocks.");
LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
return -1;
}
@ -1788,15 +1722,11 @@ int Channel::SetMinimumPlayoutDelay(int delayMs) {
"Channel::SetMinimumPlayoutDelay()");
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetMinimumPlayoutDelay() invalid min delay");
LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
return -1;
}
if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetMinimumPlayoutDelay() failed to set min playout delay");
LOG(LS_ERROR) << "SetMinimumPlayoutDelay() failed to set min playout delay";
return -1;
}
return 0;
@ -1809,9 +1739,7 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
playout_timestamp_rtp = playout_timestamp_rtp_;
}
if (playout_timestamp_rtp == 0) {
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_VALUE, kTraceStateInfo,
"GetPlayoutTimestamp() failed to retrieve timestamp");
LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
return -1;
}
timestamp = playout_timestamp_rtp;
@ -1839,9 +1767,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() failed to read playout"
" delay from the ADM");
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
"UpdatePlayoutTimestamp() failed to retrieve playout delay");
return;
}

15
voice_engine/channel.h
View File

@ -90,7 +90,6 @@ namespace voe {
class RtcEventLogProxy;
class RtcpRttStatsProxy;
class RtpPacketSenderProxy;
class Statistics;
class TransportFeedbackProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
@ -157,10 +156,8 @@ class Channel
const VoEBase::ChannelConfig& config);
int32_t Init();
void Terminate();
int32_t SetEngineInformation(Statistics& engineStatistics,
ProcessThread& moduleProcessThread,
int32_t SetEngineInformation(ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
rtc::CriticalSection* callbackCritSect,
rtc::TaskQueue* encoder_queue);
void SetSink(std::unique_ptr<AudioSinkInterface> sink);
@ -197,8 +194,7 @@ class Channel
int max_frame_length_ms);
// Network
int32_t RegisterExternalTransport(Transport* transport);
int32_t DeRegisterExternalTransport();
void RegisterTransport(Transport* transport);
// TODO(nisse, solenberg): Delete when VoENetwork is deleted.
int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
void OnRtpPacket(const RtpPacketReceived& packet);
@ -286,10 +282,6 @@ class Channel
int32_t ChannelId() const { return _channelId; }
bool Playing() const { return channel_state_.Get().playing; }
bool Sending() const { return channel_state_.Get().sending; }
bool ExternalTransport() const {
rtc::CritScope cs(&_callbackCritSect);
return _externalTransport;
}
RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
@ -395,7 +387,6 @@ class Channel
acm2::RentACodec rent_a_codec_;
std::unique_ptr<AudioSinkInterface> audio_sink_;
AudioLevel _outputAudioLevel;
bool _externalTransport;
// Downsamples to the codec rate if necessary.
PushResampler<int16_t> input_resampler_;
uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
@ -420,10 +411,8 @@ class Channel
int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
// uses
Statistics* _engineStatisticsPtr;
ProcessThread* _moduleProcessThreadPtr;
AudioDeviceModule* _audioDeviceModulePtr;
rtc::CriticalSection* _callbackCritSectPtr; // owned by base
Transport* _transportPtr; // WebRtc socket or external transport
RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);

10
voice_engine/channel_proxy.cc
View File

@ -213,15 +213,9 @@ void ChannelProxy::SetInputMute(bool muted) {
channel()->SetInputMute(muted);
}
void ChannelProxy::RegisterExternalTransport(Transport* transport) {
void ChannelProxy::RegisterTransport(Transport* transport) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->RegisterExternalTransport(transport);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::DeRegisterExternalTransport() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->DeRegisterExternalTransport();
channel()->RegisterTransport(transport);
}
void ChannelProxy::OnRtpPacket(const RtpPacketReceived& packet) {

3
voice_engine/channel_proxy.h
View File

@ -94,8 +94,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
virtual void SetInputMute(bool muted);
virtual void RegisterExternalTransport(Transport* transport);
virtual void DeRegisterExternalTransport();
virtual void RegisterTransport(Transport* transport);
// Implements RtpPacketSinkInterface
void OnRtpPacket(const RtpPacketReceived& packet) override;

15
voice_engine/shared_data.cc
View File

@ -24,7 +24,6 @@ static int32_t _gInstanceCounter = 0;
SharedData::SharedData()
: _instanceId(++_gInstanceCounter),
_channelManager(_gInstanceCounter),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
_moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
encoder_queue_("AudioEncoderQueue") {
@ -84,20 +83,6 @@ int SharedData::NumOfPlayingChannels() {
return playout_channels;
}
void SharedData::SetLastError(int32_t error) const {
_engineStatistics.SetLastError(error);
}
void SharedData::SetLastError(int32_t error,
TraceLevel level) const {
_engineStatistics.SetLastError(error, level);
}
void SharedData::SetLastError(int32_t error, TraceLevel level,
const char* msg) const {
_engineStatistics.SetLastError(error, level, msg);
}
} // namespace voe
} // namespace webrtc

9
voice_engine/shared_data.h
View File

@ -22,7 +22,6 @@
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/channel_manager.h"
#include "voice_engine/statistics.h"
#include "voice_engine/voice_engine_defines.h"
class ProcessThread;
@ -37,7 +36,6 @@ class SharedData
public:
// Public accessors.
uint32_t instance_id() const { return _instanceId; }
Statistics& statistics() { return _engineStatistics; }
ChannelManager& channel_manager() { return _channelManager; }
AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); }
void set_audio_device(
@ -51,18 +49,11 @@ public:
int NumOfSendingChannels();
int NumOfPlayingChannels();
// Convenience methods for calling statistics().SetLastError().
void SetLastError(int32_t error) const;
void SetLastError(int32_t error, TraceLevel level) const;
void SetLastError(int32_t error, TraceLevel level,
const char* msg) const;
protected:
rtc::ThreadChecker construction_thread_;
const uint32_t _instanceId;
rtc::CriticalSection _apiCritPtr;
ChannelManager _channelManager;
Statistics _engineStatistics;
rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
TransmitMixer* _transmitMixerPtr;
std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;

86
voice_engine/statistics.cc
View File

@ -1,86 +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 <assert.h>
#include <stdio.h>
#include "voice_engine/statistics.h"
#include "system_wrappers/include/trace.h"
namespace webrtc {
namespace voe {
Statistics::Statistics(uint32_t instanceId) :
_instanceId(instanceId),
_lastError(0),
_isInitialized(false)
{
}
Statistics::~Statistics()
{
}
int32_t Statistics::SetInitialized()
{
_isInitialized = true;
return 0;
}
int32_t Statistics::SetUnInitialized()
{
_isInitialized = false;
return 0;
}
bool Statistics::Initialized() const
{
return _isInitialized;
}
int32_t Statistics::SetLastError(int32_t error) const
{
rtc::CritScope cs(&lock_);
_lastError = error;
return 0;
}
int32_t Statistics::SetLastError(int32_t error,
TraceLevel level) const
{
WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
"error code is set to %d",
error);
rtc::CritScope cs(&lock_);
_lastError = error;
return 0;
}
int32_t Statistics::SetLastError(
int32_t error,
TraceLevel level, const char* msg) const
{
char traceMessage[KTraceMaxMessageSize];
assert(strlen(msg) < KTraceMaxMessageSize);
sprintf(traceMessage, "%s (error=%d)", msg, error);
WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1), "%s",
traceMessage);
rtc::CritScope cs(&lock_);
_lastError = error;
return 0;
}
} // namespace voe
} // namespace webrtc

51
voice_engine/statistics.h
View File

@ -1,51 +0,0 @@
/*
* Copyright (c) 2011 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 VOICE_ENGINE_STATISTICS_H_
#define VOICE_ENGINE_STATISTICS_H_
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/criticalsection.h"
#include "typedefs.h" // NOLINT(build/include)
#include "voice_engine/include/voe_errors.h"
#include "voice_engine/voice_engine_defines.h"
namespace webrtc {
namespace voe {
class Statistics
{
public:
enum {KTraceMaxMessageSize = 256};
public:
Statistics(uint32_t instanceId);
~Statistics();
int32_t SetInitialized();
int32_t SetUnInitialized();
bool Initialized() const;
int32_t SetLastError(int32_t error) const;
int32_t SetLastError(int32_t error, TraceLevel level) const;
int32_t SetLastError(int32_t error,
TraceLevel level,
const char* msg) const;
private:
rtc::CriticalSection lock_;
const uint32_t _instanceId;
mutable int32_t _lastError;
bool _isInitialized;
};
} // namespace voe
} // namespace webrtc
#endif // VOICE_ENGINE_STATISTICS_H_

1
voice_engine/transmit_mixer.cc
View File

@ -20,7 +20,6 @@
#include "system_wrappers/include/trace.h"
#include "voice_engine/channel.h"
#include "voice_engine/channel_manager.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
namespace webrtc {

1
voice_engine/transmit_mixer.h
View File

@ -36,7 +36,6 @@ namespace voe {
class ChannelManager;
class MixedAudio;
class Statistics;
class TransmitMixer {
public:

137
voice_engine/voe_base_impl.cc
View File

@ -42,7 +42,6 @@ VoEBaseImpl::~VoEBaseImpl() {
}
void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
rtc::CritScope cs(&callbackCritSect_);
if (error == AudioDeviceObserver::kRecordingError) {
LOG_F(LS_ERROR) << "VE_RUNTIME_REC_ERROR";
} else if (error == AudioDeviceObserver::kPlayoutError) {
@ -51,7 +50,6 @@ void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
}
void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) {
rtc::CritScope cs(&callbackCritSect_);
if (warning == AudioDeviceObserver::kRecordingWarning) {
LOG_F(LS_WARNING) << "VE_RUNTIME_REC_WARNING";
} else if (warning == AudioDeviceObserver::kPlayoutWarning) {
@ -167,9 +165,6 @@ int VoEBaseImpl::Init(
RTC_DCHECK(audio_processing);
rtc::CritScope cs(shared_->crit_sec());
WebRtcSpl_Init();
if (shared_->statistics().Initialized()) {
return 0;
}
if (shared_->process_thread()) {
shared_->process_thread()->Start();
}
@ -185,8 +180,7 @@ int VoEBaseImpl::Init(
VoEId(shared_->instance_id(), -1),
AudioDeviceModule::kPlatformDefaultAudio));
if (shared_->audio_device() == nullptr) {
shared_->SetLastError(VE_NO_MEMORY, kTraceCritical,
"Init() failed to create the ADM");
LOG(LS_ERROR) << "Init() failed to create the ADM";
return -1;
}
#endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
@ -211,55 +205,44 @@ int VoEBaseImpl::Init(
// Register the AudioObserver implementation
if (shared_->audio_device()->RegisterEventObserver(this) != 0) {
shared_->SetLastError(
VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"Init() failed to register event observer for the ADM");
LOG(LS_ERROR) << "Init() failed to register event observer for the ADM";
}
// Register the AudioTransport implementation
if (shared_->audio_device()->RegisterAudioCallback(this) != 0) {
shared_->SetLastError(
VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"Init() failed to register audio callback for the ADM");
LOG(LS_ERROR) << "Init() failed to register audio callback for the ADM";
}
// ADM initialization
if (shared_->audio_device()->Init() != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"Init() failed to initialize the ADM");
LOG(LS_ERROR) << "Init() failed to initialize the ADM";
return -1;
}
// Initialize the default speaker
if (shared_->audio_device()->SetPlayoutDevice(
WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
"Init() failed to set the default output device");
LOG(LS_ERROR) << "Init() failed to set the default output device";
}
if (shared_->audio_device()->InitSpeaker() != 0) {
shared_->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
"Init() failed to initialize the speaker");
LOG(LS_ERROR) << "Init() failed to initialize the speaker";
}
// Initialize the default microphone
if (shared_->audio_device()->SetRecordingDevice(
WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
"Init() failed to set the default input device");
LOG(LS_ERROR) << "Init() failed to set the default input device";
}
if (shared_->audio_device()->InitMicrophone() != 0) {
shared_->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
"Init() failed to initialize the microphone");
LOG(LS_ERROR) << "Init() failed to initialize the microphone";
}
// Set number of channels
if (shared_->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to query stereo playout mode");
LOG(LS_ERROR) << "Init() failed to query stereo playout mode";
}
if (shared_->audio_device()->SetStereoPlayout(available) != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo playout mode");
LOG(LS_ERROR) << "Init() failed to set mono/stereo playout mode";
}
// TODO(andrew): These functions don't tell us whether stereo recording
@ -271,14 +254,11 @@ int VoEBaseImpl::Init(
// http://code.google.com/p/webrtc/issues/detail?id=204
shared_->audio_device()->StereoRecordingIsAvailable(&available);
if (shared_->audio_device()->SetStereoRecording(available) != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo recording mode");
LOG(LS_ERROR) << "Init() failed to set mono/stereo recording mode";
}
shared_->set_audio_processing(audio_processing);
// Set the error state for any failures in this block.
shared_->SetLastError(VE_APM_ERROR);
// Configure AudioProcessing components.
// TODO(peah): Move this initialization to webrtcvoiceengine.cc.
if (audio_processing->high_pass_filter()->Enable(true) != 0) {
@ -309,14 +289,12 @@ int VoEBaseImpl::Init(
LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
return -1;
}
shared_->SetLastError(0); // Clear error state.
#ifdef WEBRTC_VOICE_ENGINE_AGC
bool agc_enabled =
agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR);
// TODO(ajm): No error return here due to
// https://code.google.com/p/webrtc/issues/detail?id=1464
}
@ -327,7 +305,7 @@ int VoEBaseImpl::Init(
else
decoder_factory_ = CreateBuiltinAudioDecoderFactory();
return shared_->statistics().SetInitialized();
return 0;
}
int VoEBaseImpl::Terminate() {
@ -341,11 +319,6 @@ int VoEBaseImpl::CreateChannel() {
int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
rtc::CritScope cs(shared_->crit_sec());
if (!shared_->statistics().Initialized()) {
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
ChannelConfig config_copy(config);
config_copy.acm_config.decoder_factory = decoder_factory_;
voe::ChannelOwner channel_owner =
@ -355,21 +328,16 @@ int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
if (channel_owner->channel()->SetEngineInformation(
shared_->statistics(),
*shared_->process_thread(), *shared_->audio_device(),
&callbackCritSect_, shared_->encoder_queue()) != 0) {
shared_->SetLastError(
VE_CHANNEL_NOT_CREATED, kTraceError,
"CreateChannel() failed to associate engine and channel."
" Destroying channel.");
shared_->encoder_queue()) != 0) {
LOG(LS_ERROR) << "CreateChannel() failed to associate engine and channel."
" Destroying channel.";
shared_->channel_manager().DestroyChannel(
channel_owner->channel()->ChannelId());
return -1;
} else if (channel_owner->channel()->Init() != 0) {
shared_->SetLastError(
VE_CHANNEL_NOT_CREATED, kTraceError,
"CreateChannel() failed to initialize channel. Destroying"
" channel.");
LOG(LS_ERROR) << "CreateChannel() failed to initialize channel. Destroying"
" channel.";
shared_->channel_manager().DestroyChannel(
channel_owner->channel()->ChannelId());
return -1;
@ -379,17 +347,11 @@ int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
int VoEBaseImpl::DeleteChannel(int channel) {
rtc::CritScope cs(shared_->crit_sec());
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 == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"DeleteChannel() failed to locate channel");
LOG(LS_ERROR) << "DeleteChannel() failed to locate channel";
return -1;
}
}
@ -406,23 +368,17 @@ int VoEBaseImpl::DeleteChannel(int channel) {
int VoEBaseImpl::StartPlayout(int channel) {
rtc::CritScope cs(shared_->crit_sec());
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 == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StartPlayout() failed to locate channel");
LOG(LS_ERROR) << "StartPlayout() failed to locate channel";
return -1;
}
if (channelPtr->Playing()) {
return 0;
}
if (StartPlayout() != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"StartPlayout() failed to start playout");
LOG(LS_ERROR) << "StartPlayout() failed to start playout";
return -1;
}
return channelPtr->StartPlayout();
@ -430,15 +386,10 @@ int VoEBaseImpl::StartPlayout(int channel) {
int VoEBaseImpl::StopPlayout(int channel) {
rtc::CritScope cs(shared_->crit_sec());
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 == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StopPlayout() failed to locate channel");
LOG(LS_ERROR) << "StopPlayout() failed to locate channel";
return -1;
}
if (channelPtr->StopPlayout() != 0) {
@ -450,23 +401,17 @@ int VoEBaseImpl::StopPlayout(int channel) {
int VoEBaseImpl::StartSend(int channel) {
rtc::CritScope cs(shared_->crit_sec());
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 == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StartSend() failed to locate channel");
LOG(LS_ERROR) << "StartSend() failed to locate channel";
return -1;
}
if (channelPtr->Sending()) {
return 0;
}
if (StartSend() != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"StartSend() failed to start recording");
LOG(LS_ERROR) << "StartSend() failed to start recording";
return -1;
}
return channelPtr->StartSend();
@ -474,15 +419,10 @@ int VoEBaseImpl::StartSend(int channel) {
int VoEBaseImpl::StopSend(int channel) {
rtc::CritScope cs(shared_->crit_sec());
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 == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StopSend() failed to locate channel");
LOG(LS_ERROR) << "StopSend() failed to locate channel";
return -1;
}
channelPtr->StopSend();
@ -507,8 +447,7 @@ int32_t VoEBaseImpl::StopPlayout() {
// Stop audio-device playing if no channel is playing out
if (shared_->NumOfPlayingChannels() == 0) {
if (shared_->audio_device()->StopPlayout() != 0) {
shared_->SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError,
"StopPlayout() failed to stop playout");
LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
return -1;
}
}
@ -536,8 +475,7 @@ int32_t VoEBaseImpl::StopSend() {
if (shared_->NumOfSendingChannels() == 0) {
// Stop audio-device recording if no channel is recording
if (shared_->audio_device()->StopRecording() != 0) {
shared_->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
"StopSend() failed to stop recording");
LOG(LS_ERROR) << "StopSend() failed to stop recording";
return -1;
}
shared_->transmit_mixer()->StopSend();
@ -559,34 +497,27 @@ int32_t VoEBaseImpl::TerminateInternal() {
if (shared_->audio_device()) {
if (shared_->audio_device()->StopPlayout() != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"TerminateInternal() failed to stop playout");
LOG(LS_ERROR) << "TerminateInternal() failed to stop playout";
}
if (shared_->audio_device()->StopRecording() != 0) {
shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"TerminateInternal() failed to stop recording");
LOG(LS_ERROR) << "TerminateInternal() failed to stop recording";
}
if (shared_->audio_device()->RegisterEventObserver(nullptr) != 0) {
shared_->SetLastError(
VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"TerminateInternal() failed to de-register event observer "
"for the ADM");
LOG(LS_ERROR) << "TerminateInternal() failed to de-register event "
"observer for the ADM";
}
if (shared_->audio_device()->RegisterAudioCallback(nullptr) != 0) {
shared_->SetLastError(
VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"TerminateInternal() failed to de-register audio callback "
"for the ADM");
LOG(LS_ERROR) << "TerminateInternal() failed to de-register audio "
"callback for the ADM";
}
if (shared_->audio_device()->Terminate() != 0) {
shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"TerminateInternal() failed to terminate the ADM");
LOG(LS_ERROR) << "TerminateInternal() failed to terminate the ADM";
}
shared_->set_audio_device(nullptr);
}
shared_->set_audio_processing(nullptr);
return shared_->statistics().SetUnInitialized();
return 0;
}
} // namespace webrtc

1
voice_engine/voe_base_impl.h
View File

@ -104,7 +104,6 @@ class VoEBaseImpl : public VoEBase,
// Initialize channel by setting Engine Information then initializing
// channel.
int InitializeChannel(voe::ChannelOwner* channel_owner);
rtc::CriticalSection callbackCritSect_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
AudioFrame audioFrame_;

5
voice_engine/voe_base_unittest.cc
View File

@ -41,11 +41,6 @@ TEST_F(VoEBaseTest, InitWithExternalAudioDevice) {
EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
}
TEST_F(VoEBaseTest, CreateChannelBeforeInitShouldFail) {
int channelID = base_->CreateChannel();
EXPECT_EQ(channelID, -1);
}
TEST_F(VoEBaseTest, CreateChannelAfterInit) {
EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
int channelID = base_->CreateChannel();

1
voice_engine/voice_engine_impl.cc
View File

@ -65,7 +65,6 @@ std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
int channel_id) {
RTC_DCHECK(channel_id >= 0);
rtc::CritScope cs(crit_sec());
RTC_DCHECK(statistics().Initialized());
return std::unique_ptr<voe::ChannelProxy>(
new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
}