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Clean out unused methods from VoiceEngine and VoEBase.

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BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3018523002
Cr-Commit-Position: refs/heads/master@{#19880}
This commit is contained in:
solenberg 2017-09-18 01:57:01 -07:00 committed by Commit Bot
parent 2c9ac29c5b
commit 35dee81321
18 changed files with 30 additions and 325 deletions
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1
media/BUILD.gn
View File

@ -160,7 +160,6 @@ rtc_static_library("rtc_audio_video") {
"engine/videodecodersoftwarefallbackwrapper.h",
"engine/videoencodersoftwarefallbackwrapper.cc",
"engine/videoencodersoftwarefallbackwrapper.h",
"engine/webrtccommon.h",
"engine/webrtcmediaengine.cc",
"engine/webrtcmediaengine.h",
"engine/webrtcvideocapturer.cc",

6
media/engine/fakewebrtcvoiceengine.h
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@ -98,16 +98,10 @@ class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
channels_.erase(channel);
return 0;
}
WEBRTC_STUB(StartReceive, (int channel));
WEBRTC_STUB(StartPlayout, (int channel));
WEBRTC_STUB(StartSend, (int channel));
WEBRTC_STUB(StopReceive, (int channel));
WEBRTC_STUB(StopPlayout, (int channel));
WEBRTC_STUB(StopSend, (int channel));
WEBRTC_STUB(GetVersion, (char version[1024]));
WEBRTC_STUB(LastError, ());
WEBRTC_STUB(AssociateSendChannel, (int channel,
int accociate_send_channel));
size_t GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);

58
media/engine/webrtccommon.h
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@ -1,58 +0,0 @@
/*
* Copyright (c) 2004 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 MEDIA_ENGINE_WEBRTCCOMMON_H_
#define MEDIA_ENGINE_WEBRTCCOMMON_H_
#include "common_types.h" // NOLINT(build/include)
namespace cricket {
// Tracing helpers, for easy logging when WebRTC calls fail.
// Example: "LOG_RTCERR1(StartSend, channel);" produces the trace
// "StartSend(1) failed, err=XXXX"
// The method GetLastEngineError must be defined in the calling scope.
#define LOG_RTCERR0(func) \
LOG_RTCERR0_EX(func, GetLastEngineError())
#define LOG_RTCERR1(func, a1) \
LOG_RTCERR1_EX(func, a1, GetLastEngineError())
#define LOG_RTCERR2(func, a1, a2) \
LOG_RTCERR2_EX(func, a1, a2, GetLastEngineError())
#define LOG_RTCERR3(func, a1, a2, a3) \
LOG_RTCERR3_EX(func, a1, a2, a3, GetLastEngineError())
#define LOG_RTCERR4(func, a1, a2, a3, a4) \
LOG_RTCERR4_EX(func, a1, a2, a3, a4, GetLastEngineError())
#define LOG_RTCERR5(func, a1, a2, a3, a4, a5) \
LOG_RTCERR5_EX(func, a1, a2, a3, a4, a5, GetLastEngineError())
#define LOG_RTCERR6(func, a1, a2, a3, a4, a5, a6) \
LOG_RTCERR6_EX(func, a1, a2, a3, a4, a5, a6, GetLastEngineError())
#define LOG_RTCERR0_EX(func, err) LOG(LS_WARNING) \
<< "" << #func << "() failed, err=" << err
#define LOG_RTCERR1_EX(func, a1, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ") failed, err=" << err
#define LOG_RTCERR2_EX(func, a1, a2, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ", " << a2 << ") failed, err=" \
<< err
#define LOG_RTCERR3_EX(func, a1, a2, a3, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
<< ") failed, err=" << err
#define LOG_RTCERR4_EX(func, a1, a2, a3, a4, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
<< ", " << a4 << ") failed, err=" << err
#define LOG_RTCERR5_EX(func, a1, a2, a3, a4, a5, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
<< ", " << a4 << ", " << a5 << ") failed, err=" << err
#define LOG_RTCERR6_EX(func, a1, a2, a3, a4, a5, a6, err) LOG(LS_WARNING) \
<< "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
<< ", " << a4 << ", " << a5 << ", " << a6 << ") failed, err=" << err
} // namespace cricket
#endif // MEDIA_ENGINE_WEBRTCCOMMON_H_

3
media/engine/webrtcvoe.h
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@ -13,8 +13,6 @@
#include <memory>
#include "media/engine/webrtccommon.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_device/include/audio_device.h"
#include "voice_engine/include/voe_base.h"
@ -83,7 +81,6 @@ class VoEWrapper {
~VoEWrapper() {}
webrtc::VoiceEngine* engine() const { return engine_.get(); }
webrtc::VoEBase* base() const { return base_.get(); }
int error() { return base_->LastError(); }
private:
scoped_voe_engine engine_;

18
media/engine/webrtcvoiceengine.cc
View File

@ -290,7 +290,6 @@ void WebRtcVoiceEngine::Init() {
// Temporarily turn logging level up for the Init() call.
webrtc::Trace::SetTraceCallback(this);
webrtc::Trace::set_level_filter(kElevatedTraceFilter);
LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
RTC_CHECK_EQ(0,
voe_wrapper_->base()->Init(adm_.get(), apm(), decoder_factory_));
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
@ -628,14 +627,18 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
LOG(LS_INFO) << "Recording sample rate is "
<< *options.recording_sample_rate;
if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
LOG(LS_WARNING) << "SetRecordingSampleRate("
<< *options.recording_sample_rate << ") failed, err="
<< adm()->LastError();
}
}
if (options.playout_sample_rate) {
LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
LOG(LS_WARNING) << "SetPlayoutSampleRate("
<< *options.playout_sample_rate << ") failed, err="
<< adm()->LastError();
}
}
return true;
@ -673,11 +676,6 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
return capabilities;
}
int WebRtcVoiceEngine::GetLastEngineError() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->error();
}
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
// Note: This callback can happen on any thread!
@ -1832,7 +1830,7 @@ bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
int WebRtcVoiceMediaChannel::CreateVoEChannel() {
int id = engine()->CreateVoEChannel();
if (id == -1) {
LOG_RTCERR0(CreateVoEChannel);
LOG(LS_WARNING) << "CreateVoEChannel() failed.";
return -1;
}
@ -1841,7 +1839,7 @@ int WebRtcVoiceMediaChannel::CreateVoEChannel() {
bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
LOG_RTCERR1(DeleteChannel, channel);
LOG(LS_WARNING) << "DeleteChannel(" << channel << ") failed.";
return false;
}
return true;

3
media/engine/webrtcvoiceengine.h
View File

@ -22,7 +22,6 @@
#include "call/call.h"
#include "media/base/rtputils.h"
#include "media/engine/apm_helpers.h"
#include "media/engine/webrtccommon.h"
#include "media/engine/webrtcvoe.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "pc/channel.h"
@ -89,7 +88,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
VoEWrapper* voe() { return voe_wrapper_.get(); }
int GetLastEngineError();
// Starts AEC dump using an existing file. A maximum file size in bytes can be
// specified. When the maximum file size is reached, logging is stopped and
@ -249,7 +247,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
bool MuteStream(uint32_t ssrc, bool mute);
WebRtcVoiceEngine* engine() { return engine_; }
int GetLastEngineError() { return engine()->GetLastEngineError(); }
void ChangePlayout(bool playout);
int CreateVoEChannel();
bool DeleteVoEChannel(int channel);

6
test/mock_voice_engine.h
View File

@ -107,17 +107,11 @@ class MockVoiceEngine : public VoiceEngineImpl {
MOCK_METHOD0(CreateChannel, int());
MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config));
MOCK_METHOD1(DeleteChannel, int(int channel));
MOCK_METHOD1(StartReceive, int(int channel));
MOCK_METHOD1(StopReceive, int(int channel));
MOCK_METHOD1(StartPlayout, int(int channel));
MOCK_METHOD1(StopPlayout, int(int channel));
MOCK_METHOD1(StartSend, int(int channel));
MOCK_METHOD1(StopSend, int(int channel));
MOCK_METHOD1(GetVersion, int(char version[1024]));
MOCK_METHOD0(LastError, int());
MOCK_METHOD0(audio_transport, AudioTransport*());
MOCK_METHOD2(AssociateSendChannel,
int(int channel, int accociate_send_channel));
// VoECodec
MOCK_METHOD0(NumOfCodecs, int());

33
voice_engine/include/voe_base.h
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@ -73,20 +73,6 @@ class WEBRTC_DLLEXPORT VoiceEngine {
// references have been released.
static bool Delete(VoiceEngine*& voiceEngine);
// Specifies the amount and type of trace information which will be
// created by the VoiceEngine.
static int SetTraceFilter(unsigned int filter);
// Sets the name of the trace file and enables non-encrypted trace messages.
static int SetTraceFile(const char* fileNameUTF8,
bool addFileCounter = false);
// Installs the TraceCallback implementation to ensure that the user
// receives callbacks for generated trace messages.
static int SetTraceCallback(TraceCallback* callback);
static std::string GetVersionString();
protected:
VoiceEngine() {}
~VoiceEngine() {}
@ -158,13 +144,6 @@ class WEBRTC_DLLEXPORT VoEBase {
// Returns -1 in case of an error, 0 otherwise.
virtual int DeleteChannel(int channel) = 0;
// Prepares and initiates the VoiceEngine for reception of
// incoming RTP/RTCP packets on the specified |channel|.
virtual int StartReceive(int channel) = 0;
// Stops receiving incoming RTP/RTCP packets on the specified |channel|.
virtual int StopReceive(int channel) { return 0; }
// Starts forwarding the packets to the mixer/soundcard for a
// specified |channel|.
virtual int StartPlayout(int channel) = 0;
@ -180,22 +159,10 @@ class WEBRTC_DLLEXPORT VoEBase {
// Stops sending packets from a specified |channel|.
virtual int StopSend(int channel) = 0;
// Gets the version information for VoiceEngine and its components.
virtual int GetVersion(char version[1024]) = 0;
// Gets the last VoiceEngine error code.
virtual int LastError() = 0;
// TODO(xians): Make the interface pure virtual after libjingle
// implements the interface in its FakeWebRtcVoiceEngine.
virtual AudioTransport* audio_transport() { return NULL; }
// Associate a send channel to a receive channel.
// Used for obtaining RTT for a receive-only channel.
// One should be careful not to crate a circular association, e.g.,
// 1 <- 2 <- 1.
virtual int AssociateSendChannel(int channel, int accociate_send_channel) = 0;
protected:
VoEBase() {}
virtual ~VoEBase() {}

29
voice_engine/mock/mock_voe_observer.h
View File

@ -1,29 +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 VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
#define VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
#include "test/gmock.h"
#include "voice_engine/include/voe_base.h"
namespace webrtc {
class MockVoEObserver: public VoiceEngineObserver {
public:
MockVoEObserver() {}
virtual ~MockVoEObserver() {}
MOCK_METHOD2(CallbackOnError, void(int channel, int error_code));
};
}
#endif // VOICE_ENGINE_MOCK_VOE_OBSERVER_H_

12
voice_engine/statistics.cc
View File

@ -81,18 +81,6 @@ int32_t Statistics::SetLastError(
return 0;
}
int32_t Statistics::LastError() const
{
int32_t ret;
{
rtc::CritScope cs(&lock_);
ret = _lastError;
}
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
"LastError() => %d", ret);
return ret;
}
} // namespace voe
} // namespace webrtc

1
voice_engine/statistics.h
View File

@ -36,7 +36,6 @@ class Statistics
int32_t SetLastError(int32_t error,
TraceLevel level,
const char* msg) const;
int32_t LastError() const;
private:
rtc::CriticalSection lock_;

61
voice_engine/voe_base_impl.cc
View File

@ -468,22 +468,6 @@ int VoEBaseImpl::DeleteChannel(int channel) {
return 0;
}
int VoEBaseImpl::StartReceive(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,
"StartReceive() failed to locate channel");
return -1;
}
return 0;
}
int VoEBaseImpl::StartPlayout(int channel) {
rtc::CritScope cs(shared_->crit_sec());
if (!shared_->statistics().Initialized()) {
@ -569,22 +553,6 @@ int VoEBaseImpl::StopSend(int channel) {
return StopSend();
}
int VoEBaseImpl::GetVersion(char version[1024]) {
if (version == nullptr) {
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError);
return -1;
}
std::string versionString = VoiceEngine::GetVersionString();
RTC_DCHECK_GT(1024, versionString.size() + 1);
char* end = std::copy(versionString.cbegin(), versionString.cend(), version);
end[0] = '\n';
end[1] = '\0';
return 0;
}
int VoEBaseImpl::LastError() { return (shared_->statistics().LastError()); }
int32_t VoEBaseImpl::StartPlayout() {
if (!shared_->audio_device()->Playing()) {
if (shared_->audio_device()->InitPlayout() != 0) {
@ -716,33 +684,4 @@ void VoEBaseImpl::GetPlayoutData(int sample_rate, size_t number_of_channels,
*ntp_time_ms = audioFrame_.ntp_time_ms_;
}
int VoEBaseImpl::AssociateSendChannel(int channel,
int accociate_send_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* channel_ptr = ch.channel();
if (channel_ptr == NULL) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"AssociateSendChannel() failed to locate channel");
return -1;
}
ch = shared_->channel_manager().GetChannel(accociate_send_channel);
voe::Channel* accociate_send_channel_ptr = ch.channel();
if (accociate_send_channel_ptr == NULL) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"AssociateSendChannel() failed to locate accociate_send_channel");
return -1;
}
channel_ptr->set_associate_send_channel(ch);
return 0;
}
} // namespace webrtc

7
voice_engine/voe_base_impl.h
View File

@ -44,20 +44,13 @@ class VoEBaseImpl : public VoEBase,
int CreateChannel(const ChannelConfig& config) override;
int DeleteChannel(int channel) override;
int StartReceive(int channel) override;
int StartPlayout(int channel) override;
int StartSend(int channel) override;
int StopPlayout(int channel) override;
int StopSend(int channel) override;
int GetVersion(char version[1024]) override;
int LastError() override;
AudioTransport* audio_transport() override { return this; }
int AssociateSendChannel(int channel, int accociate_send_channel) override;
// AudioTransport
int32_t RecordedDataIsAvailable(const void* audio_data,
const size_t number_of_frames,

65
voice_engine/voe_base_unittest.cc
View File

@ -10,20 +10,35 @@
#include "voice_engine/include/voe_base.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_device/include/fake_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "test/gtest.h"
#include "voice_engine/channel_manager.h"
#include "voice_engine/shared_data.h"
#include "voice_engine/voice_engine_fixture.h"
#include "voice_engine/voice_engine_impl.h"
namespace webrtc {
class VoEBaseTest : public VoiceEngineFixture {};
class VoEBaseTest : public ::testing::Test {
protected:
VoEBaseTest()
: voe_(VoiceEngine::Create()),
base_(VoEBase::GetInterface(voe_)) {
EXPECT_NE(nullptr, base_);
apm_ = new rtc::RefCountedObject<test::MockAudioProcessing>();
}
~VoEBaseTest() {
EXPECT_EQ(0, base_->Terminate());
EXPECT_EQ(1, base_->Release());
EXPECT_TRUE(VoiceEngine::Delete(voe_));
}
VoiceEngine* voe_;
VoEBase* base_;
FakeAudioDeviceModule adm_;
rtc::scoped_refptr<AudioProcessing> apm_;
};
TEST_F(VoEBaseTest, InitWithExternalAudioDevice) {
EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
EXPECT_EQ(0, base_->LastError());
}
TEST_F(VoEBaseTest, CreateChannelBeforeInitShouldFail) {
@ -38,40 +53,4 @@ TEST_F(VoEBaseTest, CreateChannelAfterInit) {
EXPECT_EQ(0, base_->DeleteChannel(channelID));
}
TEST_F(VoEBaseTest, AssociateSendChannel) {
EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
const int channel_1 = base_->CreateChannel();
// Associating with a channel that does not exist should fail.
EXPECT_EQ(-1, base_->AssociateSendChannel(channel_1, channel_1 + 1));
const int channel_2 = base_->CreateChannel();
// Let the two channels associate with each other. This is not a normal use
// case. Actually, circular association should be avoided in practice. This
// is just to test that no crash is caused.
EXPECT_EQ(0, base_->AssociateSendChannel(channel_1, channel_2));
EXPECT_EQ(0, base_->AssociateSendChannel(channel_2, channel_1));
voe::SharedData* shared_data = static_cast<voe::SharedData*>(
static_cast<VoiceEngineImpl*>(voe_));
voe::ChannelOwner reference = shared_data->channel_manager()
.GetChannel(channel_1);
EXPECT_EQ(0, base_->DeleteChannel(channel_1));
// Make sure that the only use of the channel-to-delete is |reference|
// at this point.
EXPECT_EQ(1, reference.use_count());
reference = shared_data->channel_manager().GetChannel(channel_2);
EXPECT_EQ(0, base_->DeleteChannel(channel_2));
EXPECT_EQ(1, reference.use_count());
}
TEST_F(VoEBaseTest, GetVersion) {
char v1[1024] = {75};
base_->GetVersion(v1);
std::string v2 = VoiceEngine::GetVersionString() + "\n";
EXPECT_EQ(v2, v1);
}
} // namespace webrtc

5
voice_engine/voice_engine_defines.h
View File

@ -64,11 +64,6 @@ enum { kVoiceEngineMaxRtpExtensionId = 14 };
} // namespace webrtc
#define NOT_SUPPORTED(stat) \
LOG_F(LS_ERROR) << "not supported"; \
stat.SetLastError(VE_FUNC_NOT_SUPPORTED); \
return -1;
namespace webrtc {
inline int VoEId(int veId, int chId) {

2
voice_engine/voice_engine_fixture.cc
View File

@ -19,13 +19,11 @@ VoiceEngineFixture::VoiceEngineFixture()
network_(VoENetwork::GetInterface(voe_)) {
EXPECT_NE(nullptr, base_);
EXPECT_NE(nullptr, network_);
EXPECT_EQ(0, base_->RegisterVoiceEngineObserver(observer_));
apm_ = new rtc::RefCountedObject<test::MockAudioProcessing>();
}
VoiceEngineFixture::~VoiceEngineFixture() {
EXPECT_EQ(2, network_->Release());
EXPECT_EQ(0, base_->DeRegisterVoiceEngineObserver());
EXPECT_EQ(0, base_->Terminate());
EXPECT_EQ(1, base_->Release());
EXPECT_TRUE(VoiceEngine::Delete(voe_));

2
voice_engine/voice_engine_fixture.h
View File

@ -14,7 +14,6 @@
#include "test/mock_transport.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/include/voe_network.h"
#include "voice_engine/mock/mock_voe_observer.h"
namespace webrtc {
@ -26,7 +25,6 @@ class VoiceEngineFixture : public ::testing::Test {
VoiceEngine* voe_;
VoEBase* base_;
VoENetwork* network_;
MockVoEObserver observer_;
FakeAudioDeviceModule adm_;
MockTransport transport_;
rtc::scoped_refptr<AudioProcessing> apm_;

43
voice_engine/voice_engine_impl.cc
View File

@ -74,40 +74,6 @@ VoiceEngine* VoiceEngine::Create() {
return GetVoiceEngine();
}
int VoiceEngine::SetTraceFilter(unsigned int filter) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
VoEId(gVoiceEngineInstanceCounter, -1),
"SetTraceFilter(filter=0x%x)", filter);
// Remember old filter
uint32_t oldFilter = Trace::level_filter();
Trace::set_level_filter(filter);
// If previous log was ignored, log again after changing filter
if (kTraceNone == oldFilter) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1, "SetTraceFilter(filter=0x%x)",
filter);
}
return 0;
}
int VoiceEngine::SetTraceFile(const char* fileNameUTF8, bool addFileCounter) {
int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
VoEId(gVoiceEngineInstanceCounter, -1),
"SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)", fileNameUTF8,
addFileCounter);
return (ret);
}
int VoiceEngine::SetTraceCallback(TraceCallback* callback) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
VoEId(gVoiceEngineInstanceCounter, -1),
"SetTraceCallback(callback=0x%x)", callback);
return (Trace::SetTraceCallback(callback));
}
bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
if (voiceEngine == NULL)
return false;
@ -127,13 +93,4 @@ bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
return true;
}
std::string VoiceEngine::GetVersionString() {
std::string version = "VoiceEngine 4.1.0";
#ifdef WEBRTC_EXTERNAL_TRANSPORT
version += " (External transport build)";
#endif
return version;
}
} // namespace webrtc