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Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.

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This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1459083007

Cr-Commit-Position: refs/heads/master@{#10788}
This commit is contained in:
solenberg
2015-11-25 08:16:52 -08:00
committed by Commit bot
parent 54eb5e2e9a
commit 13725089ef
17 changed files with 277 additions and 64 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
webrtc/audio/audio_receive_stream.cc
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@ -18,12 +18,14 @@
#include "webrtc/base/logging.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioReceiveStream::Config::Rtp::ToString() const {
@ -74,24 +76,26 @@ AudioReceiveStream::AudioReceiveStream(
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(rtp_header_parser_);
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
const int channel_id = config.voe_channel_id;
ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
int error = rtp->SetLocalSSRC(channel_id, config.rtp.local_ssrc);
RTC_DCHECK_EQ(0, error);
for (const auto& extension : config.rtp.extensions) {
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(extension.id, 1);
RTC_DCHECK_LE(extension.id, 14);
if (extension.name == RtpExtension::kAudioLevel) {
error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true,
extension.id);
int error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true,
extension.id);
RTC_DCHECK_EQ(0, error);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, extension.id);
RTC_DCHECK(registered);
} else if (extension.name == RtpExtension::kAbsSendTime) {
error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true,
extension.id);
int error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true,
extension.id);
RTC_DCHECK_EQ(0, error);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, extension.id);

7
webrtc/audio/audio_receive_stream.h
View File

@ -17,11 +17,13 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
class RemoteBitrateEstimator;
namespace internal {
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioReceiveStream final : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
@ -51,6 +53,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};

16
webrtc/audio/audio_receive_stream_unittest.cc
View File

@ -14,6 +14,7 @@
#include "webrtc/audio/conversion.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
namespace webrtc {
@ -53,6 +54,8 @@ const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
ConfigHelper() {
using testing::Invoke;
EXPECT_CALL(voice_engine_,
RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
EXPECT_CALL(voice_engine_,
@ -61,8 +64,13 @@ struct ConfigHelper {
config.voice_engine = &voice_engine_;
audio_state_ = AudioState::Create(config);
EXPECT_CALL(voice_engine_, SetLocalSSRC(kChannelId, kLocalSsrc))
.WillOnce(Return(0));
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
.WillOnce(Invoke([this](int channel_id) {
EXPECT_FALSE(channel_proxy_);
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
return channel_proxy_;
}));
EXPECT_CALL(voice_engine_,
SetReceiveAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
.WillOnce(Return(0));
@ -76,7 +84,7 @@ struct ConfigHelper {
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
}
}
MockRemoteBitrateEstimator* remote_bitrate_estimator() {
return &remote_bitrate_estimator_;
@ -89,6 +97,7 @@ struct ConfigHelper {
using testing::DoAll;
using testing::SetArgPointee;
using testing::SetArgReferee;
EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
@ -110,6 +119,7 @@ struct ConfigHelper {
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioReceiveStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
};
void BuildAbsoluteSendTimeExtension(uint8_t* buffer,

23
webrtc/audio/audio_send_stream.cc
View File

@ -17,10 +17,12 @@
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioSendStream::Config::Rtp::ToString() const {
@ -51,7 +53,6 @@ std::string AudioSendStream::Config::ToString() const {
}
namespace internal {
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
@ -60,25 +61,25 @@ AudioSendStream::AudioSendStream(
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
const int channel_id = config.voe_channel_id;
ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
int error = rtp->SetRTCPStatus(channel_id, true);
RTC_DCHECK_EQ(0, error);
error = rtp->SetLocalSSRC(channel_id, config.rtp.ssrc);
RTC_DCHECK_EQ(0, error);
error = rtp->SetRTCP_CNAME(channel_id, config.rtp.c_name.c_str());
RTC_DCHECK_EQ(0, error);
for (const auto& extension : config.rtp.extensions) {
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(extension.id, 1);
RTC_DCHECK_LE(extension.id, 14);
if (extension.name == RtpExtension::kAbsSendTime) {
error = rtp->SetSendAbsoluteSenderTimeStatus(channel_id, true,
extension.id);
int error = rtp->SetSendAbsoluteSenderTimeStatus(channel_id, true,
extension.id);
RTC_DCHECK_EQ(0, error);
} else if (extension.name == RtpExtension::kAudioLevel) {
error = rtp->SetSendAudioLevelIndicationStatus(channel_id, true,
extension.id);
int error = rtp->SetSendAudioLevelIndicationStatus(channel_id, true,
extension.id);
RTC_DCHECK_EQ(0, error);
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";

8
webrtc/audio/audio_send_stream.h
View File

@ -14,13 +14,16 @@
#include "webrtc/audio_send_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
class VoiceEngine;
namespace internal {
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioSendStream final : public webrtc::AudioSendStream {
public:
AudioSendStream(const webrtc::AudioSendStream::Config& config,
@ -44,6 +47,7 @@ class AudioSendStream final : public webrtc::AudioSendStream {
rtc::ThreadChecker thread_checker_;
const webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};

18
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -13,6 +13,7 @@
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
namespace webrtc {
@ -39,6 +40,7 @@ const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
struct ConfigHelper {
ConfigHelper() : stream_config_(nullptr) {
using testing::Invoke;
using testing::StrEq;
EXPECT_CALL(voice_engine_,
@ -49,12 +51,15 @@ struct ConfigHelper {
config.voice_engine = &voice_engine_;
audio_state_ = AudioState::Create(config);
EXPECT_CALL(voice_engine_, SetRTCPStatus(kChannelId, true))
.WillOnce(Return(0));
EXPECT_CALL(voice_engine_, SetLocalSSRC(kChannelId, kSsrc))
.WillOnce(Return(0));
EXPECT_CALL(voice_engine_, SetRTCP_CNAME(kChannelId, StrEq(kCName)))
.WillOnce(Return(0));
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
.WillOnce(Invoke([this](int channel_id) {
EXPECT_FALSE(channel_proxy_);
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
return channel_proxy_;
}));
EXPECT_CALL(voice_engine_,
SetSendAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
.WillOnce(Return(0));
@ -109,6 +114,7 @@ struct ConfigHelper {
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
};
} // namespace

30
webrtc/test/mock_voe_channel_proxy.h Normal file
View File

@ -0,0 +1,30 @@
/*
* Copyright (c) 2015 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_TEST_MOCK_VOE_CHANNEL_PROXY_H_
#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
#include <string>
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/voice_engine/channel_proxy.h"
namespace webrtc {
namespace test {
class MockVoEChannelProxy : public voe::ChannelProxy {
public:
MOCK_METHOD1(SetRTCPStatus, void(bool enable));
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_

15
webrtc/test/mock_voice_engine.h
View File

@ -12,6 +12,7 @@
#define WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
@ -25,12 +26,26 @@ class MockVoiceEngine : public VoiceEngineImpl {
// Increase ref count so this object isn't automatically deleted whenever
// interfaces are Release():d.
++_ref_count;
// We add this default behavior to make the mock easier to use in tests. It
// will create a NiceMock of a voe::ChannelProxy.
ON_CALL(*this, ChannelProxyFactory(testing::_))
.WillByDefault(
testing::Invoke([](int channel_id) {
return new testing::NiceMock<MockVoEChannelProxy>();
}));
}
~MockVoiceEngine() override {
// Decrease ref count before base class d-tor is called; otherwise it will
// trigger an assertion.
--_ref_count;
}
// Allows injecting a ChannelProxy factory.
MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id));
// VoiceEngineImpl
rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id) override {
return rtc::scoped_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id));
}
// VoEAudioProcessing
MOCK_METHOD2(SetNsStatus, int(bool enable, NsModes mode));

1
webrtc/test/webrtc_test_common.gyp
View File

@ -37,6 +37,7 @@
'layer_filtering_transport.cc',
'layer_filtering_transport.h',
'mock_transport.h',
'mock_voe_channel_proxy.h',
'mock_voice_engine.h',
'null_transport.cc',
'null_transport.h',

2
webrtc/voice_engine/BUILD.gn
View File

@ -14,6 +14,8 @@ source_set("voice_engine") {
"channel.h",
"channel_manager.cc",
"channel_manager.h",
"channel_proxy.cc",
"channel_proxy.h",
"dtmf_inband.cc",
"dtmf_inband.h",
"dtmf_inband_queue.cc",

44
webrtc/voice_engine/channel_proxy.cc Normal file
View File

@ -0,0 +1,44 @@
/*
* Copyright (c) 2015 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/channel_proxy.h"
#include "webrtc/base/checks.h"
#include "webrtc/voice_engine/channel.h"
namespace webrtc {
namespace voe {
ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
channel_owner_(channel_owner) {
RTC_CHECK(channel_owner_.channel());
}
void ChannelProxy::SetRTCPStatus(bool enable) {
RTC_DCHECK(channel_owner_.channel());
channel_owner_.channel()->SetRTCPStatus(enable);
}
void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK(channel_owner_.channel());
int error = channel_owner_.channel()->SetLocalSSRC(ssrc);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
// Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
std::string c_name_limited = c_name.substr(0, 255);
RTC_DCHECK(channel_owner_.channel());
int error = channel_owner_.channel()->SetRTCP_CNAME(c_name_limited.c_str());
RTC_DCHECK_EQ(0, error);
}
} // namespace voe
} // namespace webrtc

44
webrtc/voice_engine/channel_proxy.h Normal file
View File

@ -0,0 +1,44 @@
/*
* Copyright (c) 2015 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_CHANNEL_PROXY_H_
#define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
#include "webrtc/voice_engine/channel_manager.h"
#include <string>
namespace webrtc {
namespace voe {
// This class provides the "view" of a voe::Channel that we need to implement
// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
// purposes:
// 1. Allow mocking just the interfaces used, instead of the entire
// voe::Channel class.
// 2. Provide a refined interface for the stream classes, including assumptions
// on return values and input adaptation.
class ChannelProxy {
public:
ChannelProxy();
explicit ChannelProxy(const ChannelOwner& channel_owner);
virtual ~ChannelProxy() {}
virtual void SetRTCPStatus(bool enable);
virtual void SetLocalSSRC(uint32_t ssrc);
virtual void SetRTCP_CNAME(const std::string& c_name);
private:
ChannelOwner channel_owner_;
};
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_

31
webrtc/voice_engine/voe_base_impl.cc
View File

@ -47,7 +47,7 @@ VoEBaseImpl::~VoEBaseImpl() {
delete &callbackCritSect_;
}
void VoEBaseImpl::OnErrorIsReported(ErrorCode error) {
void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
CriticalSectionScoped cs(&callbackCritSect_);
int errCode = 0;
if (error == AudioDeviceObserver::kRecordingError) {
@ -63,7 +63,7 @@ void VoEBaseImpl::OnErrorIsReported(ErrorCode error) {
}
}
void VoEBaseImpl::OnWarningIsReported(WarningCode warning) {
void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) {
CriticalSectionScoped cs(&callbackCritSect_);
int warningCode = 0;
if (warning == AudioDeviceObserver::kRecordingWarning) {
@ -79,21 +79,28 @@ void VoEBaseImpl::OnWarningIsReported(WarningCode warning) {
}
}
int32_t VoEBaseImpl::RecordedDataIsAvailable(
const void* audioSamples, size_t nSamples, size_t nBytesPerSample,
uint8_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS,
int32_t clockDrift, uint32_t micLevel, bool keyPressed,
uint32_t& newMicLevel) {
int32_t VoEBaseImpl::RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) {
newMicLevel = static_cast<uint32_t>(ProcessRecordedDataWithAPM(
nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples,
totalDelayMS, clockDrift, micLevel, keyPressed));
totalDelayMS, clockDrift, currentMicLevel, keyPressed));
return 0;
}
int32_t VoEBaseImpl::NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
uint8_t nChannels, uint32_t samplesPerSec,
void* audioSamples, size_t& nSamplesOut,
int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
GetPlayoutData(static_cast<int>(samplesPerSec), static_cast<int>(nChannels),

64
webrtc/voice_engine/voe_base_impl.h
View File

@ -54,37 +54,57 @@ class VoEBaseImpl : public VoEBase,
int AssociateSendChannel(int channel, int accociate_send_channel) override;
// AudioTransport
int32_t RecordedDataIsAvailable(const void* audioSamples, size_t nSamples,
size_t nBytesPerSample, uint8_t nChannels,
uint32_t samplesPerSec, uint32_t totalDelayMS,
int32_t clockDrift, uint32_t micLevel,
bool keyPressed,
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) override;
int32_t NeedMorePlayData(size_t nSamples, size_t nBytesPerSample,
uint8_t nChannels, uint32_t samplesPerSec,
void* audioSamples, size_t& nSamplesOut,
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const uint8_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
int OnDataAvailable(const int voe_channels[], int number_of_voe_channels,
const int16_t* audio_data, int sample_rate,
int number_of_channels, size_t number_of_frames,
int audio_delay_milliseconds, int volume,
bool key_pressed, bool need_audio_processing) override;
void OnData(int voe_channel, const void* audio_data, int bits_per_sample,
int sample_rate, int number_of_channels,
int OnDataAvailable(const int voe_channels[],
int number_of_voe_channels,
const int16_t* audio_data,
int sample_rate,
int number_of_channels,
size_t number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool key_pressed,
bool need_audio_processing) override;
void OnData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) override;
void PushCaptureData(int voe_channel, const void* audio_data,
int bits_per_sample, int sample_rate,
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) override;
void PullRenderData(int bits_per_sample, int sample_rate,
int number_of_channels, size_t number_of_frames,
void* audio_data, int64_t* elapsed_time_ms,
void PullRenderData(int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
// AudioDeviceObserver
void OnErrorIsReported(ErrorCode error) override;
void OnWarningIsReported(WarningCode warning) override;
void OnErrorIsReported(const ErrorCode error) override;
void OnWarningIsReported(const WarningCode warning) override;
protected:
VoEBaseImpl(voe::SharedData* shared);

2
webrtc/voice_engine/voice_engine.gyp
View File

@ -49,6 +49,8 @@
'channel.h',
'channel_manager.cc',
'channel_manager.h',
'channel_proxy.cc',
'channel_proxy.h',
'dtmf_inband.cc',
'dtmf_inband.h',
'dtmf_inband_queue.cc',

12
webrtc/voice_engine/voice_engine_impl.cc
View File

@ -15,8 +15,11 @@
#include "webrtc/modules/utility/include/jvm_android.h"
#endif
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
@ -77,6 +80,15 @@ int VoiceEngineImpl::Release() {
return new_ref;
}
rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
int channel_id) {
RTC_DCHECK(channel_id >= 0);
CriticalSectionScoped cs(crit_sec());
RTC_DCHECK(statistics().Initialized());
return rtc::scoped_ptr<voe::ChannelProxy>(
new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
}
VoiceEngine* VoiceEngine::Create() {
Config* config = new Config();
return GetVoiceEngine(config, true);

8
webrtc/voice_engine/voice_engine_impl.h
View File

@ -11,6 +11,7 @@
#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/voice_engine/voe_base_impl.h"
@ -48,6 +49,9 @@
#endif
namespace webrtc {
namespace voe {
class ChannelProxy;
} // namespace voe
class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
public VoiceEngine,
@ -128,6 +132,10 @@ class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
// This implements the Release() method for all the inherited interfaces.
int Release() override;
// Backdoor to access a voe::Channel object without a channel ID. This is only
// to be used while refactoring the VoE API!
virtual rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id);
// This is *protected* so that FakeVoiceEngine can inherit from the class and
// manipulate the reference count. See: fake_voice_engine.h.
protected: