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Remove legacy VoiceEngine.

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Now that voe::Channel is owned by Audio[Send|Receive]Stream, the legacy
VoiceEngine and the VoEBase interface is unused.

Also removes Atomic32, which was only used for ref counting VoiceEngine.

Bug: webrtc:4690
Change-Id: I73b8a083df544a8ab6383d57075a65ce955c592a
Reviewed-on: https://webrtc-review.googlesource.com/38723
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21595}
This commit is contained in:
Fredrik Solenberg
2018-01-12 10:31:52 +01:00
committed by Commit Bot
parent f0839f2c96
commit 2d49e4e18d
12 changed files with 0 additions and 572 deletions
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1
PRESUBMIT.py
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@ -94,7 +94,6 @@ LEGACY_API_DIRS = (
'modules/video_coding/include', 'modules/video_coding/include',
'rtc_base', 'rtc_base',
'system_wrappers/include', 'system_wrappers/include',
'voice_engine/include',
) )
# NOTE: The set of directories in API_DIRS should be the same as those # NOTE: The set of directories in API_DIRS should be the same as those

1
native-api.md
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@ -34,7 +34,6 @@ Legacy API directory | Including subdirectories?
`pc` | No `pc` | No
`rtc_base` | No `rtc_base` | No
`system_wrappers/include` | No `system_wrappers/include` | No
`voice_engine/include` | No
While the files, types, functions, macros, build targets, etc. in the While the files, types, functions, macros, build targets, etc. in the
API and legacy API directories will sometimes undergo incompatible API and legacy API directories will sometimes undergo incompatible

2
system_wrappers/BUILD.gn
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@ -17,7 +17,6 @@ rtc_static_library("system_wrappers") {
sources = [ sources = [
"include/aligned_array.h", "include/aligned_array.h",
"include/aligned_malloc.h", "include/aligned_malloc.h",
"include/atomic32.h",
"include/clock.h", "include/clock.h",
"include/cpu_info.h", "include/cpu_info.h",
"include/event_wrapper.h", "include/event_wrapper.h",
@ -28,7 +27,6 @@ rtc_static_library("system_wrappers") {
"include/sleep.h", "include/sleep.h",
"include/timestamp_extrapolator.h", "include/timestamp_extrapolator.h",
"source/aligned_malloc.cc", "source/aligned_malloc.cc",
"source/atomic32.cc",
"source/clock.cc", "source/clock.cc",
"source/cpu_features.cc", "source/cpu_features.cc",
"source/cpu_info.cc", "source/cpu_info.cc",

59
system_wrappers/include/atomic32.h
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@ -1,59 +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.
*/
// Atomic, system independent 32-bit integer. Unless you know what you're
// doing, use locks instead! :-)
//
// Note: assumes 32-bit (or higher) system
#ifndef SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_
#define SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_
#include <atomic>
#include <stddef.h>
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
namespace webrtc {
// DEPRECATED: Please use std::atomic<int32_t> instead.
// TODO(yuweih): Replace Atomic32 uses with std::atomic<int32_t> and remove this
// class. (bugs.webrtc.org/8428)
// 32 bit atomic variable. Note that this class relies on the compiler to
// align the 32 bit value correctly (on a 32 bit boundary), so as long as you're
// not doing things like reinterpret_cast over some custom allocated memory
// without being careful with alignment, you should be fine.
class Atomic32 {
public:
Atomic32(int32_t initial_value = 0);
~Atomic32();
// Prefix operator!
int32_t operator++();
int32_t operator--();
int32_t operator+=(int32_t value);
int32_t operator-=(int32_t value);
// Sets the value atomically to new_value if the value equals compare value.
// The function returns true if the exchange happened.
bool CompareExchange(int32_t new_value, int32_t compare_value);
int32_t Value() const;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Atomic32);
std::atomic<int32_t> value_;
};
} // namespace webrtc
#endif // SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_

47
system_wrappers/source/atomic32.cc
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@ -1,47 +0,0 @@
/*
* Copyright (c) 2017 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 "system_wrappers/include/atomic32.h"
#include <assert.h>
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
Atomic32::Atomic32(int32_t initial_value) : value_(initial_value) {}
Atomic32::~Atomic32() {}
int32_t Atomic32::operator++() {
return ++value_;
}
int32_t Atomic32::operator--() {
return --value_;
}
int32_t Atomic32::operator+=(int32_t value) {
return value_ += value;
}
int32_t Atomic32::operator-=(int32_t value) {
return value_ -= value;
}
bool Atomic32::CompareExchange(int32_t new_value, int32_t compare_value) {
return value_.compare_exchange_strong(compare_value, new_value);
}
int32_t Atomic32::Value() const {
return value_.load();
}
} // namespace webrtc

6
voice_engine/BUILD.gn
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@ -17,16 +17,10 @@ rtc_static_library("voice_engine") {
"channel.h", "channel.h",
"channel_proxy.cc", "channel_proxy.cc",
"channel_proxy.h", "channel_proxy.h",
"include/voe_base.h",
"include/voe_errors.h",
"transport_feedback_packet_loss_tracker.cc", "transport_feedback_packet_loss_tracker.cc",
"transport_feedback_packet_loss_tracker.h", "transport_feedback_packet_loss_tracker.h",
"utility.cc", "utility.cc",
"utility.h", "utility.h",
"voe_base_impl.cc",
"voe_base_impl.h",
"voice_engine_impl.cc",
"voice_engine_impl.h",
] ]
if (is_win) { if (is_win) {

118
voice_engine/include/voe_base.h
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@ -1,118 +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:
//
// - Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law).
// - Initialization and termination.
// - Trace information on text files or via callbacks.
// - Multi-channel support (mixing, sending to multiple destinations etc.).
//
// To support other codecs than G.711, the VoECodec sub-API must be utilized.
//
// Usage example, omitting error checking:
//
// using namespace webrtc;
// VoiceEngine* voe = VoiceEngine::Create();
// VoEBase* base = VoEBase::GetInterface(voe);
// base->Init();
// int ch = base->CreateChannel();
// base->StartPlayout(ch);
// ...
// base->DeleteChannel(ch);
// base->Terminate();
// base->Release();
// VoiceEngine::Delete(voe);
//
#ifndef VOICE_ENGINE_VOE_BASE_H_
#define VOICE_ENGINE_VOE_BASE_H_
#include "api/audio_codecs/audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModule;
class AudioProcessing;
// VoiceEngine
class WEBRTC_DLLEXPORT VoiceEngine {
public:
// Creates a VoiceEngine object, which can then be used to acquire
// sub-APIs. Returns NULL on failure.
static VoiceEngine* Create();
// Deletes a created VoiceEngine object and releases the utilized resources.
// Note that if there are outstanding references held via other interfaces,
// the voice engine instance will not actually be deleted until those
// references have been released.
static bool Delete(VoiceEngine*& voiceEngine);
protected:
VoiceEngine() {}
~VoiceEngine() {}
};
// VoEBase
class WEBRTC_DLLEXPORT VoEBase {
public:
struct ChannelConfig {
AudioCodingModule::Config acm_config;
bool enable_voice_pacing = false;
};
// Factory for the VoEBase sub-API. Increases an internal reference
// counter if successful. Returns NULL if the API is not supported or if
// construction fails.
static VoEBase* GetInterface(VoiceEngine* voiceEngine);
// Releases the VoEBase sub-API and decreases an internal reference
// counter. Returns the new reference count. This value should be zero
// for all sub-APIs before the VoiceEngine object can be safely deleted.
virtual int Release() = 0;
// Initializes all common parts of the VoiceEngine; e.g. all
// encoders/decoders, the sound card and core receiving components.
// This method also makes it possible to install some user-defined external
// modules:
// - The Audio Device Module (ADM) which implements all the audio layer
// functionality in a separate (reference counted) module.
// - The AudioProcessing module is unused - only kept for API compatibility.
// - An AudioDecoderFactory - used to create audio decoders.
virtual int Init(
AudioDeviceModule* audio_device,
AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) = 0;
// Terminates all VoiceEngine functions and releases allocated resources.
virtual void Terminate() = 0;
// Creates a new channel and allocates the required resources for it.
// The second version accepts a |config| struct which includes an Audio Coding
// Module config and an option to enable voice pacing. Note that the
// decoder_factory member of the ACM config will be ignored (the decoder
// factory set through Init() will always be used).
// Returns channel ID or -1 in case of an error.
virtual int CreateChannel(const ChannelConfig& config) = 0;
// Deletes an existing channel and releases the utilized resources.
// Returns -1 in case of an error, 0 otherwise.
virtual int DeleteChannel(int channel) = 0;
protected:
VoEBase() {}
virtual ~VoEBase() {}
};
} // namespace webrtc
#endif // VOICE_ENGINE_VOE_BASE_H_

165
voice_engine/include/voe_errors.h
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@ -1,165 +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_VOE_ERRORS_H_
#define VOICE_ENGINE_VOE_ERRORS_H_
// Warnings
#define VE_PORT_NOT_DEFINED 8001
#define VE_CHANNEL_NOT_VALID 8002
#define VE_FUNC_NOT_SUPPORTED 8003
#define VE_INVALID_LISTNR 8004
#define VE_INVALID_ARGUMENT 8005
#define VE_INVALID_PORT_NMBR 8006
#define VE_INVALID_PLNAME 8007
#define VE_INVALID_PLFREQ 8008
#define VE_INVALID_PLTYPE 8009
#define VE_INVALID_PACSIZE 8010
#define VE_NOT_SUPPORTED 8011
#define VE_ALREADY_LISTENING 8012
#define VE_CHANNEL_NOT_CREATED 8013
#define VE_MAX_ACTIVE_CHANNELS_REACHED 8014
#define VE_REC_CANNOT_PREPARE_HEADER 8015
#define VE_REC_CANNOT_ADD_BUFFER 8016
#define VE_PLAY_CANNOT_PREPARE_HEADER 8017
#define VE_ALREADY_SENDING 8018
#define VE_INVALID_IP_ADDRESS 8019
#define VE_ALREADY_PLAYING 8020
#define VE_NOT_ALL_VERSION_INFO 8021
// 8022 is not used
#define VE_INVALID_CHANNELS 8023
#define VE_SET_PLTYPE_FAILED 8024
// 8025 is not used
#define VE_NOT_INITED 8026
#define VE_NOT_SENDING 8027
#define VE_EXT_TRANSPORT_NOT_SUPPORTED 8028
#define VE_EXTERNAL_TRANSPORT_ENABLED 8029
#define VE_STOP_RECORDING_FAILED 8030
#define VE_INVALID_RATE 8031
#define VE_INVALID_PACKET 8032
#define VE_NO_GQOS 8033
#define VE_INVALID_TIMESTAMP 8034
#define VE_RECEIVE_PACKET_TIMEOUT 8035
// 8036 is not used
#define VE_INIT_FAILED_WRONG_EXPIRY 8037
#define VE_SENDING 8038
#define VE_ENABLE_IPV6_FAILED 8039
#define VE_FUNC_NO_STEREO 8040
// Range 8041-8080 is not used
#define VE_FW_TRAVERSAL_ALREADY_INITIALIZED 8081
#define VE_PACKET_RECEIPT_RESTARTED 8082
#define VE_NOT_ALL_INFO 8083
#define VE_CANNOT_SET_SEND_CODEC 8084
#define VE_CODEC_ERROR 8085
#define VE_NETEQ_ERROR 8086
#define VE_RTCP_ERROR 8087
#define VE_INVALID_OPERATION 8088
#define VE_CPU_INFO_ERROR 8089
#define VE_SOUNDCARD_ERROR 8090
#define VE_SPEECH_LEVEL_ERROR 8091
#define VE_SEND_ERROR 8092
#define VE_CANNOT_REMOVE_CONF_CHANNEL 8093
#define VE_PLTYPE_ERROR 8094
#define VE_SET_RED_FAILED 8095
#define VE_CANNOT_GET_PLAY_DATA 8096
#define VE_APM_ERROR 8097
#define VE_RUNTIME_PLAY_WARNING 8098
#define VE_RUNTIME_REC_WARNING 8099
#define VE_NOT_PLAYING 8100
#define VE_SOCKETS_NOT_INITED 8101
#define VE_CANNOT_GET_SOCKET_INFO 8102
#define VE_INVALID_MULTICAST_ADDRESS 8103
#define VE_DESTINATION_NOT_INITED 8104
#define VE_RECEIVE_SOCKETS_CONFLICT 8105
#define VE_SEND_SOCKETS_CONFLICT 8106
// 8107 is not used
#define VE_NOISE_WARNING 8109
#define VE_CANNOT_GET_SEND_CODEC 8110
#define VE_CANNOT_GET_REC_CODEC 8111
#define VE_ALREADY_INITED 8112
#define VE_CANNOT_SET_SECONDARY_SEND_CODEC 8113
#define VE_CANNOT_GET_SECONDARY_SEND_CODEC 8114
#define VE_CANNOT_REMOVE_SECONDARY_SEND_CODEC 8115
// 8116 is not used
// Errors causing limited functionality
#define VE_RTCP_SOCKET_ERROR 9001
#define VE_MIC_VOL_ERROR 9002
#define VE_SPEAKER_VOL_ERROR 9003
#define VE_CANNOT_ACCESS_MIC_VOL 9004
#define VE_CANNOT_ACCESS_SPEAKER_VOL 9005
#define VE_GET_MIC_VOL_ERROR 9006
#define VE_GET_SPEAKER_VOL_ERROR 9007
#define VE_THREAD_RTCP_ERROR 9008
#define VE_CANNOT_INIT_APM 9009
#define VE_SEND_SOCKET_TOS_ERROR 9010
#define VE_CANNOT_RETRIEVE_DEVICE_NAME 9013
#define VE_SRTP_ERROR 9014
// 9015 is not used
#define VE_INTERFACE_NOT_FOUND 9016
#define VE_TOS_GQOS_CONFLICT 9017
#define VE_CANNOT_ADD_CONF_CHANNEL 9018
#define VE_BUFFER_TOO_SMALL 9019
#define VE_CANNOT_EXECUTE_SETTING 9020
#define VE_CANNOT_RETRIEVE_SETTING 9021
// 9022 is not used
#define VE_RTP_KEEPALIVE_FAILED 9023
#define VE_SEND_DTMF_FAILED 9024
#define VE_CANNOT_RETRIEVE_CNAME 9025
// 9026 is not used
// 9027 is not used
#define VE_CANNOT_RETRIEVE_RTP_STAT 9028
#define VE_GQOS_ERROR 9029
#define VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED 9030
#define VE_TOS_INVALID 9031
#define VE_TOS_ERROR 9032
#define VE_CANNOT_RETRIEVE_VALUE 9033
// Critical errors that stops voice functionality
#define VE_PLAY_UNDEFINED_SC_ERR 10001
#define VE_REC_CANNOT_OPEN_SC 10002
#define VE_SOCKET_ERROR 10003
#define VE_MMSYSERR_INVALHANDLE 10004
#define VE_MMSYSERR_NODRIVER 10005
#define VE_MMSYSERR_NOMEM 10006
#define VE_WAVERR_UNPREPARED 10007
#define VE_WAVERR_STILLPLAYING 10008
#define VE_UNDEFINED_SC_ERR 10009
#define VE_UNDEFINED_SC_REC_ERR 10010
#define VE_THREAD_ERROR 10011
#define VE_CANNOT_START_RECORDING 10012
#define VE_PLAY_CANNOT_OPEN_SC 10013
#define VE_NO_WINSOCK_2 10014
#define VE_SEND_SOCKET_ERROR 10015
#define VE_BAD_FILE 10016
#define VE_EXPIRED_COPY 10017
#define VE_NOT_AUTHORISED 10018
#define VE_RUNTIME_PLAY_ERROR 10019
#define VE_RUNTIME_REC_ERROR 10020
#define VE_BAD_ARGUMENT 10021
#define VE_LINUX_API_ONLY 10022
#define VE_REC_DEVICE_REMOVED 10023
#define VE_NO_MEMORY 10024
#define VE_BAD_HANDLE 10025
#define VE_RTP_RTCP_MODULE_ERROR 10026
#define VE_AUDIO_CODING_MODULE_ERROR 10027
#define VE_AUDIO_DEVICE_MODULE_ERROR 10028
#define VE_CANNOT_START_PLAYOUT 10029
#define VE_CANNOT_STOP_RECORDING 10030
#define VE_CANNOT_STOP_PLAYOUT 10031
#define VE_CANNOT_INIT_CHANNEL 10032
#define VE_RECV_SOCKET_ERROR 10033
#define VE_SOCKET_TRANSPORT_MODULE_ERROR 10034
#define VE_AUDIO_CONF_MIX_MODULE_ERROR 10035
// Warnings for other platforms (reserved range 8061-8080)
#define VE_IGNORED_FUNCTION 8061
#endif // VOICE_ENGINE_VOE_ERRORS_H_

29
voice_engine/voe_base_impl.cc
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@ -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.
*/
#include "voice_engine/voe_base_impl.h"
#include "voice_engine/voice_engine_impl.h"
namespace webrtc {
VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) {
if (nullptr == voiceEngine) {
return nullptr;
}
VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
s->AddRef();
return s;
}
VoEBaseImpl::VoEBaseImpl() {}
VoEBaseImpl::~VoEBaseImpl() {}
} // namespace webrtc

38
voice_engine/voe_base_impl.h
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@ -1,38 +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_VOE_BASE_IMPL_H_
#define VOICE_ENGINE_VOE_BASE_IMPL_H_
#include "voice_engine/include/voe_base.h"
namespace webrtc {
class VoEBaseImpl : public VoEBase {
public:
int Init(
AudioDeviceModule* audio_device,
AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override {
return 0;
}
void Terminate() override {}
int CreateChannel(const ChannelConfig& config) override { return 1; }
int DeleteChannel(int channel) override { return 0; }
protected:
VoEBaseImpl();
~VoEBaseImpl() override;
};
} // namespace webrtc
#endif // VOICE_ENGINE_VOE_BASE_IMPL_H_

63
voice_engine/voice_engine_impl.cc
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@ -1,63 +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 "voice_engine/voice_engine_impl.h"
namespace webrtc {
// Counter to be ensure that we can add a correct ID in all static trace
// methods. It is not the nicest solution, especially not since we already
// have a counter in VoEBaseImpl. In other words, there is room for
// improvement here.
static int32_t gVoiceEngineInstanceCounter = 0;
VoiceEngine* GetVoiceEngine() {
VoiceEngineImpl* self = new VoiceEngineImpl();
if (self != NULL) {
self->AddRef(); // First reference. Released in VoiceEngine::Delete.
gVoiceEngineInstanceCounter++;
}
return self;
}
int VoiceEngineImpl::AddRef() {
return ++_ref_count;
}
// This implements the Release() method for all the inherited interfaces.
int VoiceEngineImpl::Release() {
int new_ref = --_ref_count;
assert(new_ref >= 0);
if (new_ref == 0) {
// Clear any pointers before starting destruction. Otherwise worker-
// threads will still have pointers to a partially destructed object.
// Example: AudioDeviceBuffer::RequestPlayoutData() can access a
// partially deconstructed |_ptrCbAudioTransport| during destruction
// if we don't call Terminate here.
Terminate();
delete this;
}
return new_ref;
}
VoiceEngine* VoiceEngine::Create() {
return GetVoiceEngine();
}
bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
if (voiceEngine == NULL)
return false;
VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
s->Release();
voiceEngine = NULL;
return true;
}
} // namespace webrtc

43
voice_engine/voice_engine_impl.h
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@ -1,43 +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_VOICE_ENGINE_IMPL_H_
#define VOICE_ENGINE_VOICE_ENGINE_IMPL_H_
#include <memory>
#include "system_wrappers/include/atomic32.h"
#include "typedefs.h" // NOLINT(build/include)
#include "voice_engine/voe_base_impl.h"
namespace webrtc {
class VoiceEngineImpl : public VoiceEngine,
public VoEBaseImpl {
public:
VoiceEngineImpl()
: VoEBaseImpl(),
_ref_count(0) {}
~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); }
int AddRef();
// This implements the Release() method for all the inherited interfaces.
int Release() override;
// This is *protected* so that FakeVoiceEngine can inherit from the class and
// manipulate the reference count. See: fake_voice_engine.h.
protected:
Atomic32 _ref_count;
};
} // namespace webrtc
#endif // VOICE_ENGINE_VOICE_ENGINE_IMPL_H_