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Remove the Windows Wave audio device implementation.

Browse Source 
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.

Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.

BUG=webrtc:7183
R=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
This commit is contained in:
Tommi 2017-02-17 23:48:07 +01:00
parent 8fefe9889d
commit cc8588c040
8 changed files with 20 additions and 4224 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/common_types.h
View File

@ -467,7 +467,6 @@ enum StereoChannel { kStereoLeft = 0, kStereoRight, kStereoBoth };
// Audio device layers
enum AudioLayers {
kAudioPlatformDefault = 0,
kAudioWindowsWave = 1,
kAudioWindowsCore = 2,
kAudioLinuxAlsa = 3,
kAudioLinuxPulse = 4

2
webrtc/modules/audio_device/BUILD.gn
View File

@ -211,8 +211,6 @@ rtc_static_library("audio_device") {
sources += [
"win/audio_device_core_win.cc",
"win/audio_device_core_win.h",
"win/audio_device_wave_win.cc",
"win/audio_device_wave_win.h",
"win/audio_mixer_manager_win.cc",
"win/audio_mixer_manager_win.h",
]

23
webrtc/modules/audio_device/audio_device_impl.cc
View File

@ -23,7 +23,6 @@
#include <string.h>
#if defined(_WIN32)
#include "audio_device_wave_win.h"
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#include "audio_device_core_win.h"
#endif
@ -200,17 +199,6 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
// Create the *Windows* implementation of the Audio Device
//
#if defined(_WIN32)
if ((audioLayer == kWindowsWaveAudio)
#if !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// Wave audio is default if Core audio is not supported in this build
|| (audioLayer == kPlatformDefaultAudio)
#endif
) {
// create *Windows Wave Audio* implementation
ptrAudioDevice = new AudioDeviceWindowsWave(Id());
LOG(INFO) << "Windows Wave APIs will be utilized";
}
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
if ((audioLayer == kWindowsCoreAudio) ||
(audioLayer == kPlatformDefaultAudio)) {
@ -220,20 +208,9 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
// create *Windows Core Audio* implementation
ptrAudioDevice = new AudioDeviceWindowsCore(Id());
LOG(INFO) << "Windows Core Audio APIs will be utilized";
} else {
// create *Windows Wave Audio* implementation
ptrAudioDevice = new AudioDeviceWindowsWave(Id());
if (ptrAudioDevice != NULL) {
// Core Audio was not supported => revert to Windows Wave instead
_platformAudioLayer =
kWindowsWaveAudio; // modify the state set at construction
LOG(WARNING) << "Windows Core Audio is *not* supported => Wave APIs "
"will be utilized instead";
}
}
}
#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#endif // #if defined(_WIN32)
#if defined(WEBRTC_ANDROID)
// Create an Android audio manager.

1
webrtc/modules/audio_device/include/audio_device.h
View File

@ -26,7 +26,6 @@ class AudioDeviceModule : public RefCountedModule {
enum AudioLayer {
kPlatformDefaultAudio = 0,
kWindowsWaveAudio = 1,
kWindowsCoreAudio = 2,
kLinuxAlsaAudio = 3,
kLinuxPulseAudio = 4,

145
webrtc/modules/audio_device/test/audio_device_test_api.cc
View File

@ -163,45 +163,22 @@ class AudioDeviceAPITest: public testing::Test {
// Windows:
// if (WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// user can select between default (Core) or Wave
// else
// user can select between default (Wave) or Wave
// user can select only the default (Core)
const int32_t kId = 444;
#if defined(_WIN32)
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined!\n\n");
// create default implementation (=Core Audio) instance
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
EXPECT_EQ(0, audio_device_.release()->Release());
// create non-default (=Wave Audio) instance
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
EXPECT_EQ(0, audio_device_.release()->Release());
// explicitly specify usage of Core Audio (same as default)
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsCoreAudio)) != NULL);
#else
TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined!\n");
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
// create default implementation (=Wave Audio) instance
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
EXPECT_EQ(0, audio_device_.release()->Release());
// explicitly specify usage of Wave Audio (same as default)
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
#endif
#endif
#if defined(ANDROID)
// Fails tests
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
@ -212,8 +189,6 @@ class AudioDeviceAPITest: public testing::Test {
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
#elif defined(WEBRTC_LINUX)
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
// create default implementation instance
@ -228,8 +203,6 @@ class AudioDeviceAPITest: public testing::Test {
#if defined(WEBRTC_MAC)
// Fails tests
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
@ -471,7 +444,7 @@ TEST_F(AudioDeviceAPITest, SetRecordingDevice) {
TEST_F(AudioDeviceAPITest, PlayoutIsAvailable) {
bool available;
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
EXPECT_TRUE(audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
EXPECT_EQ(0, audio_device_->PlayoutIsAvailable(&available));
@ -494,7 +467,7 @@ TEST_F(AudioDeviceAPITest, PlayoutIsAvailable) {
TEST_F(AudioDeviceAPITest, RecordingIsAvailable) {
bool available;
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
EXPECT_EQ(0, audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice));
EXPECT_EQ(0, audio_device_->RecordingIsAvailable(&available));
@ -623,7 +596,7 @@ TEST_F(AudioDeviceAPITest, StartAndStopPlayout) {
EXPECT_EQ(-1, audio_device_->StartPlayout());
EXPECT_EQ(0, audio_device_->StopPlayout());
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -682,7 +655,7 @@ TEST_F(AudioDeviceAPITest, StartAndStopRecording) {
EXPECT_EQ(-1, audio_device_->StartRecording());
EXPECT_EQ(0, audio_device_->StopRecording());
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -732,58 +705,6 @@ TEST_F(AudioDeviceAPITest, StartAndStopRecording) {
}
}
#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
TEST_F(AudioDeviceAPITest, SetAndGetWaveOutVolume) {
uint32_t vol(0);
// NOTE 1: Windows Wave only!
// NOTE 2: It seems like the waveOutSetVolume API returns
// MMSYSERR_NOTSUPPORTED on some Vista machines!
const uint16_t maxVol(0xFFFF);
uint16_t volL, volR;
CheckInitialPlayoutStates();
// make dummy test to see if this API is supported
int32_t works = audio_device_->SetWaveOutVolume(vol, vol);
WARNING(works == 0);
if (works == 0)
{
// set volume without open playout device
for (vol = 0; vol <= maxVol; vol += (maxVol/5))
{
EXPECT_EQ(0, audio_device_->SetWaveOutVolume(vol, vol));
EXPECT_EQ(0, audio_device_->WaveOutVolume(volL, volR));
EXPECT_TRUE((volL == vol) && (volR == vol));
}
// repeat test but this time with an open (default) output device
EXPECT_EQ(0, audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultDevice));
EXPECT_EQ(0, audio_device_->InitPlayout());
EXPECT_TRUE(audio_device_->PlayoutIsInitialized());
for (vol = 0; vol <= maxVol; vol += (maxVol/5))
{
EXPECT_EQ(0, audio_device_->SetWaveOutVolume(vol, vol));
EXPECT_EQ(0, audio_device_->WaveOutVolume(volL, volR));
EXPECT_TRUE((volL == vol) && (volR == vol));
}
// as above but while playout is active
EXPECT_EQ(0, audio_device_->StartPlayout());
EXPECT_TRUE(audio_device_->Playing());
for (vol = 0; vol <= maxVol; vol += (maxVol/5))
{
EXPECT_EQ(0, audio_device_->SetWaveOutVolume(vol, vol));
EXPECT_EQ(0, audio_device_->WaveOutVolume(volL, volR));
EXPECT_TRUE((volL == vol) && (volR == vol));
}
}
EXPECT_EQ(0, audio_device_->StopPlayout());
EXPECT_FALSE(audio_device_->Playing());
}
#endif // defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
TEST_F(AudioDeviceAPITest, InitSpeaker) {
// NOTE: By calling Terminate (in TearDown) followed by Init (in SetUp) we
@ -857,7 +778,7 @@ TEST_F(AudioDeviceAPITest, SpeakerVolumeIsAvailable) {
CheckInitialPlayoutStates();
bool available;
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// check the kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -904,20 +825,7 @@ TEST_F(AudioDeviceAPITest, SpeakerVolumeTests) {
EXPECT_EQ(-1, audio_device_->MinSpeakerVolume(&minVolume));
EXPECT_EQ(-1, audio_device_->SpeakerVolumeStepSize(&stepSize));
#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// test for warning (can e.g. happen on Vista with Wave API)
EXPECT_EQ(0,
audio_device_->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice));
EXPECT_EQ(0, audio_device_->SpeakerVolumeIsAvailable(&available));
if (available) {
EXPECT_EQ(0, audio_device_->InitSpeaker());
EXPECT_EQ(0, audio_device_->SetSpeakerVolume(19001));
EXPECT_EQ(0, audio_device_->SpeakerVolume(&volume));
WARNING(volume == 19001);
}
#endif
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// use kDefaultCommunicationDevice and modify/retrieve the volume
EXPECT_TRUE(audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1001,7 +909,7 @@ TEST_F(AudioDeviceAPITest, MicrophoneVolumeIsAvailable) {
CheckInitialRecordingStates();
bool available;
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// check the kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1054,21 +962,7 @@ TEST_F(AudioDeviceAPITest, MAYBE_MicrophoneVolumeTests) {
EXPECT_EQ(-1, audio_device_->MinMicrophoneVolume(&minVolume));
EXPECT_EQ(-1, audio_device_->MicrophoneVolumeStepSize(&stepSize));
#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// test for warning (can e.g. happen on Vista with Wave API)
EXPECT_EQ(0, audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultDevice));
EXPECT_EQ(0, audio_device_->MicrophoneVolumeIsAvailable(&available));
if (available)
{
EXPECT_EQ(0, audio_device_->InitMicrophone());
EXPECT_EQ(0, audio_device_->SetMicrophoneVolume(19001));
EXPECT_EQ(0, audio_device_->MicrophoneVolume(&volume));
WARNING(volume == 19001);
}
#endif
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// initialize kDefaultCommunicationDevice and modify/retrieve the volume
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1134,7 +1028,7 @@ TEST_F(AudioDeviceAPITest, MAYBE_MicrophoneVolumeTests) {
TEST_F(AudioDeviceAPITest, SpeakerMuteIsAvailable) {
bool available;
CheckInitialPlayoutStates();
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// check the kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1160,7 +1054,7 @@ TEST_F(AudioDeviceAPITest, SpeakerMuteIsAvailable) {
TEST_F(AudioDeviceAPITest, MicrophoneMuteIsAvailable) {
bool available;
CheckInitialRecordingStates();
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// check the kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1186,7 +1080,7 @@ TEST_F(AudioDeviceAPITest, MicrophoneMuteIsAvailable) {
TEST_F(AudioDeviceAPITest, MicrophoneBoostIsAvailable) {
bool available;
CheckInitialRecordingStates();
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// check the kDefaultCommunicationDevice
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1218,7 +1112,7 @@ TEST_F(AudioDeviceAPITest, SpeakerMuteTests) {
// requires initialization
EXPECT_EQ(-1, audio_device_->SpeakerMute(&enabled));
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// initialize kDefaultCommunicationDevice and modify/retrieve the mute state
EXPECT_EQ(0, audio_device_->SetPlayoutDevice(
AudioDeviceModule::kDefaultCommunicationDevice));
@ -1272,7 +1166,7 @@ TEST_F(AudioDeviceAPITest, MicrophoneMuteTests) {
bool enabled;
EXPECT_EQ(-1, audio_device_->MicrophoneMute(&enabled));
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// initialize kDefaultCommunicationDevice and modify/retrieve the mute
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1326,7 +1220,7 @@ TEST_F(AudioDeviceAPITest, MicrophoneBoostTests) {
// requires initialization
EXPECT_EQ(-1, audio_device_->MicrophoneBoost(&enabled));
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// initialize kDefaultCommunicationDevice and modify/retrieve the boost
EXPECT_TRUE(audio_device_->SetRecordingDevice(
AudioDeviceModule::kDefaultCommunicationDevice) == 0);
@ -1505,7 +1399,8 @@ TEST_F(AudioDeviceAPITest, PlayoutBufferTests) {
CheckInitialPlayoutStates();
EXPECT_EQ(0, audio_device_->PlayoutBuffer(&bufferType, &sizeMS));
#if defined(_WIN32) || defined(ANDROID) || defined(WEBRTC_IOS)
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD) || defined(ANDROID) || \
defined(WEBRTC_IOS)
EXPECT_EQ(AudioDeviceModule::kAdaptiveBufferSize, bufferType);
#else
EXPECT_EQ(AudioDeviceModule::kFixedBufferSize, bufferType);
@ -1532,7 +1427,7 @@ TEST_F(AudioDeviceAPITest, PlayoutBufferTests) {
// bulk tests (all should be successful)
EXPECT_FALSE(audio_device_->PlayoutIsInitialized());
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
EXPECT_EQ(0, audio_device_->SetPlayoutBuffer(
AudioDeviceModule::kAdaptiveBufferSize, 0));
EXPECT_EQ(0, audio_device_->PlayoutBuffer(&bufferType, &sizeMS));
@ -1564,7 +1459,7 @@ TEST_F(AudioDeviceAPITest, PlayoutBufferTests) {
EXPECT_EQ(100, sizeMS);
#endif
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
// restore default
EXPECT_EQ(0, audio_device_->SetPlayoutBuffer(
AudioDeviceModule::kAdaptiveBufferSize, 0));
@ -1596,7 +1491,7 @@ TEST_F(AudioDeviceAPITest, CPULoad) {
uint16_t load(0);
// bulk tests
#ifdef _WIN32
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
EXPECT_EQ(0, audio_device_->CPULoad(&load));
EXPECT_EQ(0, load);
#else

3723
webrtc/modules/audio_device/win/audio_device_wave_win.cc
View File

File diff suppressed because it is too large Load Diff

343
webrtc/modules/audio_device/win/audio_device_wave_win.h
View File

@ -1,343 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
#include <memory>
#include "webrtc/base/platform_thread.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/win/audio_mixer_manager_win.h"
#pragma comment( lib, "winmm.lib" )
namespace webrtc {
class EventTimerWrapper;
class EventWrapper;
const uint32_t TIMER_PERIOD_MS = 2;
const uint32_t REC_CHECK_TIME_PERIOD_MS = 4;
const uint16_t REC_PUT_BACK_DELAY = 4;
const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
const uint32_t N_REC_CHANNELS = 1; // default is mono recording
const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
// NOTE - CPU load will not be correct for other sizes than 10ms
const uint32_t REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC/100);
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC/100);
enum { N_BUFFERS_IN = 200 };
enum { N_BUFFERS_OUT = 200 };
class AudioDeviceWindowsWave : public AudioDeviceGeneric
{
public:
AudioDeviceWindowsWave(const int32_t id);
~AudioDeviceWindowsWave();
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
virtual InitStatus Init();
virtual int32_t Terminate();
virtual bool Initialized() const;
// Device enumeration
virtual int16_t PlayoutDevices();
virtual int16_t RecordingDevices();
virtual int32_t PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]);
virtual int32_t RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]);
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index);
virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
virtual int32_t SetRecordingDevice(uint16_t index);
virtual int32_t SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
// Audio transport initialization
virtual int32_t PlayoutIsAvailable(bool& available);
virtual int32_t InitPlayout();
virtual bool PlayoutIsInitialized() const;
virtual int32_t RecordingIsAvailable(bool& available);
virtual int32_t InitRecording();
virtual bool RecordingIsInitialized() const;
// Audio transport control
virtual int32_t StartPlayout();
virtual int32_t StopPlayout();
virtual bool Playing() const;
virtual int32_t StartRecording();
virtual int32_t StopRecording();
virtual bool Recording() const;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable);
virtual bool AGC() const;
// Volume control based on the Windows Wave API (Windows only)
virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
virtual int32_t WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) const;
// Audio mixer initialization
virtual int32_t InitSpeaker();
virtual bool SpeakerIsInitialized() const;
virtual int32_t InitMicrophone();
virtual bool MicrophoneIsInitialized() const;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool& available);
virtual int32_t SetSpeakerVolume(uint32_t volume);
virtual int32_t SpeakerVolume(uint32_t& volume) const;
virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
virtual int32_t SetMicrophoneVolume(uint32_t volume);
virtual int32_t MicrophoneVolume(uint32_t& volume) const;
virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool& available);
virtual int32_t SetSpeakerMute(bool enable);
virtual int32_t SpeakerMute(bool& enabled) const;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool& available);
virtual int32_t SetMicrophoneMute(bool enable);
virtual int32_t MicrophoneMute(bool& enabled) const;
// Microphone boost control
virtual int32_t MicrophoneBoostIsAvailable(bool& available);
virtual int32_t SetMicrophoneBoost(bool enable);
virtual int32_t MicrophoneBoost(bool& enabled) const;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool& available);
virtual int32_t SetStereoPlayout(bool enable);
virtual int32_t StereoPlayout(bool& enabled) const;
virtual int32_t StereoRecordingIsAvailable(bool& available);
virtual int32_t SetStereoRecording(bool enable);
virtual int32_t StereoRecording(bool& enabled) const;
// Delay information and control
virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS);
virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const;
virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
virtual int32_t RecordingDelay(uint16_t& delayMS) const;
// CPU load
virtual int32_t CPULoad(uint16_t& load) const;
public:
virtual bool PlayoutWarning() const;
virtual bool PlayoutError() const;
virtual bool RecordingWarning() const;
virtual bool RecordingError() const;
virtual void ClearPlayoutWarning();
virtual void ClearPlayoutError();
virtual void ClearRecordingWarning();
virtual void ClearRecordingError();
public:
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
private:
void Lock() { _critSect.Enter(); };
void UnLock() { _critSect.Leave(); };
int32_t Id() {return _id;}
bool IsUsingOutputDeviceIndex() const {return _usingOutputDeviceIndex;}
AudioDeviceModule::WindowsDeviceType OutputDevice() const {return _outputDevice;}
uint16_t OutputDeviceIndex() const {return _outputDeviceIndex;}
bool IsUsingInputDeviceIndex() const {return _usingInputDeviceIndex;}
AudioDeviceModule::WindowsDeviceType InputDevice() const {return _inputDevice;}
uint16_t InputDeviceIndex() const {return _inputDeviceIndex;}
private:
inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
private:
bool KeyPressed() const;
private:
int32_t EnumeratePlayoutDevices();
int32_t EnumerateRecordingDevices();
void TraceSupportFlags(DWORD dwSupport) const;
void TraceWaveInError(MMRESULT error) const;
void TraceWaveOutError(MMRESULT error) const;
int32_t PrepareStartRecording();
int32_t PrepareStartPlayout();
int32_t RecProc(LONGLONG& consumedTime);
int PlayProc(LONGLONG& consumedTime);
int32_t GetPlayoutBufferDelay(uint32_t& writtenSamples, uint32_t& playedSamples);
int32_t GetRecordingBufferDelay(uint32_t& readSamples, uint32_t& recSamples);
int32_t Write(int8_t* data, uint16_t nSamples);
int32_t GetClockDrift(const uint32_t plSamp, const uint32_t rcSamp);
int32_t MonitorRecording(const uint32_t time);
int32_t RestartTimerIfNeeded(const uint32_t time);
private:
static bool ThreadFunc(void*);
bool ThreadProcess();
static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
DWORD DoGetCaptureVolumeThread();
static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
DWORD DoSetCaptureVolumeThread();
private:
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
EventTimerWrapper& _timeEvent;
EventWrapper& _recStartEvent;
EventWrapper& _playStartEvent;
HANDLE _hGetCaptureVolumeThread;
HANDLE _hShutdownGetVolumeEvent;
HANDLE _hSetCaptureVolumeThread;
HANDLE _hShutdownSetVolumeEvent;
HANDLE _hSetCaptureVolumeEvent;
// TODO(pbos): Remove unique_ptr usage and use PlatformThread directly
std::unique_ptr<rtc::PlatformThread> _ptrThread;
CriticalSectionWrapper& _critSectCb;
int32_t _id;
AudioMixerManager _mixerManager;
bool _usingInputDeviceIndex;
bool _usingOutputDeviceIndex;
AudioDeviceModule::WindowsDeviceType _inputDevice;
AudioDeviceModule::WindowsDeviceType _outputDevice;
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
WAVEFORMATEX _waveFormatIn;
WAVEFORMATEX _waveFormatOut;
HWAVEIN _hWaveIn;
HWAVEOUT _hWaveOut;
WAVEHDR _waveHeaderIn[N_BUFFERS_IN];
WAVEHDR _waveHeaderOut[N_BUFFERS_OUT];
uint8_t _recChannels;
uint8_t _playChannels;
uint16_t _recBufCount;
uint16_t _recDelayCount;
uint16_t _recPutBackDelay;
int8_t _recBuffer[N_BUFFERS_IN][4*REC_BUF_SIZE_IN_SAMPLES];
int8_t _playBuffer[N_BUFFERS_OUT][4*PLAY_BUF_SIZE_IN_SAMPLES];
AudioDeviceModule::BufferType _playBufType;
private:
bool _initialized;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _startRec;
bool _stopRec;
bool _startPlay;
bool _stopPlay;
bool _AGC;
private:
uint32_t _prevPlayTime;
uint32_t _prevRecTime;
uint32_t _prevTimerCheckTime;
uint16_t _playBufCount; // playout buffer index
uint16_t _dTcheckPlayBufDelay; // dT for check of play buffer, {2,5,10} [ms]
uint16_t _playBufDelay; // playback delay
uint16_t _playBufDelayFixed; // fixed playback delay
uint16_t _minPlayBufDelay; // minimum playback delay
uint16_t _MAX_minBuffer; // level of (adaptive) min threshold must be < _MAX_minBuffer
int32_t _erZeroCounter; // counts "buffer-is-empty" events
int32_t _intro;
int32_t _waitCounter;
uint32_t _writtenSamples;
uint32_t _writtenSamplesOld;
uint32_t _playedSamplesOld;
uint32_t _sndCardPlayDelay;
uint32_t _sndCardRecDelay;
uint32_t _plSampOld;
uint32_t _rcSampOld;
uint32_t _read_samples;
uint32_t _read_samples_old;
uint32_t _rec_samples_old;
// State that detects driver problems:
int32_t _dc_diff_mean;
int32_t _dc_y_prev;
int32_t _dc_penalty_counter;
int32_t _dc_prevtime;
uint32_t _dc_prevplay;
uint32_t _recordedBytes; // accumulated #recorded bytes (reset periodically)
uint32_t _prevRecByteCheckTime; // time when we last checked the recording process
// CPU load measurements
LARGE_INTEGER _perfFreq;
LONGLONG _playAcc; // accumulated time for playout callback
float _avgCPULoad; // average total (rec+play) CPU load
int32_t _wrapCounter;
int32_t _useHeader;
int16_t _timesdwBytes;
int32_t _no_of_msecleft_warnings;
int32_t _writeErrors;
int32_t _timerFaults;
int32_t _timerRestartAttempts;
uint16_t _playWarning;
uint16_t _playError;
uint16_t _recWarning;
uint16_t _recError;
uint32_t _newMicLevel;
uint32_t _minMicVolume;
uint32_t _maxMicVolume;
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H

6
webrtc/voice_engine/voe_hardware_impl.cc
View File

@ -57,9 +57,6 @@ int VoEHardwareImpl::SetAudioDeviceLayer(AudioLayers audioLayer) {
case kAudioWindowsCore:
wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
break;
case kAudioWindowsWave:
wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
break;
case kAudioLinuxAlsa:
wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
break;
@ -100,9 +97,6 @@ int VoEHardwareImpl::GetAudioDeviceLayer(AudioLayers& audioLayer) {
case AudioDeviceModule::kWindowsCoreAudio:
audioLayer = kAudioWindowsCore;
break;
case AudioDeviceModule::kWindowsWaveAudio:
audioLayer = kAudioWindowsWave;
break;
case AudioDeviceModule::kLinuxAlsaAudio:
audioLayer = kAudioLinuxAlsa;
break;