Remove the Windows Wave audio device implementation.
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-Original-Commit-Position: refs/heads/master@{#16678}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: cc8588c0400df7c03f305da0a96c32bae3b5b2fa
diff --git a/common_types.h b/common_types.h
index 73e8364..faa875a 100644
--- a/common_types.h
+++ b/common_types.h
@@ -467,7 +467,6 @@
// Audio device layers
enum AudioLayers {
kAudioPlatformDefault = 0,
- kAudioWindowsWave = 1,
kAudioWindowsCore = 2,
kAudioLinuxAlsa = 3,
kAudioLinuxPulse = 4
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index ac8fdbe..216ae0f 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -211,8 +211,6 @@
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",
]
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index ad643e9..5040189 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -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 @@
// 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 @@
// 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.
diff --git a/modules/audio_device/include/audio_device.h b/modules/audio_device/include/audio_device.h
index f38a24a..dfa58aa 100644
--- a/modules/audio_device/include/audio_device.h
+++ b/modules/audio_device/include/audio_device.h
@@ -26,7 +26,6 @@
enum AudioLayer {
kPlatformDefaultAudio = 0,
- kWindowsWaveAudio = 1,
kWindowsCoreAudio = 2,
kLinuxAlsaAudio = 3,
kLinuxPulseAudio = 4,
diff --git a/modules/audio_device/test/audio_device_test_api.cc b/modules/audio_device/test/audio_device_test_api.cc
index 2f27806..e36fc1d 100644
--- a/modules/audio_device/test/audio_device_test_api.cc
+++ b/modules/audio_device/test/audio_device_test_api.cc
@@ -163,46 +163,23 @@
// 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(
kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
@@ -213,8 +190,6 @@
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
EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
@@ -229,8 +204,6 @@
#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(
kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
@@ -471,7 +444,7 @@
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, 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 @@
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 @@
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 @@
}
}
-#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 @@
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 @@
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 @@
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 @@
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, 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, 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, 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 @@
// 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 @@
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 @@
// 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 @@
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 @@
// 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 @@
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 @@
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
diff --git a/modules/audio_device/win/audio_device_wave_win.cc b/modules/audio_device/win/audio_device_wave_win.cc
deleted file mode 100644
index 7874575..0000000
--- a/modules/audio_device/win/audio_device_wave_win.cc
+++ /dev/null
@@ -1,3723 +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 "webrtc/base/logging.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/win/audio_device_wave_win.h"
-
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-#include <windows.h>
-#include <objbase.h> // CoTaskMemAlloc, CoTaskMemFree
-#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
-#include <assert.h>
-
-// Avoids the need of Windows 7 SDK
-#ifndef WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE
-#define WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE 0x0010
-#endif
-
-// Supported in Windows Vista and Windows 7.
-// http://msdn.microsoft.com/en-us/library/dd370819(v=VS.85).aspx
-// Taken from Mmddk.h.
-#define DRV_RESERVED 0x0800
-#define DRV_QUERYFUNCTIONINSTANCEID (DRV_RESERVED + 17)
-#define DRV_QUERYFUNCTIONINSTANCEIDSIZE (DRV_RESERVED + 18)
-
-#define POW2(A) (2 << ((A) - 1))
-
-namespace webrtc {
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsWave - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsWave::AudioDeviceWindowsWave(const int32_t id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _timeEvent(*EventTimerWrapper::Create()),
- _recStartEvent(*EventWrapper::Create()),
- _playStartEvent(*EventWrapper::Create()),
- _hGetCaptureVolumeThread(NULL),
- _hShutdownGetVolumeEvent(NULL),
- _hSetCaptureVolumeThread(NULL),
- _hShutdownSetVolumeEvent(NULL),
- _hSetCaptureVolumeEvent(NULL),
- _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _mixerManager(id),
- _usingInputDeviceIndex(false),
- _usingOutputDeviceIndex(false),
- _inputDevice(AudioDeviceModule::kDefaultDevice),
- _outputDevice(AudioDeviceModule::kDefaultDevice),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _initialized(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _recording(false),
- _playing(false),
- _startRec(false),
- _stopRec(false),
- _startPlay(false),
- _stopPlay(false),
- _AGC(false),
- _hWaveIn(NULL),
- _hWaveOut(NULL),
- _recChannels(N_REC_CHANNELS),
- _playChannels(N_PLAY_CHANNELS),
- _recBufCount(0),
- _recPutBackDelay(0),
- _recDelayCount(0),
- _playBufCount(0),
- _prevPlayTime(0),
- _prevRecTime(0),
- _prevTimerCheckTime(0),
- _timesdwBytes(0),
- _timerFaults(0),
- _timerRestartAttempts(0),
- _no_of_msecleft_warnings(0),
- _MAX_minBuffer(65),
- _useHeader(0),
- _dTcheckPlayBufDelay(10),
- _playBufDelay(80),
- _playBufDelayFixed(80),
- _minPlayBufDelay(20),
- _avgCPULoad(0),
- _sndCardPlayDelay(0),
- _sndCardRecDelay(0),
- _plSampOld(0),
- _rcSampOld(0),
- _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
- _recordedBytes(0),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _newMicLevel(0),
- _minMicVolume(0),
- _maxMicVolume(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-
- // Initialize value, set to 0 if it fails
- if (!QueryPerformanceFrequency(&_perfFreq))
- {
- _perfFreq.QuadPart = 0;
- }
-
- _hShutdownGetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hShutdownSetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsWave - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsWave::~AudioDeviceWindowsWave()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- delete &_recStartEvent;
- delete &_playStartEvent;
- delete &_timeEvent;
- delete &_critSect;
- delete &_critSectCb;
-
- if (NULL != _hShutdownGetVolumeEvent)
- {
- CloseHandle(_hShutdownGetVolumeEvent);
- _hShutdownGetVolumeEvent = NULL;
- }
-
- if (NULL != _hShutdownSetVolumeEvent)
- {
- CloseHandle(_hShutdownSetVolumeEvent);
- _hShutdownSetVolumeEvent = NULL;
- }
-
- if (NULL != _hSetCaptureVolumeEvent)
- {
- CloseHandle(_hSetCaptureVolumeEvent);
- _hSetCaptureVolumeEvent = NULL;
- }
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // inform the AudioBuffer about default settings for this implementation
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
- _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kWindowsWaveAudio;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-AudioDeviceGeneric::InitStatus AudioDeviceWindowsWave::Init() {
- CriticalSectionScoped lock(&_critSect);
-
- if (_initialized) {
- return InitStatus::OK;
- }
-
- const uint32_t nowTime(rtc::TimeMillis());
-
- _recordedBytes = 0;
- _prevRecByteCheckTime = nowTime;
- _prevRecTime = nowTime;
- _prevPlayTime = nowTime;
- _prevTimerCheckTime = nowTime;
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- _mixerManager.EnumerateAll();
-
- if (_ptrThread) {
- // thread is already created and active
- return InitStatus::OK;
- }
-
- const char* threadName = "webrtc_audio_module_thread";
- _ptrThread.reset(new rtc::PlatformThread(ThreadFunc, this, threadName));
- _ptrThread->Start();
- _ptrThread->SetPriority(rtc::kRealtimePriority);
-
- const bool periodic(true);
- if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS)) {
- LOG(LS_ERROR) << "failed to start the timer event";
- _ptrThread->Stop();
- _ptrThread.reset();
- return InitStatus::OTHER_ERROR;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
-
- _hGetCaptureVolumeThread =
- CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
- if (_hGetCaptureVolumeThread == NULL) {
- LOG(LS_ERROR) << " failed to create the volume getter thread";
- return InitStatus::OTHER_ERROR;
- }
-
- SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
-
- _hSetCaptureVolumeThread =
- CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
- if (_hSetCaptureVolumeThread == NULL) {
- LOG(LS_ERROR) << " failed to create the volume setter thread";
- return InitStatus::OTHER_ERROR;
- }
-
- SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
-
- _initialized = true;
-
- return InitStatus::OK;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::Terminate()
-{
-
- if (!_initialized)
- {
- return 0;
- }
-
- _critSect.Enter();
-
- _mixerManager.Close();
-
- if (_ptrThread)
- {
- rtc::PlatformThread* tmpThread = _ptrThread.release();
- _critSect.Leave();
-
- _timeEvent.Set();
-
- tmpThread->Stop();
- delete tmpThread;
- }
- else
- {
- _critSect.Leave();
- }
-
- _critSect.Enter();
- SetEvent(_hShutdownGetVolumeEvent);
- _critSect.Leave();
- int32_t ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume getter thread");
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
- return -1;
- }
- _critSect.Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume getter thread is now closed");
-
- SetEvent(_hShutdownSetVolumeEvent);
- _critSect.Leave();
- ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume setter thread");
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
- return -1;
- }
- _critSect.Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume setter thread is now closed");
-
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
-
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
-
- _critSect.Leave();
-
- _timeEvent.StopTimer();
-
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
-
- return 0;
-}
-
-
-DWORD WINAPI AudioDeviceWindowsWave::GetCaptureVolumeThread(LPVOID context)
-{
- return(((AudioDeviceWindowsWave*)context)->DoGetCaptureVolumeThread());
-}
-
-DWORD WINAPI AudioDeviceWindowsWave::SetCaptureVolumeThread(LPVOID context)
-{
- return(((AudioDeviceWindowsWave*)context)->DoSetCaptureVolumeThread());
-}
-
-DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
-{
- HANDLE waitObject = _hShutdownGetVolumeEvent;
-
- while (1)
- {
- DWORD waitResult = WaitForSingleObject(waitObject,
- GET_MIC_VOLUME_INTERVAL_MS);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownGetVolumeEvent
- return 0;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on get volume thread");
- return 1;
- }
-
- if (AGC())
- {
- uint32_t currentMicLevel = 0;
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // This doesn't set the system volume, just stores it.
- _critSect.Enter();
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- _critSect.Leave();
- }
- }
- }
-}
-
-DWORD AudioDeviceWindowsWave::DoSetCaptureVolumeThread()
-{
- HANDLE waitArray[2] = {_hShutdownSetVolumeEvent, _hSetCaptureVolumeEvent};
-
- while (1)
- {
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownSetVolumeEvent
- return 0;
- case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on set volume thread");
- return 1;
- }
-
- _critSect.Enter();
- uint32_t newMicLevel = _newMicLevel;
- _critSect.Leave();
-
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the microphone volume failed");
- }
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Initialized() const
-{
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::InitSpeaker()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (_mixerManager.EnumerateSpeakers() == -1)
- {
- // failed to locate any valid/controllable speaker
- return -1;
- }
-
- if (IsUsingOutputDeviceIndex())
- {
- if (_mixerManager.OpenSpeaker(OutputDeviceIndex()) == -1)
- {
- return -1;
- }
- }
- else
- {
- if (_mixerManager.OpenSpeaker(OutputDevice()) == -1)
- {
- return -1;
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::InitMicrophone()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (_mixerManager.EnumerateMicrophones() == -1)
- {
- // failed to locate any valid/controllable microphone
- return -1;
- }
-
- if (IsUsingInputDeviceIndex())
- {
- if (_mixerManager.OpenMicrophone(InputDeviceIndex()) == -1)
- {
- return -1;
- }
- }
- else
- {
- if (_mixerManager.OpenMicrophone(InputDevice()) == -1)
- {
- return -1;
- }
- }
-
- uint32_t maxVol = 0;
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unable to retrieve max microphone volume");
- }
- _maxMicVolume = maxVol;
-
- uint32_t minVol = 0;
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unable to retrieve min microphone volume");
- }
- _minMicVolume = minVol;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable speakers and make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (InitSpeaker() == -1)
- {
- // failed to find a valid speaker
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a volume control
- //
- _mixerManager.SpeakerVolumeIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized output mixer
- //
- _mixerManager.CloseSpeaker();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetSpeakerVolume(uint32_t volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SpeakerVolume(uint32_t& volume) const
-{
-
- uint32_t level(0);
-
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-//
-// The low-order word contains the left-channel volume setting, and the
-// high-order word contains the right-channel setting.
-// A value of 0xFFFF represents full volume, and a value of 0x0000 is silence.
-//
-// If a device does not support both left and right volume control,
-// the low-order word of dwVolume specifies the volume level,
-// and the high-order word is ignored.
-//
-// Most devices do not support the full 16 bits of volume-level control
-// and will not use the least-significant bits of the requested volume setting.
-// For example, if a device supports 4 bits of volume control, the values
-// 0x4000, 0x4FFF, and 0x43BE will all be truncated to 0x4000.
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight)
-{
-
- MMRESULT res(0);
- WAVEOUTCAPS caps;
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
- }
-
- // To determine whether the device supports volume control on both
- // the left and right channels, use the WAVECAPS_LRVOLUME flag.
- //
- res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- if (!(caps.dwSupport & WAVECAPS_VOLUME))
- {
- // this device does not support volume control using the waveOutSetVolume API
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
- return -1;
- }
- if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
- {
- // high-order word (right channel) is ignored
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
- }
-
- DWORD dwVolume(0x00000000);
- dwVolume = (DWORD)(((volumeRight & 0xFFFF) << 16) | (volumeLeft & 0xFFFF));
-
- res = waveOutSetVolume(_hWaveOut, dwVolume);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutSetVolume() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-//
-// The low-order word of this location contains the left-channel volume setting,
-// and the high-order word contains the right-channel setting.
-// A value of 0xFFFF (65535) represents full volume, and a value of 0x0000
-// is silence.
-//
-// If a device does not support both left and right volume control,
-// the low-order word of the specified location contains the mono volume level.
-//
-// The full 16-bit setting(s) set with the waveOutSetVolume function is returned,
-// regardless of whether the device supports the full 16 bits of volume-level
-// control.
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) const
-{
-
- MMRESULT res(0);
- WAVEOUTCAPS caps;
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
- }
-
- // To determine whether the device supports volume control on both
- // the left and right channels, use the WAVECAPS_LRVOLUME flag.
- //
- res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- if (!(caps.dwSupport & WAVECAPS_VOLUME))
- {
- // this device does not support volume control using the waveOutSetVolume API
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
- return -1;
- }
- if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
- {
- // high-order word (right channel) is ignored
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
- }
-
- DWORD dwVolume(0x00000000);
-
- res = waveOutGetVolume(_hWaveOut, &dwVolume);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutGetVolume() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- WORD wVolumeLeft = LOWORD(dwVolume);
- WORD wVolumeRight = HIWORD(dwVolume);
-
- volumeLeft = static_cast<uint16_t> (wVolumeLeft);
- volumeRight = static_cast<uint16_t> (wVolumeRight);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
-
- uint32_t maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MinSpeakerVolume(uint32_t& minVolume) const
-{
-
- uint32_t minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
-
- uint16_t delta(0);
-
- if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable speakers and make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- //
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized output mixer
- //
- _mixerManager.CloseSpeaker();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetSpeakerMute(bool enable)
-{
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetMicrophoneMute(bool enable)
-{
- return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneBoostIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a boost control
- //
- _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetMicrophoneBoost(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneBoost(bool& enabled) const
-{
-
- bool onOff(0);
-
- if (_mixerManager.MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- enabled = onOff;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StereoRecordingIsAvailable(bool& available)
-{
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetStereoRecording(bool enable)
-{
-
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StereoPlayoutIsAvailable(bool& available)
-{
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-//
-// Specifies the number of output channels.
-//
-// NOTE - the setting will only have an effect after InitPlayout has
-// been called.
-//
-// 16-bit mono:
-//
-// Each sample is 2 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
-// For each sample, the first byte is the low-order byte of channel 0 and the
-// second byte is the high-order byte of channel 0.
-//
-// 16-bit stereo:
-//
-// Each sample is 4 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
-// For each sample, the first byte is the low-order byte of channel 0 (left channel);
-// the second byte is the high-order byte of channel 0; the third byte is the
-// low-order byte of channel 1 (right channel); and the fourth byte is the
-// high-order byte of channel 1.
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::AGC() const
-{
- return _AGC;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (InitMicrophone() == -1)
- {
- // Failed to find valid microphone
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a volume control
- //
- _mixerManager.MicrophoneVolumeIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetMicrophoneVolume(uint32_t volume)
-{
- return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneVolume(uint32_t& volume) const
-{
- uint32_t level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to retrive current microphone level");
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
- // _maxMicVolume can be zero in AudioMixerManager::MaxMicrophoneVolume():
- // (1) API GetLineControl() returns failure at querying the max Mic level.
- // (2) API GetLineControl() returns maxVolume as zero in rare cases.
- // Both cases show we don't have access to the mixer controls.
- // We return -1 here to indicate that.
- if (_maxMicVolume == 0)
- {
- return -1;
- }
-
- maxVolume = _maxMicVolume;;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MinMicrophoneVolume(uint32_t& minVolume) const
-{
- minVolume = _minMicVolume;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
-
- uint16_t delta(0);
-
- if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-int16_t AudioDeviceWindowsWave::PlayoutDevices()
-{
-
- return (waveOutGetNumDevs());
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetPlayoutDevice(uint16_t index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- UINT nDevices = waveOutGetNumDevs();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio output devices is %u", nDevices);
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _usingOutputDeviceIndex = true;
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_playIsInitialized)
- {
- return -1;
- }
-
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- }
-
- _usingOutputDeviceIndex = false;
- _outputDevice = device;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PlayoutDeviceName(
- uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
-
- uint16_t nDevices(PlayoutDevices());
-
- // Special fix for the case when the user asks for the name of the default device.
- //
- if (index == (uint16_t)(-1))
- {
- index = 0;
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- WAVEOUTCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
- MMRESULT res;
-
- res = waveOutGetDevCapsW(index, &caps, sizeof(WAVEOUTCAPSW));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCapsW() failed (err=%d)", res);
- return -1;
- }
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
- }
-
- if (guid == NULL)
- {
- return 0;
- }
-
- // It is possible to get the unique endpoint ID string using the Wave API.
- // However, it is only supported on Windows Vista and Windows 7.
-
- size_t cbEndpointId(0);
-
- // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
- // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
- res = waveOutMessage((HWAVEOUT)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEIDSIZE,
- (DWORD_PTR)&cbEndpointId, NULL);
- if (res != MMSYSERR_NOERROR)
- {
- // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
- TraceWaveOutError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
- }
- return 0;
- }
-
- // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
-
- WCHAR *pstrEndpointId = NULL;
- pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
-
- // Get the endpoint ID string for this waveOut device.
- res = waveOutMessage((HWAVEOUT)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEID,
- (DWORD_PTR)pstrEndpointId,
- cbEndpointId);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
- TraceWaveOutError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
- return 0;
- }
-
- if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::RecordingDeviceName(
- uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
-
- uint16_t nDevices(RecordingDevices());
-
- // Special fix for the case when the user asks for the name of the default device.
- //
- if (index == (uint16_t)(-1))
- {
- index = 0;
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- WAVEINCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
- MMRESULT res;
-
- res = waveInGetDevCapsW(index, &caps, sizeof(WAVEINCAPSW));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCapsW() failed (err=%d)", res);
- return -1;
- }
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
- }
-
- if (guid == NULL)
- {
- return 0;
- }
-
- // It is possible to get the unique endpoint ID string using the Wave API.
- // However, it is only supported on Windows Vista and Windows 7.
-
- size_t cbEndpointId(0);
-
- // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
- // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
- res = waveInMessage((HWAVEIN)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEIDSIZE,
- (DWORD_PTR)&cbEndpointId, NULL);
- if (res != MMSYSERR_NOERROR)
- {
- // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
- TraceWaveInError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
- }
- return 0;
- }
-
- // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
-
- WCHAR *pstrEndpointId = NULL;
- pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
-
- // Get the endpoint ID string for this waveOut device.
- res = waveInMessage((HWAVEIN)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEID,
- (DWORD_PTR)pstrEndpointId,
- cbEndpointId);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
- TraceWaveInError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
- return 0;
- }
-
- if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-int16_t AudioDeviceWindowsWave::RecordingDevices()
-{
-
- return (waveInGetNumDevs());
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetRecordingDevice(uint16_t index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- UINT nDevices = waveInGetNumDevs();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio input devices is %u", nDevices);
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _usingInputDeviceIndex = true;
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- }
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- _usingInputDeviceIndex = false;
- _inputDevice = device;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- int32_t res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the recording side
- int32_t res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::InitPlayout()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
- }
-
- // Enumerate all availiable output devices
- EnumeratePlayoutDevices();
-
- // Start by closing any existing wave-output devices
- //
- MMRESULT res(MMSYSERR_ERROR);
-
- if (_hWaveOut != NULL)
- {
- res = waveOutClose(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- }
-
- // Set the output wave format
- //
- WAVEFORMATEX waveFormat;
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = _playChannels; // mono <=> 1, stereo <=> 2
- waveFormat.nSamplesPerSec = N_PLAY_SAMPLES_PER_SEC;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // Open the given waveform-audio output device for playout
- //
- HWAVEOUT hWaveOut(NULL);
-
- if (IsUsingOutputDeviceIndex())
- {
- // verify settings first
- res = waveOutOpen(NULL, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // open the given waveform-audio output device for recording
- res = waveOutOpen(&hWaveOut, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening output device corresponding to device ID %u", _outputDeviceIndex);
- }
- }
- else
- {
- if (_outputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
- }
- }
- else if (_outputDevice == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
- }
- }
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- // Log information about the aquired output device
- //
- WAVEOUTCAPS caps;
-
- res = waveOutGetDevCaps((UINT_PTR)hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- UINT deviceID(0);
- res = waveOutGetID(hWaveOut, &deviceID);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetID() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
-
- // Store valid handle for the open waveform-audio output device
- _hWaveOut = hWaveOut;
-
- // Store the input wave header as well
- _waveFormatOut = waveFormat;
-
- // Prepare wave-out headers
- //
- const uint8_t bytesPerSample = 2*_playChannels;
-
- for (int n = 0; n < N_BUFFERS_OUT; n++)
- {
- // set up the output wave header
- _waveHeaderOut[n].lpData = reinterpret_cast<LPSTR>(&_playBuffer[n]);
- _waveHeaderOut[n].dwBufferLength = bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES;
- _waveHeaderOut[n].dwFlags = 0;
- _waveHeaderOut[n].dwLoops = 0;
-
- memset(_playBuffer[n], 0, bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES);
-
- // The waveOutPrepareHeader function prepares a waveform-audio data block for playback.
- // The lpData, dwBufferLength, and dwFlags members of the WAVEHDR structure must be set
- // before calling this function.
- //
- res = waveOutPrepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (err=%d)", n, res);
- TraceWaveOutError(res);
- }
-
- // perform extra check to ensure that the header is prepared
- if (_waveHeaderOut[n].dwFlags != WHDR_PREPARED)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (dwFlags != WHDR_PREPARED)", n);
- }
- }
-
- // Mark playout side as initialized
- _playIsInitialized = true;
-
- _dTcheckPlayBufDelay = 10; // check playback buffer delay every 10 ms
- _playBufCount = 0; // index of active output wave header (<=> output buffer index)
- _playBufDelay = 80; // buffer delay/size is initialized to 80 ms and slowly decreased until er < 25
- _minPlayBufDelay = 25; // minimum playout buffer delay
- _MAX_minBuffer = 65; // adaptive minimum playout buffer delay cannot be larger than this value
- _intro = 1; // Used to make sure that adaption starts after (2000-1700)/100 seconds
- _waitCounter = 1700; // Counter for start of adaption of playback buffer
- _erZeroCounter = 0; // Log how many times er = 0 in consequtive calls to RecTimeProc
- _useHeader = 0; // Counts number of "useHeader" detections. Stops at 2.
-
- _writtenSamples = 0;
- _writtenSamplesOld = 0;
- _playedSamplesOld = 0;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id,"initial playout status: _playBufDelay=%d, _minPlayBufDelay=%d",
- _playBufDelay, _minPlayBufDelay);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::InitRecording()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- _avgCPULoad = 0;
- _playAcc = 0;
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
- }
-
- // Enumerate all availiable input devices
- EnumerateRecordingDevices();
-
- // Start by closing any existing wave-input devices
- //
- MMRESULT res(MMSYSERR_ERROR);
-
- if (_hWaveIn != NULL)
- {
- res = waveInClose(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- }
-
- // Set the input wave format
- //
- WAVEFORMATEX waveFormat;
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = _recChannels; // mono <=> 1, stereo <=> 2
- waveFormat.nSamplesPerSec = N_REC_SAMPLES_PER_SEC;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // Open the given waveform-audio input device for recording
- //
- HWAVEIN hWaveIn(NULL);
-
- if (IsUsingInputDeviceIndex())
- {
- // verify settings first
- res = waveInOpen(NULL, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // open the given waveform-audio input device for recording
- res = waveInOpen(&hWaveIn, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening input device corresponding to device ID %u", _inputDeviceIndex);
- }
- }
- else
- {
- if (_inputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
- }
- }
- else if (_inputDevice == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
- }
- }
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
- TraceWaveInError(res);
- return -1;
- }
-
- // Log information about the aquired input device
- //
- WAVEINCAPS caps;
-
- res = waveInGetDevCaps((UINT_PTR)hWaveIn, &caps, sizeof(WAVEINCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- UINT deviceID(0);
- res = waveInGetID(hWaveIn, &deviceID);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetID() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
-
- // Store valid handle for the open waveform-audio input device
- _hWaveIn = hWaveIn;
-
- // Store the input wave header as well
- _waveFormatIn = waveFormat;
-
- // Mark recording side as initialized
- _recIsInitialized = true;
-
- _recBufCount = 0; // index of active input wave header (<=> input buffer index)
- _recDelayCount = 0; // ensures that input buffers are returned with certain delay
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- // set state to ensure that the recording starts from the audio thread
- _startRec = true;
-
- // the audio thread will signal when recording has stopped
- if (kEventTimeout == _recStartEvent.Wait(10000))
- {
- _startRec = false;
- StopRecording();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
- return -1;
- }
-
- if (_recording)
- {
- // the recording state is set by the audio thread after recording has started
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StopRecording()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- if (_hWaveIn == NULL)
- {
- return -1;
- }
-
- bool wasRecording = _recording;
- _recIsInitialized = false;
- _recording = false;
-
- MMRESULT res;
-
- // Stop waveform-adio input. If there are any buffers in the queue, the
- // current buffer will be marked as done (the dwBytesRecorded member in
- // the header will contain the length of data), but any empty buffers in
- // the queue will remain there.
- //
- res = waveInStop(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStop() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Stop input on the given waveform-audio input device and resets the current
- // position to zero. All pending buffers are marked as done and returned to
- // the application.
- //
- res = waveInReset(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInReset() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Clean up the preparation performed by the waveInPrepareHeader function.
- // Only unprepare header if recording was ever started (and headers are prepared).
- //
- if (wasRecording)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInUnprepareHeader() will be performed");
- for (int n = 0; n < N_BUFFERS_IN; n++)
- {
- res = waveInUnprepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- }
- }
-
- // Close the given waveform-audio input device.
- //
- res = waveInClose(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Set the wave input handle to NULL
- //
- _hWaveIn = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveIn is now set to NULL");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Recording() const
-{
- return (_recording);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StartPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- // set state to ensure that playout starts from the audio thread
- _startPlay = true;
-
- // the audio thread will signal when recording has started
- if (kEventTimeout == _playStartEvent.Wait(10000))
- {
- _startPlay = false;
- StopPlayout();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playout");
- return -1;
- }
-
- if (_playing)
- {
- // the playing state is set by the audio thread after playout has started
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playing");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::StopPlayout()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- _playIsInitialized = false;
- _playing = false;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- MMRESULT res;
-
- // The waveOutReset function stops playback on the given waveform-audio
- // output device and resets the current position to zero. All pending
- // playback buffers are marked as done (WHDR_DONE) and returned to the application.
- // After this function returns, the application can send new playback buffers
- // to the device by calling waveOutWrite, or close the device by calling waveOutClose.
- //
- res = waveOutReset(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutReset() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- // The waveOutUnprepareHeader function cleans up the preparation performed
- // by the waveOutPrepareHeader function. This function must be called after
- // the device driver is finished with a data block.
- // You must call this function before freeing the buffer.
- //
- for (int n = 0; n < N_BUFFERS_OUT; n++)
- {
- res = waveOutUnprepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutUnprepareHeader() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- }
-
- // The waveOutClose function closes the given waveform-audio output device.
- // The close operation fails if the device is still playing a waveform-audio
- // buffer that was previously sent by calling waveOutWrite. Before calling
- // waveOutClose, the application must wait for all buffers to finish playing
- // or call the waveOutReset function to terminate playback.
- //
- res = waveOutClose(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- _hWaveOut = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveOut is now set to NULL");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PlayoutDelay(uint16_t& delayMS) const
-{
- CriticalSectionScoped lock(&_critSect);
- delayMS = (uint16_t)_sndCardPlayDelay;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::RecordingDelay(uint16_t& delayMS) const
-{
- CriticalSectionScoped lock(&_critSect);
- delayMS = (uint16_t)_sndCardRecDelay;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Playing() const
-{
- return (_playing);
-}
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS)
-{
- CriticalSectionScoped lock(&_critSect);
- _playBufType = type;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- _playBufDelayFixed = sizeMS;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const
-{
- CriticalSectionScoped lock(&_critSect);
- type = _playBufType;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- sizeMS = _playBufDelayFixed;
- }
- else
- {
- sizeMS = _playBufDelay;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::CPULoad(uint16_t& load) const
-{
-
- load = static_cast<uint16_t>(100*_avgCPULoad);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutWarning() const
-{
- return ( _playWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutError() const
-{
- return ( _playError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingWarning() const
-{
- return ( _recWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingError() const
-{
- return ( _recError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// InputSanityCheckAfterUnlockedPeriod
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::InputSanityCheckAfterUnlockedPeriod() const
-{
- if (_hWaveIn == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OutputSanityCheckAfterUnlockedPeriod
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::OutputSanityCheckAfterUnlockedPeriod() const
-{
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumeratePlayoutDevices
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::EnumeratePlayoutDevices()
-{
-
- uint16_t nDevices(PlayoutDevices());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#output devices: %u", nDevices);
-
- WAVEOUTCAPS caps;
- MMRESULT res;
-
- for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
- {
- res = waveOutGetDevCaps(deviceID, &caps, sizeof(WAVEOUTCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
- if (caps.dwFormats & WAVE_FORMAT_48S16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
- }
- if (caps.dwFormats & WAVE_FORMAT_48M16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
- TraceSupportFlags(caps.dwSupport);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumerateRecordingDevices
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::EnumerateRecordingDevices()
-{
-
- uint16_t nDevices(RecordingDevices());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#input devices: %u", nDevices);
-
- WAVEINCAPS caps;
- MMRESULT res;
-
- for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
- {
- res = waveInGetDevCaps(deviceID, &caps, sizeof(WAVEINCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
- if (caps.dwFormats & WAVE_FORMAT_48S16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
- }
- if (caps.dwFormats & WAVE_FORMAT_48M16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// TraceSupportFlags
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceSupportFlags(DWORD dwSupport) const
-{
- TCHAR buf[256];
-
- StringCchPrintf(buf, 128, TEXT("support flags : 0x%x "), dwSupport);
-
- if (dwSupport & WAVECAPS_PITCH)
- {
- // supports pitch control
- StringCchCat(buf, 256, TEXT("(PITCH)"));
- }
- if (dwSupport & WAVECAPS_PLAYBACKRATE)
- {
- // supports playback rate control
- StringCchCat(buf, 256, TEXT("(PLAYBACKRATE)"));
- }
- if (dwSupport & WAVECAPS_VOLUME)
- {
- // supports volume control
- StringCchCat(buf, 256, TEXT("(VOLUME)"));
- }
- if (dwSupport & WAVECAPS_LRVOLUME)
- {
- // supports separate left and right volume control
- StringCchCat(buf, 256, TEXT("(LRVOLUME)"));
- }
- if (dwSupport & WAVECAPS_SYNC)
- {
- // the driver is synchronous and will block while playing a buffer
- StringCchCat(buf, 256, TEXT("(SYNC)"));
- }
- if (dwSupport & WAVECAPS_SAMPLEACCURATE)
- {
- // returns sample-accurate position information
- StringCchCat(buf, 256, TEXT("(SAMPLEACCURATE)"));
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%S", buf);
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveInError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceWaveInError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveInGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%S", buf);
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveOutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceWaveOutError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveOutGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%S", buf);
-}
-
-// ----------------------------------------------------------------------------
-// PrepareStartPlayout
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PrepareStartPlayout()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- // A total of 30ms of data is immediately placed in the SC buffer
- //
- int8_t zeroVec[4*PLAY_BUF_SIZE_IN_SAMPLES]; // max allocation
- memset(zeroVec, 0, 4*PLAY_BUF_SIZE_IN_SAMPLES);
-
- {
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- }
-
- _playAcc = 0;
- _playWarning = 0;
- _playError = 0;
- _dc_diff_mean = 0;
- _dc_y_prev = 0;
- _dc_penalty_counter = 20;
- _dc_prevtime = 0;
- _dc_prevplay = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PrepareStartRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::PrepareStartRecording()
-{
-
- CriticalSectionScoped lock(&_critSect);
-
- if (_hWaveIn == NULL)
- {
- return -1;
- }
-
- _playAcc = 0;
- _recordedBytes = 0;
- _recPutBackDelay = REC_PUT_BACK_DELAY;
-
- MMRESULT res;
- MMTIME mmtime;
- mmtime.wType = TIME_SAMPLES;
-
- res = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition(TIME_SAMPLES) failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- _read_samples = mmtime.u.sample;
- _read_samples_old = _read_samples;
- _rec_samples_old = mmtime.u.sample;
- _wrapCounter = 0;
-
- for (int n = 0; n < N_BUFFERS_IN; n++)
- {
- const uint8_t nBytesPerSample = 2*_recChannels;
-
- // set up the input wave header
- _waveHeaderIn[n].lpData = reinterpret_cast<LPSTR>(&_recBuffer[n]);
- _waveHeaderIn[n].dwBufferLength = nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
- _waveHeaderIn[n].dwFlags = 0;
- _waveHeaderIn[n].dwBytesRecorded = 0;
- _waveHeaderIn[n].dwUser = 0;
-
- memset(_recBuffer[n], 0, nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES);
-
- // prepare a buffer for waveform-audio input
- res = waveInPrepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", n, res);
- TraceWaveInError(res);
- }
-
- // send an input buffer to the given waveform-audio input device
- res = waveInAddBuffer(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", n, res);
- TraceWaveInError(res);
- }
- }
-
- // start input on the given waveform-audio input device
- res = waveInStart(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStart() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetPlayoutBufferDelay
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::GetPlayoutBufferDelay(uint32_t& writtenSamples, uint32_t& playedSamples)
-{
- int i;
- int ms_Header;
- long playedDifference;
- int msecInPlayoutBuffer(0); // #milliseconds of audio in the playout buffer
-
- const uint16_t nSamplesPerMs = (uint16_t)(N_PLAY_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
-
- MMRESULT res;
- MMTIME mmtime;
-
- if (!_playing)
- {
- playedSamples = 0;
- return (0);
- }
-
- // Retrieve the current playback position.
- //
- mmtime.wType = TIME_SAMPLES; // number of waveform-audio samples
- res = waveOutGetPosition(_hWaveOut, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetPosition() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- writtenSamples = _writtenSamples; // #samples written to the playout buffer
- playedSamples = mmtime.u.sample; // current playout position in the playout buffer
-
- // derive remaining amount (in ms) of data in the playout buffer
- msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
-
- playedDifference = (long) (_playedSamplesOld - playedSamples);
-
- if (playedDifference > 64000)
- {
- // If the sound cards number-of-played-out-samples variable wraps around before
- // written_sampels wraps around this needs to be adjusted. This can happen on
- // sound cards that uses less than 32 bits to keep track of number of played out
- // sampels. To avoid being fooled by sound cards that sometimes produces false
- // output we compare old value minus the new value with a large value. This is
- // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
- // would trigger the wrap-around function if we didn't compare with a large value.
- // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
-
- i = 31;
- while((_playedSamplesOld <= (unsigned long)POW2(i)) && (i > 14)) {
- i--;
- }
-
- if((i < 31) && (i > 14)) {
- // Avoid adjusting when there is 32-bit wrap-around since that is
- // something neccessary.
- //
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "msecleft() => wrap around occured: %d bits used by sound card)", (i+1));
-
- _writtenSamples = _writtenSamples - POW2(i + 1);
- writtenSamples = _writtenSamples;
- msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
- }
- }
- else if ((_writtenSamplesOld > (unsigned long)POW2(31)) && (writtenSamples < 96000))
- {
- // Wrap around as expected after having used all 32 bits. (But we still
- // test if the wrap around happened earlier which it should not)
-
- i = 31;
- while (_writtenSamplesOld <= (unsigned long)POW2(i)) {
- i--;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured after having used all 32 bits)");
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
- msecInPlayoutBuffer = (int)((writtenSamples + POW2(i + 1) - playedSamples)/nSamplesPerMs);
-
- }
- else if ((writtenSamples < 96000) && (playedSamples > (unsigned long)POW2(31)))
- {
- // Wrap around has, as expected, happened for written_sampels before
- // playedSampels so we have to adjust for this until also playedSampels
- // has had wrap around.
-
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured: correction of output is done)");
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
- msecInPlayoutBuffer = (int)((writtenSamples + POW2(32) - playedSamples)/nSamplesPerMs);
- }
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
-
-
- // We use the following formaula to track that playout works as it should
- // y=playedSamples/48 - timeGetTime();
- // y represent the clock drift between system clock and sound card clock - should be fairly stable
- // When the exponential mean value of diff(y) goes away from zero something is wrong
- // The exponential formula will accept 1% clock drift but not more
- // The driver error means that we will play to little audio and have a high negative clock drift
- // We kick in our alternative method when the clock drift reaches 20%
-
- int diff,y;
- int unsigned time =0;
-
- // If we have other problems that causes playout glitches
- // we don't want to switch playout method.
- // Check if playout buffer is extremely low, or if we haven't been able to
- // exectue our code in more than 40 ms
-
- time = timeGetTime();
-
- if ((msecInPlayoutBuffer < 20) || (time - _dc_prevtime > 40))
- {
- _dc_penalty_counter = 100;
- }
-
- if ((playedSamples != 0))
- {
- y = playedSamples/48 - time;
- if ((_dc_y_prev != 0) && (_dc_penalty_counter == 0))
- {
- diff = y - _dc_y_prev;
- _dc_diff_mean = (990*_dc_diff_mean)/1000 + 10*diff;
- }
- _dc_y_prev = y;
- }
-
- if (_dc_penalty_counter)
- {
- _dc_penalty_counter--;
- }
-
- if (_dc_diff_mean < -200)
- {
- // Always reset the filter
- _dc_diff_mean = 0;
-
- // Problem is detected. Switch delay method and set min buffer to 80.
- // Reset the filter and keep monitoring the filter output.
- // If issue is detected a second time, increase min buffer to 100.
- // If that does not help, we must modify this scheme further.
-
- _useHeader++;
- if (_useHeader == 1)
- {
- _minPlayBufDelay = 80;
- _playWarning = 1; // only warn first time
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #1: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
- }
- else if (_useHeader == 2)
- {
- _minPlayBufDelay = 100; // add some more safety
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #2: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
- }
- else
- {
- // This state should not be entered... (HA)
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "further actions are required!");
- }
- if (_playWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout warning exists");
- }
- _playWarning = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kPlayoutWarning message posted: switching to alternative playout delay method");
- }
- _dc_prevtime = time;
- _dc_prevplay = playedSamples;
-
- // Try a very rough method of looking at how many buffers are still playing
- ms_Header = 0;
- for (i = 0; i < N_BUFFERS_OUT; i++) {
- if ((_waveHeaderOut[i].dwFlags & WHDR_INQUEUE)!=0) {
- ms_Header += 10;
- }
- }
-
- if ((ms_Header-50) > msecInPlayoutBuffer) {
- // Test for cases when GetPosition appears to be screwed up (currently just log....)
- TCHAR infoStr[300];
- if (_no_of_msecleft_warnings%20==0)
- {
- StringCchPrintf(infoStr, 300, TEXT("writtenSamples=%i, playedSamples=%i, msecInPlayoutBuffer=%i, ms_Header=%i"), writtenSamples, playedSamples, msecInPlayoutBuffer, ms_Header);
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "%S", infoStr);
- }
- _no_of_msecleft_warnings++;
- }
-
- // If this is true we have had a problem with the playout
- if (_useHeader > 0)
- {
- return (ms_Header);
- }
-
-
- if (ms_Header < msecInPlayoutBuffer)
- {
- if (_no_of_msecleft_warnings % 100 == 0)
- {
- TCHAR str[300];
- StringCchPrintf(str, 300, TEXT("_no_of_msecleft_warnings=%i, msecInPlayoutBuffer=%i ms_Header=%i (minBuffer=%i buffersize=%i writtenSamples=%i playedSamples=%i)"),
- _no_of_msecleft_warnings, msecInPlayoutBuffer, ms_Header, _minPlayBufDelay, _playBufDelay, writtenSamples, playedSamples);
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "%S", str);
- }
- _no_of_msecleft_warnings++;
- ms_Header -= 6; // Round off as we only have 10ms resolution + Header info is usually slightly delayed compared to GetPosition
-
- if (ms_Header < 0)
- ms_Header = 0;
-
- return (ms_Header);
- }
- else
- {
- return (msecInPlayoutBuffer);
- }
-}
-
-// ----------------------------------------------------------------------------
-// GetRecordingBufferDelay
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::GetRecordingBufferDelay(uint32_t& readSamples, uint32_t& recSamples)
-{
- long recDifference;
- MMTIME mmtime;
- MMRESULT mmr;
-
- const uint16_t nSamplesPerMs = (uint16_t)(N_REC_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
-
- // Retrieve the current input position of the given waveform-audio input device
- //
- mmtime.wType = TIME_SAMPLES;
- mmr = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != mmr)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition() failed (err=%d)", mmr);
- TraceWaveInError(mmr);
- }
-
- readSamples = _read_samples; // updated for each full fram in RecProc()
- recSamples = mmtime.u.sample; // remaining time in input queue (recorded but not read yet)
-
- recDifference = (long) (_rec_samples_old - recSamples);
-
- if( recDifference > 64000) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 1 (recDifference =%d)", recDifference);
- // If the sound cards number-of-recorded-samples variable wraps around before
- // read_sampels wraps around this needs to be adjusted. This can happen on
- // sound cards that uses less than 32 bits to keep track of number of played out
- // sampels. To avoid being fooled by sound cards that sometimes produces false
- // output we compare old value minus the new value with a large value. This is
- // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
- // would trigger the wrap-around function if we didn't compare with a large value.
- // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
- //
- int i = 31;
- while((_rec_samples_old <= (unsigned long)POW2(i)) && (i > 14))
- i--;
-
- if((i < 31) && (i > 14)) {
- // Avoid adjusting when there is 32-bit wrap-around since that is
- // somethying neccessary.
- //
- _read_samples = _read_samples - POW2(i + 1);
- readSamples = _read_samples;
- _wrapCounter++;
- } else {
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"AEC (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
- }
- }
-
- if((_wrapCounter>200)){
- // Do nothing, handled later
- }
- else if((_rec_samples_old > (unsigned long)POW2(31)) && (recSamples < 96000)) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 2 (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
- // Wrap around as expected after having used all 32 bits.
- _read_samples_old = readSamples;
- _rec_samples_old = recSamples;
- _wrapCounter++;
- return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
-
-
- } else if((recSamples < 96000) && (readSamples > (unsigned long)POW2(31))) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 3 (readSamples %d recSamples %d)",readSamples, recSamples);
- // Wrap around has, as expected, happened for rec_sampels before
- // readSampels so we have to adjust for this until also readSampels
- // has had wrap around.
- _read_samples_old = readSamples;
- _rec_samples_old = recSamples;
- _wrapCounter++;
- return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
- }
-
- _read_samples_old = _read_samples;
- _rec_samples_old = recSamples;
- int res=(((int)_rec_samples_old - (int)_read_samples_old)/nSamplesPerMs);
-
- if((res > 2000)||(res < 0)||(_wrapCounter>200)){
- // Reset everything
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"msec_read error (res %d wrapCounter %d)",res, _wrapCounter);
- MMTIME mmtime;
- mmtime.wType = TIME_SAMPLES;
-
- mmr=waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (mmr != MMSYSERR_NOERROR) {
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "waveInGetPosition failed (mmr=%d)", mmr);
- }
- _read_samples=mmtime.u.sample;
- _read_samples_old=_read_samples;
- _rec_samples_old=mmtime.u.sample;
-
- // Guess a decent value
- res = 20;
- }
-
- _wrapCounter = 0;
- return res;
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// ThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::ThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceWindowsWave*>(pThis)->ThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// ThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::ThreadProcess()
-{
- uint32_t time(0);
- uint32_t playDiff(0);
- uint32_t recDiff(0);
-
- LONGLONG playTime(0);
- LONGLONG recTime(0);
-
- switch (_timeEvent.Wait(1000))
- {
- case kEventSignaled:
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "EventWrapper::Wait() failed => restarting timer");
- _timeEvent.StopTimer();
- _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
- return true;
- case kEventTimeout:
- return true;
- }
-
- time = rtc::TimeMillis();
-
- if (_startPlay)
- {
- if (PrepareStartPlayout() == 0)
- {
- _prevTimerCheckTime = time;
- _prevPlayTime = time;
- _startPlay = false;
- _playing = true;
- _playStartEvent.Set();
- }
- }
-
- if (_startRec)
- {
- if (PrepareStartRecording() == 0)
- {
- _prevTimerCheckTime = time;
- _prevRecTime = time;
- _prevRecByteCheckTime = time;
- _startRec = false;
- _recording = true;
- _recStartEvent.Set();
- }
- }
-
- if (_playing)
- {
- playDiff = time - _prevPlayTime;
- }
-
- if (_recording)
- {
- recDiff = time - _prevRecTime;
- }
-
- if (_playing || _recording)
- {
- RestartTimerIfNeeded(time);
- }
-
- if (_playing &&
- (playDiff > (uint32_t)(_dTcheckPlayBufDelay - 1)) ||
- (playDiff < 0))
- {
- Lock();
- if (_playing)
- {
- if (PlayProc(playTime) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
- }
- _prevPlayTime = time;
- if (playTime != 0)
- _playAcc += playTime;
- }
- UnLock();
- }
-
- if (_playing && (playDiff > 12))
- {
- // It has been a long time since we were able to play out, try to
- // compensate by calling PlayProc again.
- //
- Lock();
- if (_playing)
- {
- if (PlayProc(playTime))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
- }
- _prevPlayTime = time;
- if (playTime != 0)
- _playAcc += playTime;
- }
- UnLock();
- }
-
- if (_recording &&
- (recDiff > REC_CHECK_TIME_PERIOD_MS) ||
- (recDiff < 0))
- {
- Lock();
- if (_recording)
- {
- int32_t nRecordedBytes(0);
- uint16_t maxIter(10);
-
- // Deliver all availiable recorded buffers and update the CPU load measurement.
- // We use a while loop here to compensate for the fact that the multi-media timer
- // can sometimed enter a "bad state" after hibernation where the resolution is
- // reduced from ~1ms to ~10-15 ms.
- //
- while ((nRecordedBytes = RecProc(recTime)) > 0)
- {
- maxIter--;
- _recordedBytes += nRecordedBytes;
- if (recTime && _perfFreq.QuadPart)
- {
- // Measure the average CPU load:
- // This is a simplified expression where an exponential filter is used:
- // _avgCPULoad = 0.99 * _avgCPULoad + 0.01 * newCPU,
- // newCPU = (recTime+playAcc)/f is time in seconds
- // newCPU / 0.01 is the fraction of a 10 ms period
- // The two 0.01 cancels each other.
- // NOTE - assumes 10ms audio buffers.
- //
- _avgCPULoad = (float)(_avgCPULoad*.99 + (recTime+_playAcc)/(double)(_perfFreq.QuadPart));
- _playAcc = 0;
- }
- if (maxIter == 0)
- {
- // If we get this message ofte, our compensation scheme is not sufficient.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "failed to compensate for reduced MM-timer resolution");
- }
- }
-
- if (nRecordedBytes == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "RecProc() failed");
- }
-
- _prevRecTime = time;
-
- // Monitor the recording process and generate error/warning callbacks if needed
- MonitorRecording(time);
- }
- UnLock();
- }
-
- if (!_recording)
- {
- _prevRecByteCheckTime = time;
- _avgCPULoad = 0;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// RecProc
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
-{
- MMRESULT res;
- uint32_t bufCount(0);
- uint32_t nBytesRecorded(0);
-
- consumedTime = 0;
-
- // count modulo N_BUFFERS_IN (0,1,2,...,(N_BUFFERS_IN-1),0,1,2,..)
- if (_recBufCount == N_BUFFERS_IN)
- {
- _recBufCount = 0;
- }
-
- bufCount = _recBufCount;
-
- // take mono/stereo mode into account when deriving size of a full buffer
- const uint16_t bytesPerSample = 2*_recChannels;
- const uint32_t fullBufferSizeInBytes = bytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
-
- // read number of recorded bytes for the given input-buffer
- nBytesRecorded = _waveHeaderIn[bufCount].dwBytesRecorded;
-
- if (nBytesRecorded == fullBufferSizeInBytes ||
- (nBytesRecorded > 0))
- {
- int32_t msecOnPlaySide;
- int32_t msecOnRecordSide;
- uint32_t writtenSamples;
- uint32_t playedSamples;
- uint32_t readSamples, recSamples;
- bool send = true;
-
- uint32_t nSamplesRecorded = (nBytesRecorded/bytesPerSample); // divide by 2 or 4 depending on mono or stereo
-
- if (nBytesRecorded == fullBufferSizeInBytes)
- {
- _timesdwBytes = 0;
- }
- else
- {
- // Test if it is stuck on this buffer
- _timesdwBytes++;
- if (_timesdwBytes < 5)
- {
- // keep trying
- return (0);
- }
- else
- {
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id,"nBytesRecorded=%d => don't use", nBytesRecorded);
- _timesdwBytes = 0;
- send = false;
- }
- }
-
- // store the recorded buffer (no action will be taken if the #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer(_waveHeaderIn[bufCount].lpData, nSamplesRecorded);
-
- // update #samples read
- _read_samples += nSamplesRecorded;
-
- // Check how large the playout and recording buffers are on the sound card.
- // This info is needed by the AEC.
- //
- msecOnPlaySide = GetPlayoutBufferDelay(writtenSamples, playedSamples);
- msecOnRecordSide = GetRecordingBufferDelay(readSamples, recSamples);
-
- // If we use the alternative playout delay method, skip the clock drift compensation
- // since it will be an unreliable estimate and might degrade AEC performance.
- int32_t drift = (_useHeader > 0) ? 0 : GetClockDrift(playedSamples, recSamples);
-
- _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, drift);
-
- _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
- // Store the play and rec delay values for video synchronization
- _sndCardPlayDelay = msecOnPlaySide;
- _sndCardRecDelay = msecOnRecordSide;
-
- LARGE_INTEGER t1={0},t2={0};
-
- if (send)
- {
- QueryPerformanceCounter(&t1);
-
- // deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
-
- QueryPerformanceCounter(&t2);
-
- if (InputSanityCheckAfterUnlockedPeriod() == -1)
- {
- // assert(false);
- return -1;
- }
- }
-
- if (_AGC)
- {
- uint32_t newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a change is needed.
- WEBRTC_TRACE(kTraceStream, kTraceUtility, _id,"AGC change of volume: => new=%u", newMicLevel);
-
- // We store this outside of the audio buffer to avoid
- // having it overwritten by the getter thread.
- _newMicLevel = newMicLevel;
- SetEvent(_hSetCaptureVolumeEvent);
- }
- }
-
- // return utilized buffer to queue after specified delay (default is 4)
- if (_recDelayCount > (_recPutBackDelay-1))
- {
- // deley buffer counter to compensate for "put-back-delay"
- bufCount = (bufCount + N_BUFFERS_IN - _recPutBackDelay) % N_BUFFERS_IN;
-
- // reset counter so we can make new detection
- _waveHeaderIn[bufCount].dwBytesRecorded = 0;
-
- // return the utilized wave-header after certain delay (given by _recPutBackDelay)
- res = waveInUnprepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- }
-
- // ensure that the utilized header can be used again
- res = waveInPrepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- return -1;
- }
-
- // add the utilized buffer to the queue again
- res = waveInAddBuffer(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- if (_recPutBackDelay < 50)
- {
- _recPutBackDelay++;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "_recPutBackDelay increased to %d", _recPutBackDelay);
- }
- else
- {
- if (_recError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
- }
- _recError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: _recPutBackDelay=%u", _recPutBackDelay);
- }
- }
- } // if (_recDelayCount > (_recPutBackDelay-1))
-
- if (_recDelayCount < (_recPutBackDelay+1))
- {
- _recDelayCount++;
- }
-
- // increase main buffer count since one complete buffer has now been delivered
- _recBufCount++;
-
- if (send) {
- // Calculate processing time
- consumedTime = (int)(t2.QuadPart-t1.QuadPart);
- // handle wraps, time should not be higher than a second
- if ((consumedTime > _perfFreq.QuadPart) || (consumedTime < 0))
- consumedTime = 0;
- }
-
- } // if ((nBytesRecorded == fullBufferSizeInBytes))
-
- return nBytesRecorded;
-}
-
-// ----------------------------------------------------------------------------
-// PlayProc
-// ----------------------------------------------------------------------------
-
-int AudioDeviceWindowsWave::PlayProc(LONGLONG& consumedTime)
-{
- int32_t remTimeMS(0);
- int8_t playBuffer[4*PLAY_BUF_SIZE_IN_SAMPLES];
- uint32_t writtenSamples(0);
- uint32_t playedSamples(0);
-
- LARGE_INTEGER t1;
- LARGE_INTEGER t2;
-
- consumedTime = 0;
- _waitCounter++;
-
- // Get number of ms of sound that remains in the sound card buffer for playback.
- //
- remTimeMS = GetPlayoutBufferDelay(writtenSamples, playedSamples);
-
- // The threshold can be adaptive or fixed. The adaptive scheme is updated
- // also for fixed mode but the updated threshold is not utilized.
- //
- const uint16_t thresholdMS =
- (_playBufType == AudioDeviceModule::kAdaptiveBufferSize) ? _playBufDelay : _playBufDelayFixed;
-
- if (remTimeMS < thresholdMS + 9)
- {
- _dTcheckPlayBufDelay = 5;
-
- if (remTimeMS == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id, "playout buffer is empty => we must adapt...");
- if (_waitCounter > 30)
- {
- _erZeroCounter++;
- if (_erZeroCounter == 2)
- {
- _playBufDelay += 15;
- _minPlayBufDelay += 20;
- _waitCounter = 50;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0,erZero=2): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- else if (_erZeroCounter == 3)
- {
- _erZeroCounter = 0;
- _playBufDelay += 30;
- _minPlayBufDelay += 25;
- _waitCounter = 0;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=3): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- else
- {
- _minPlayBufDelay += 10;
- _playBufDelay += 15;
- _waitCounter = 50;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=1): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- }
- }
- else if (remTimeMS < _minPlayBufDelay)
- {
- // If there is less than 25 ms of audio in the play out buffer
- // increase the buffersize limit value. _waitCounter prevents
- // _playBufDelay to be increased every time this function is called.
-
- if (_waitCounter > 30)
- {
- _playBufDelay += 10;
- if (_intro == 0)
- _waitCounter = 0;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is increased: playBufDelay=%u", _playBufDelay);
- }
- }
- else if (remTimeMS < thresholdMS - 9)
- {
- _erZeroCounter = 0;
- }
- else
- {
- _erZeroCounter = 0;
- _dTcheckPlayBufDelay = 10;
- }
-
- QueryPerformanceCounter(&t1); // measure time: START
-
- // Ask for new PCM data to be played out using the AudioDeviceBuffer.
- // Ensure that this callback is executed without taking the audio-thread lock.
- //
- UnLock();
- uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES);
- Lock();
-
- if (OutputSanityCheckAfterUnlockedPeriod() == -1)
- {
- // assert(false);
- return -1;
- }
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
- }
-
- QueryPerformanceCounter(&t2); // measure time: STOP
- consumedTime = (int)(t2.QuadPart - t1.QuadPart);
-
- Write(playBuffer, PLAY_BUF_SIZE_IN_SAMPLES);
-
- } // if (er < thresholdMS + 9)
- else if (thresholdMS + 9 < remTimeMS )
- {
- _erZeroCounter = 0;
- _dTcheckPlayBufDelay = 2; // check buffer more often
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Need to check playout buffer more often (dT=%u, remTimeMS=%u)", _dTcheckPlayBufDelay, remTimeMS);
- }
-
- // If the buffersize has been stable for 20 seconds try to decrease the buffer size
- if (_waitCounter > 2000)
- {
- _intro = 0;
- _playBufDelay--;
- _waitCounter = 1990;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is decreased: playBufDelay=%u", _playBufDelay);
- }
-
- // Limit the minimum sound card (playback) delay to adaptive minimum delay
- if (_playBufDelay < _minPlayBufDelay)
- {
- _playBufDelay = _minPlayBufDelay;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %u", _minPlayBufDelay);
- }
-
- // Limit the maximum sound card (playback) delay to 150 ms
- if (_playBufDelay > 150)
- {
- _playBufDelay = 150;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %d", _playBufDelay);
- }
-
- // Upper limit of the minimum sound card (playback) delay to 65 ms.
- // Deactivated during "useHeader mode" (_useHeader > 0).
- if (_minPlayBufDelay > _MAX_minBuffer &&
- (_useHeader == 0))
- {
- _minPlayBufDelay = _MAX_minBuffer;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Minimum playout threshold is limited to %d", _MAX_minBuffer);
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// Write
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::Write(int8_t* data, uint16_t nSamples)
-{
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- MMRESULT res;
-
- const uint16_t bufCount(_playBufCount);
-
- // Place data in the memory associated with _waveHeaderOut[bufCount]
- //
- const int16_t nBytes = (2*_playChannels)*nSamples;
- memcpy(&_playBuffer[bufCount][0], &data[0], nBytes);
-
- // Send a data block to the given waveform-audio output device.
- //
- // When the buffer is finished, the WHDR_DONE bit is set in the dwFlags
- // member of the WAVEHDR structure. The buffer must be prepared with the
- // waveOutPrepareHeader function before it is passed to waveOutWrite.
- // Unless the device is paused by calling the waveOutPause function,
- // playback begins when the first data block is sent to the device.
- //
- res = waveOutWrite(_hWaveOut, &_waveHeaderOut[bufCount], sizeof(_waveHeaderOut[bufCount]));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutWrite(%d) failed (err=%d)", bufCount, res);
- TraceWaveOutError(res);
-
- _writeErrors++;
- if (_writeErrors > 10)
- {
- if (_playError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout error exists");
- }
- _playError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: _writeErrors=%u", _writeErrors);
- }
-
- return -1;
- }
-
- _playBufCount = (_playBufCount+1) % N_BUFFERS_OUT; // increase buffer counter modulo size of total buffer
- _writtenSamples += nSamples; // each sample is 2 or 4 bytes
- _writeErrors = 0;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetClockDrift
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::GetClockDrift(const uint32_t plSamp, const uint32_t rcSamp)
-{
- int drift = 0;
- unsigned int plSampDiff = 0, rcSampDiff = 0;
-
- if (plSamp >= _plSampOld)
- {
- plSampDiff = plSamp - _plSampOld;
- }
- else
- {
- // Wrap
- int i = 31;
- while(_plSampOld <= (unsigned int)POW2(i))
- {
- i--;
- }
-
- // Add the amount remaining prior to wrapping
- plSampDiff = plSamp + POW2(i + 1) - _plSampOld;
- }
-
- if (rcSamp >= _rcSampOld)
- {
- rcSampDiff = rcSamp - _rcSampOld;
- }
- else
- { // Wrap
- int i = 31;
- while(_rcSampOld <= (unsigned int)POW2(i))
- {
- i--;
- }
-
- rcSampDiff = rcSamp + POW2(i + 1) - _rcSampOld;
- }
-
- drift = plSampDiff - rcSampDiff;
-
- _plSampOld = plSamp;
- _rcSampOld = rcSamp;
-
- return drift;
-}
-
-// ----------------------------------------------------------------------------
-// MonitorRecording
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::MonitorRecording(const uint32_t time)
-{
- const uint16_t bytesPerSample = 2*_recChannels;
- const uint32_t nRecordedSamples = _recordedBytes/bytesPerSample;
-
- if (nRecordedSamples > 5*N_REC_SAMPLES_PER_SEC)
- {
- // 5 seconds of audio has been recorded...
- if ((time - _prevRecByteCheckTime) > 5700)
- {
- // ...and it was more than 5.7 seconds since we last did this check <=>
- // we have not been able to record 5 seconds of audio in 5.7 seconds,
- // hence a problem should be reported.
- // This problem can be related to USB overload.
- //
- if (_recWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording warning exists");
- }
- _recWarning = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kRecordingWarning message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
- }
-
- _recordedBytes = 0; // restart "check again when 5 seconds are recorded"
- _prevRecByteCheckTime = time; // reset timer to measure time for recording of 5 seconds
- }
-
- if ((time - _prevRecByteCheckTime) > 8000)
- {
- // It has been more than 8 seconds since we able to confirm that 5 seconds of
- // audio was recorded, hence we have not been able to record 5 seconds in
- // 8 seconds => the complete recording process is most likely dead.
- //
- if (_recError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
- }
- _recError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
-
- _prevRecByteCheckTime = time;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MonitorRecording
-//
-// Restart timer if needed (they seem to be messed up after a hibernate).
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsWave::RestartTimerIfNeeded(const uint32_t time)
-{
- const uint32_t diffMS = time - _prevTimerCheckTime;
- _prevTimerCheckTime = time;
-
- if (diffMS > 7)
- {
- // one timer-issue detected...
- _timerFaults++;
- if (_timerFaults > 5 && _timerRestartAttempts < 2)
- {
- // Reinitialize timer event if event fails to execute at least every 5ms.
- // On some machines it helps and the timer starts working as it should again;
- // however, not all machines (we have seen issues on e.g. IBM T60).
- // Therefore, the scheme below ensures that we do max 2 attempts to restart the timer.
- // For the cases where restart does not do the trick, we compensate for the reduced
- // resolution on both the recording and playout sides.
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, " timer issue detected => timer is restarted");
- _timeEvent.StopTimer();
- _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
- // make sure timer gets time to start up and we don't kill/start timer serveral times over and over again
- _timerFaults = -20;
- _timerRestartAttempts++;
- }
- }
- else
- {
- // restart timer-check scheme since we are OK
- _timerFaults = 0;
- _timerRestartAttempts = 0;
- }
-
- return 0;
-}
-
-
-bool AudioDeviceWindowsWave::KeyPressed() const{
-
- int key_down = 0;
- for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
- short res = GetAsyncKeyState(key);
- key_down |= res & 0x1; // Get the LSB
- }
- return (key_down > 0);
-}
-} // namespace webrtc
diff --git a/modules/audio_device/win/audio_device_wave_win.h b/modules/audio_device/win/audio_device_wave_win.h
deleted file mode 100644
index 6bb7c17..0000000
--- a/modules/audio_device/win/audio_device_wave_win.h
+++ /dev/null
@@ -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
diff --git a/voice_engine/voe_hardware_impl.cc b/voice_engine/voe_hardware_impl.cc
index 1748fc2..db47c73 100644
--- a/voice_engine/voe_hardware_impl.cc
+++ b/voice_engine/voe_hardware_impl.cc
@@ -57,9 +57,6 @@
case kAudioWindowsCore:
wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
break;
- case kAudioWindowsWave:
- wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
- break;
case kAudioLinuxAlsa:
wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
break;
@@ -100,9 +97,6 @@
case AudioDeviceModule::kWindowsCoreAudio:
audioLayer = kAudioWindowsCore;
break;
- case AudioDeviceModule::kWindowsWaveAudio:
- audioLayer = kAudioWindowsWave;
- break;
case AudioDeviceModule::kLinuxAlsaAudio:
audioLayer = kAudioLinuxAlsa;
break;
