UMA log for audio_device Init and Start(Playout|Recording). Make Init return a more specific error code, if possible.
BUG=webrtc:5761
R=asapersson@webrtc.org, henrika@webrtc.org
Review URL: https://codereview.webrtc.org/2103863004 .
Cr-Commit-Position: refs/heads/master@{#13361}
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 92061a2..35a99ea 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -27,6 +27,7 @@
source_set("audio_device") {
deps = [
"../..:webrtc_common",
+ "../../base:rtc_base",
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
diff --git a/webrtc/modules/audio_device/android/audio_device_template.h b/webrtc/modules/audio_device/android/audio_device_template.h
index fa7b15e..46d97f6 100644
--- a/webrtc/modules/audio_device/android/audio_device_template.h
+++ b/webrtc/modules/audio_device/android/audio_device_template.h
@@ -53,23 +53,24 @@
return 0;
}
- int32_t Init() override {
+ InitStatus Init() override {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!initialized_);
- if (!audio_manager_->Init())
- return -1;
+ if (!audio_manager_->Init()) {
+ return InitStatus::OTHER_ERROR;
+ }
if (output_.Init() != 0) {
audio_manager_->Close();
- return -1;
+ return InitStatus::PLAYOUT_ERROR;
}
if (input_.Init() != 0) {
output_.Terminate();
audio_manager_->Close();
- return -1;
+ return InitStatus::RECORDING_ERROR;
}
initialized_ = true;
- return 0;
+ return InitStatus::OK;
}
int32_t Terminate() override {
diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi
index a7c5789..0ef1da1 100644
--- a/webrtc/modules/audio_device/audio_device.gypi
+++ b/webrtc/modules/audio_device/audio_device.gypi
@@ -89,6 +89,9 @@
'audio_device_impl.cc',
'audio_device_impl.h',
],
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
+ ],
'conditions': [
['OS=="android"', {
'sources': [
diff --git a/webrtc/modules/audio_device/audio_device_generic.h b/webrtc/modules/audio_device/audio_device_generic.h
index c76ea52..326d933 100644
--- a/webrtc/modules/audio_device/audio_device_generic.h
+++ b/webrtc/modules/audio_device/audio_device_generic.h
@@ -18,12 +18,22 @@
class AudioDeviceGeneric {
public:
+ // For use with UMA logging. Must be kept in sync with histograms.xml in
+ // Chrome, located at
+ // https://cs.chromium.org/chromium/src/tools/metrics/histograms/histograms.xml
+ enum class InitStatus {
+ OK = 0,
+ PLAYOUT_ERROR = 1,
+ RECORDING_ERROR = 2,
+ OTHER_ERROR = 3,
+ NUM_STATUSES = 4
+ };
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const = 0;
// Main initializaton and termination
- virtual int32_t Init() = 0;
+ virtual InitStatus Init() = 0;
virtual int32_t Terminate() = 0;
virtual bool Initialized() const = 0;
diff --git a/webrtc/modules/audio_device/audio_device_impl.cc b/webrtc/modules/audio_device/audio_device_impl.cc
index b2a33ef..7c8840a 100644
--- a/webrtc/modules/audio_device/audio_device_impl.cc
+++ b/webrtc/modules/audio_device/audio_device_impl.cc
@@ -8,12 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
+#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/audio_device_impl.h"
+#include "webrtc/system_wrappers/include/metrics.h"
#include <assert.h>
#include <string.h>
@@ -266,7 +269,7 @@
// create *Linux PulseAudio* implementation
AudioDeviceLinuxPulse* pulseDevice = new AudioDeviceLinuxPulse(Id());
- if (pulseDevice->Init() != -1) {
+ if (pulseDevice->Init() == AudioDeviceGeneric::InitStatus::OK) {
ptrAudioDevice = pulseDevice;
LOG(INFO) << "Linux PulseAudio APIs will be utilized";
} else {
@@ -481,11 +484,14 @@
LOG(INFO) << __FUNCTION__;
if (_initialized)
return 0;
+ RTC_CHECK(_ptrAudioDevice);
- if (!_ptrAudioDevice)
- return -1;
-
- if (_ptrAudioDevice->Init() == -1) {
+ AudioDeviceGeneric::InitStatus status = _ptrAudioDevice->Init();
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.Audio.InitializationResult", static_cast<int>(status),
+ static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
+ if (status != AudioDeviceGeneric::InitStatus::OK) {
+ LOG(LS_ERROR) << "Audio device initialization failed.";
return -1;
}
@@ -1434,7 +1440,11 @@
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
_audioDeviceBuffer.InitPlayout();
- return (_ptrAudioDevice->InitPlayout());
+ int32_t result = _ptrAudioDevice->InitPlayout();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
@@ -1445,7 +1455,11 @@
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
_audioDeviceBuffer.InitRecording();
- return (_ptrAudioDevice->InitRecording());
+ int32_t result = _ptrAudioDevice->InitRecording();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
@@ -1475,7 +1489,11 @@
int32_t AudioDeviceModuleImpl::StartPlayout() {
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
- return (_ptrAudioDevice->StartPlayout());
+ int32_t result = _ptrAudioDevice->StartPlayout();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
@@ -1485,7 +1503,11 @@
int32_t AudioDeviceModuleImpl::StopPlayout() {
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
- return (_ptrAudioDevice->StopPlayout());
+ int32_t result = _ptrAudioDevice->StopPlayout();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
@@ -1505,7 +1527,11 @@
int32_t AudioDeviceModuleImpl::StartRecording() {
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
- return (_ptrAudioDevice->StartRecording());
+ int32_t result = _ptrAudioDevice->StartRecording();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
// StopRecording
@@ -1514,7 +1540,11 @@
int32_t AudioDeviceModuleImpl::StopRecording() {
LOG(INFO) << __FUNCTION__;
CHECK_INITIALIZED();
- return (_ptrAudioDevice->StopRecording());
+ int32_t result = _ptrAudioDevice->StopRecording();
+ LOG(INFO) << "output: " << result;
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
+ static_cast<int>(result == 0));
+ return result;
}
// ----------------------------------------------------------------------------
diff --git a/webrtc/modules/audio_device/dummy/audio_device_dummy.cc b/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
index 92199f6..4961937 100644
--- a/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
+++ b/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
@@ -17,7 +17,9 @@
return -1;
}
-int32_t AudioDeviceDummy::Init() { return 0; }
+AudioDeviceGeneric::InitStatus AudioDeviceDummy::Init() {
+ return InitStatus::OK;
+}
int32_t AudioDeviceDummy::Terminate() { return 0; }
diff --git a/webrtc/modules/audio_device/dummy/audio_device_dummy.h b/webrtc/modules/audio_device/dummy/audio_device_dummy.h
index 75c599e..72a0eb1 100644
--- a/webrtc/modules/audio_device/dummy/audio_device_dummy.h
+++ b/webrtc/modules/audio_device/dummy/audio_device_dummy.h
@@ -27,7 +27,7 @@
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
- int32_t Init() override;
+ InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override;
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
index 35e77a9..4ecc94f 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -55,7 +55,9 @@
return -1;
}
-int32_t FileAudioDevice::Init() { return 0; }
+AudioDeviceGeneric::InitStatus FileAudioDevice::Init() {
+ return InitStatus::OK;
+}
int32_t FileAudioDevice::Terminate() { return 0; }
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h
index be6fa3d..ae4737c 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -48,7 +48,7 @@
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
- int32_t Init() override;
+ InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override;
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.h b/webrtc/modules/audio_device/ios/audio_device_ios.h
index e6b72d6..efe2558 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -49,7 +49,7 @@
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
- int32_t Init() override;
+ InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override { return initialized_; }
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index 0a8da0f..3a745a4b 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -121,11 +121,11 @@
audio_device_buffer_ = audioBuffer;
}
-int32_t AudioDeviceIOS::Init() {
+AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
LOGI() << "Init";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (initialized_) {
- return 0;
+ return InitStatus::OK;
}
#if !defined(NDEBUG)
LogDeviceInfo();
@@ -146,7 +146,7 @@
// to guarantee mono on the "input side" of the audio unit.
UpdateAudioDeviceBuffer();
initialized_ = true;
- return 0;
+ return InitStatus::OK;
}
int32_t AudioDeviceIOS::Terminate() {
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index bdbccde..f7315de 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -10,6 +10,7 @@
#include <assert.h>
+#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/linux/audio_device_alsa_linux.h"
@@ -155,32 +156,25 @@
return 0;
}
-int32_t AudioDeviceLinuxALSA::Init()
-{
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
+ CriticalSectionScoped lock(&_critSect);
- CriticalSectionScoped lock(&_critSect);
+ // Load libasound
+ if (!AlsaSymbolTable.Load()) {
+ // Alsa is not installed on this system
+ LOG(LS_ERROR) << "failed to load symbol table";
+ return InitStatus::OTHER_ERROR;
+ }
- // Load libasound
- if (!AlsaSymbolTable.Load())
- {
- // Alsa is not installed on
- // this system
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to load symbol table");
- return -1;
- }
-
- if (_initialized)
- {
- return 0;
- }
+ if (_initialized) {
+ return InitStatus::OK;
+ }
#if defined(USE_X11)
//Get X display handle for typing detection
_XDisplay = XOpenDisplay(NULL);
- if (!_XDisplay)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to open X display, typing detection will not work");
+ if (!_XDisplay) {
+ LOG(LS_WARNING)
+ << "failed to open X display, typing detection will not work";
}
#endif
_playWarning = 0;
@@ -190,7 +184,7 @@
_initialized = true;
- return 0;
+ return InitStatus::OK;
}
int32_t AudioDeviceLinuxALSA::Terminate()
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
index 340e963..aba62e8 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -41,7 +41,7 @@
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
- int32_t Init() override;
+ InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override;
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
index 42c3ea8..e408f22 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -11,10 +11,9 @@
#include <assert.h>
#include "webrtc/base/checks.h"
-
+#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/linux/audio_device_pulse_linux.h"
-
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -163,58 +162,49 @@
return 0;
}
-int32_t AudioDeviceLinuxPulse::Init()
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_initialized)
- {
- return 0;
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_initialized) {
+ return InitStatus::OK;
+ }
+
+ // Initialize PulseAudio
+ if (InitPulseAudio() < 0) {
+ LOG(LS_ERROR) << "failed to initialize PulseAudio";
+ if (TerminatePulseAudio() < 0) {
+ LOG(LS_ERROR) << "failed to terminate PulseAudio";
}
+ return InitStatus::OTHER_ERROR;
+ }
- // Initialize PulseAudio
- if (InitPulseAudio() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize PulseAudio");
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
- if (TerminatePulseAudio() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to terminate PulseAudio");
- }
+ // Get X display handle for typing detection
+ _XDisplay = XOpenDisplay(NULL);
+ if (!_XDisplay) {
+ LOG(LS_WARNING)
+ << "failed to open X display, typing detection will not work";
+ }
- return -1;
- }
+ // RECORDING
+ _ptrThreadRec.reset(new rtc::PlatformThread(
+ RecThreadFunc, this, "webrtc_audio_module_rec_thread"));
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
+ _ptrThreadRec->Start();
+ _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
- //Get X display handle for typing detection
- _XDisplay = XOpenDisplay(NULL);
- if (!_XDisplay)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to open X display, typing detection will not work");
- }
+ // PLAYOUT
+ _ptrThreadPlay.reset(new rtc::PlatformThread(
+ PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+ _ptrThreadPlay->Start();
+ _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
- // RECORDING
- _ptrThreadRec.reset(new rtc::PlatformThread(
- RecThreadFunc, this, "webrtc_audio_module_rec_thread"));
+ _initialized = true;
- _ptrThreadRec->Start();
- _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
-
- // PLAYOUT
- _ptrThreadPlay.reset(new rtc::PlatformThread(
- PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
- _ptrThreadPlay->Start();
- _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
-
- _initialized = true;
-
- return 0;
+ return InitStatus::OK;
}
int32_t AudioDeviceLinuxPulse::Terminate()
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
index a0a15e5..3aa1698 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -103,7 +103,7 @@
AudioDeviceModule::AudioLayer& audioLayer) const override;
// Main initializaton and termination
- int32_t Init() override;
+ InitStatus Init() override;
int32_t Terminate() override;
bool Initialized() const override;
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.cc b/webrtc/modules/audio_device/mac/audio_device_mac.cc
index 0f33d11..70236b8 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.cc
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.cc
@@ -10,6 +10,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/mac/audio_device_mac.h"
@@ -221,11 +222,11 @@
return 0;
}
-int32_t AudioDeviceMac::Init() {
+AudioDeviceGeneric::InitStatus AudioDeviceMac::Init() {
CriticalSectionScoped lock(&_critSect);
if (_initialized) {
- return 0;
+ return InitStatus::OK;
}
OSStatus err = noErr;
@@ -250,7 +251,7 @@
if (bufSize == -1) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" PaUtil_InitializeRingBuffer() error");
- return -1;
+ return InitStatus::PLAYOUT_ERROR;
}
}
@@ -272,7 +273,7 @@
if (bufSize == -1) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" PaUtil_InitializeRingBuffer() error");
- return -1;
+ return InitStatus::RECORDING_ERROR;
}
}
@@ -282,7 +283,7 @@
if (kernErr != KERN_SUCCESS) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" semaphore_create() error: %d", kernErr);
- return -1;
+ return InitStatus::OTHER_ERROR;
}
kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
@@ -290,7 +291,7 @@
if (kernErr != KERN_SUCCESS) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" semaphore_create() error: %d", kernErr);
- return -1;
+ return InitStatus::OTHER_ERROR;
}
// Setting RunLoop to NULL here instructs HAL to manage its own thread for
@@ -302,8 +303,13 @@
kAudioObjectPropertyElementMaster};
CFRunLoopRef runLoop = NULL;
UInt32 size = sizeof(CFRunLoopRef);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
- kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop));
+ int aoerr = AudioObjectSetPropertyData(
+ kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop);
+ if (aoerr != noErr) {
+ LOG(LS_ERROR) << "Error in AudioObjectSetPropertyData: "
+ << (const char*)&aoerr;
+ return InitStatus::OTHER_ERROR;
+ }
// Listen for any device changes.
propertyAddress.mSelector = kAudioHardwarePropertyDevices;
@@ -338,7 +344,7 @@
_initialized = true;
- return 0;
+ return InitStatus::OK;
}
int32_t AudioDeviceMac::Terminate() {
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.h b/webrtc/modules/audio_device/mac/audio_device_mac.h
index 102c67e..ccde6f0 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.h
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.h
@@ -68,7 +68,7 @@
AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
- virtual int32_t Init();
+ virtual InitStatus Init();
virtual int32_t Terminate();
virtual bool Initialized() const;
diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.cc b/webrtc/modules/audio_device/win/audio_device_core_win.cc
index 811a04d..0c2d890 100644
--- a/webrtc/modules/audio_device/win/audio_device_core_win.cc
+++ b/webrtc/modules/audio_device/win/audio_device_core_win.cc
@@ -35,6 +35,7 @@
#include <strsafe.h>
#include <uuids.h>
+#include "webrtc/base/logging.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -333,7 +334,9 @@
int temp_ok(0);
bool available(false);
- ok |= p->Init();
+ if (p->Init() != InitStatus::OK) {
+ ok |= -1;
+ }
int16_t numDevsRec = p->RecordingDevices();
for (uint16_t i = 0; i < numDevsRec; i++)
@@ -675,31 +678,27 @@
// Init
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::Init()
-{
+AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
+ CriticalSectionScoped lock(&_critSect);
- CriticalSectionScoped lock(&_critSect);
+ if (_initialized) {
+ return InitStatus::OK;
+ }
- if (_initialized)
- {
- return 0;
- }
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
+ // Enumerate all audio rendering and capturing endpoint devices.
+ // Note that, some of these will not be able to select by the user.
+ // The complete collection is for internal use only.
+ _EnumerateEndpointDevicesAll(eRender);
+ _EnumerateEndpointDevicesAll(eCapture);
- // Enumerate all audio rendering and capturing endpoint devices.
- // Note that, some of these will not be able to select by the user.
- // The complete collection is for internal use only.
- //
- _EnumerateEndpointDevicesAll(eRender);
- _EnumerateEndpointDevicesAll(eCapture);
+ _initialized = true;
- _initialized = true;
-
- return 0;
+ return InitStatus::OK;
}
// ----------------------------------------------------------------------------
diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.h b/webrtc/modules/audio_device/win/audio_device_core_win.h
index 5c94cfd..b86606a 100644
--- a/webrtc/modules/audio_device/win/audio_device_core_win.h
+++ b/webrtc/modules/audio_device/win/audio_device_core_win.h
@@ -91,7 +91,7 @@
virtual int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
- virtual int32_t Init();
+ virtual InitStatus Init();
virtual int32_t Terminate();
virtual bool Initialized() const;
diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/webrtc/modules/audio_device/win/audio_device_wave_win.cc
index c149708..7874575 100644
--- a/webrtc/modules/audio_device/win/audio_device_wave_win.cc
+++ b/webrtc/modules/audio_device/win/audio_device_wave_win.cc
@@ -8,6 +8,7 @@
* 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"
@@ -196,79 +197,69 @@
// Init
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsWave::Init()
-{
+AudioDeviceGeneric::InitStatus AudioDeviceWindowsWave::Init() {
+ CriticalSectionScoped lock(&_critSect);
- CriticalSectionScoped lock(&_critSect);
+ if (_initialized) {
+ return InitStatus::OK;
+ }
- if (_initialized)
- {
- return 0;
- }
+ const uint32_t nowTime(rtc::TimeMillis());
- const uint32_t nowTime(rtc::TimeMillis());
+ _recordedBytes = 0;
+ _prevRecByteCheckTime = nowTime;
+ _prevRecTime = nowTime;
+ _prevPlayTime = nowTime;
+ _prevTimerCheckTime = nowTime;
- _recordedBytes = 0;
- _prevRecByteCheckTime = nowTime;
- _prevRecTime = nowTime;
- _prevPlayTime = nowTime;
- _prevTimerCheckTime = nowTime;
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
+ _mixerManager.EnumerateAll();
- _mixerManager.EnumerateAll();
+ if (_ptrThread) {
+ // thread is already created and active
+ return InitStatus::OK;
+ }
- if (_ptrThread)
- {
- // thread is already created and active
- return 0;
- }
+ const char* threadName = "webrtc_audio_module_thread";
+ _ptrThread.reset(new rtc::PlatformThread(ThreadFunc, this, threadName));
+ _ptrThread->Start();
+ _ptrThread->SetPriority(rtc::kRealtimePriority);
- 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);
- const bool periodic(true);
- if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to start the timer event");
- _ptrThread->Stop();
- _ptrThread.reset();
- return -1;
- }
- 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;
+ }
- _hGetCaptureVolumeThread =
- CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
- if (_hGetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume getter thread");
- return -1;
- }
+ SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
- 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;
+ }
- _hSetCaptureVolumeThread =
- CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
- if (_hSetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume setter thread");
- return -1;
- }
+ SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
- SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
+ _initialized = true;
- _initialized = true;
-
- return 0;
+ return InitStatus::OK;
}
// ----------------------------------------------------------------------------
diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.h b/webrtc/modules/audio_device/win/audio_device_wave_win.h
index 402d575..6bb7c17 100644
--- a/webrtc/modules/audio_device/win/audio_device_wave_win.h
+++ b/webrtc/modules/audio_device/win/audio_device_wave_win.h
@@ -50,7 +50,7 @@
virtual int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
- virtual int32_t Init();
+ virtual InitStatus Init();
virtual int32_t Terminate();
virtual bool Initialized() const;
diff --git a/webrtc/system_wrappers/include/metrics.h b/webrtc/system_wrappers/include/metrics.h
index d5e5492..0fee344 100644
--- a/webrtc/system_wrappers/include/metrics.h
+++ b/webrtc/system_wrappers/include/metrics.h
@@ -125,6 +125,14 @@
#define RTC_LOGGED_HISTOGRAM_PERCENTAGE(name, sample) \
RTC_LOGGED_HISTOGRAM_ENUMERATION(name, sample, 101)
+// Histogram for booleans.
+#define RTC_HISTOGRAM_BOOLEAN(name, sample) \
+ RTC_HISTOGRAM_ENUMERATION(name, sample, 2)
+
+// RTC_HISTOGRAM_BOOLEAN with logging.
+#define RTC_LOGGED_HISTOGRAM_BOOLEAN(name, sample) \
+ RTC_LOGGED_HISTOGRAM_ENUMERATION(name, sample, 2)
+
// Histogram for enumerators (evenly spaced buckets).
// |boundary| should be above the max enumerator sample.
#define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \
