From eff3c8905f115dd3603e99237993119a1a09a6ec Mon Sep 17 00:00:00 2001
From: "xians@webrtc.org"
 <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Tue, 6 Dec 2011 10:02:56 +0000
Subject: [PATCH] this patch fixes the valgrind warnings in the adm api test
 for pulseaudio in linux. Review URL:
 http://webrtc-codereview.appspot.com/301012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1108 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 .../main/source/audio_device_buffer.cc        |  2 ++
 .../source/linux/audio_device_pulse_linux.cc  | 19 ++++++++++++++++---
 .../main/test/audio_device_test_api.cc        |  5 ++++-
 3 files changed, 22 insertions(+), 4 deletions(-)

diff --git a/src/modules/audio_device/main/source/audio_device_buffer.cc b/src/modules/audio_device/main/source/audio_device_buffer.cc
index b5a78e3e74..e8bbfdac2a 100644
--- a/src/modules/audio_device/main/source/audio_device_buffer.cc
+++ b/src/modules/audio_device/main/source/audio_device_buffer.cc
@@ -55,6 +55,8 @@ AudioDeviceBuffer::AudioDeviceBuffer() :
 {
     // valid ID will be set later by SetId, use -1 for now
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
+    memset(_recBuffer, 0, kMaxBufferSizeBytes);
+    memset(_playBuffer, 0, kMaxBufferSizeBytes);
 }
 
 // ----------------------------------------------------------------------------
diff --git a/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc b/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
index 512121634c..4c3be521f9 100644
--- a/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
+++ b/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
@@ -133,6 +133,8 @@ AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const WebRtc_Word32 id) :
                  "%s created", __FUNCTION__);
 
     memset(_paServerVersion, 0, sizeof(_paServerVersion));
+    memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
+    memset(&_recBufferAttr, 0, sizeof(_recBufferAttr));
 }
 
 AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
@@ -1115,7 +1117,7 @@ WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutDevice(WebRtc_UWord16 index)
         return -1;
     }
 
-    const WebRtc_UWord16 nDevices(PlayoutDevices());
+    const WebRtc_UWord16 nDevices = PlayoutDevices();
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "  number of availiable output devices is %u", nDevices);
@@ -1149,7 +1151,7 @@ WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutDeviceName(
     WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
                  "AudioDeviceLinuxPulse::PlayoutDeviceName(index=%u)", index);
 
-    const WebRtc_UWord16 nDevices(PlayoutDevices());
+    const WebRtc_UWord16 nDevices = PlayoutDevices();
 
     if ((index > (nDevices - 1)) || (name == NULL))
     {
@@ -2239,7 +2241,7 @@ WebRtc_Word32 AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
                                                           WebRtc_Word8* name,
                                                           WebRtc_UWord16& index)
 {
-    WebRtc_Word8 tmpName[kAdmMaxDeviceNameSize];
+    WebRtc_Word8 tmpName[kAdmMaxDeviceNameSize] = {0};
     // subtract length of "default: "
     WebRtc_UWord16 nameLen = kAdmMaxDeviceNameSize - 9;
     WebRtc_Word8* pName = NULL;
@@ -2333,6 +2335,11 @@ WebRtc_Word32 AudioDeviceLinuxPulse::InitPulseAudio()
 
     // Create a mainloop API and connection to the default server
     // the mainloop is the internal asynchronous API event loop
+    if (_paMainloop) {
+        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                     "  PA mainloop has already existed");
+        return -1;
+    }
     _paMainloop = LATE(pa_threaded_mainloop_new)();
     if (!_paMainloop)
     {
@@ -2365,6 +2372,12 @@ WebRtc_Word32 AudioDeviceLinuxPulse::InitPulseAudio()
     }
 
     // Create a new PulseAudio context
+    if (_paContext){
+        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                     "  PA context has already existed");
+        PaUnLock();
+        return -1;
+    }
     _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
 
     if (!_paContext)
diff --git a/src/modules/audio_device/main/test/audio_device_test_api.cc b/src/modules/audio_device/main/test/audio_device_test_api.cc
index 5f5ce26ba5..289908bd99 100644
--- a/src/modules/audio_device/main/test/audio_device_test_api.cc
+++ b/src/modules/audio_device/main/test/audio_device_test_api.cc
@@ -83,6 +83,8 @@ class AudioEventObserverAPI: public AudioDeviceObserver
 {
 public:
     AudioEventObserverAPI(AudioDeviceModule* audioDevice) :
+        _error(kRecordingError),
+        _warning(kRecordingWarning),
         _audioDevice(audioDevice)
     {
     }
@@ -277,10 +279,11 @@ int api_test()
         myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
     TEST((audioDevice = AudioDeviceModuleImpl::Create(
         myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
-    // create default implementation (=ALSA Audio) instance
+    // create default implementation instance
     TEST((audioDevice = AudioDeviceModuleImpl::Create(
         myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
     audioDevice->AddRef();
+    TEST(audioDevice->Terminate() == 0);
     TEST(audioDevice->Release() == 0);
     // explicitly specify usage of Pulse Audio (same as default)
     TEST((audioDevice = AudioDeviceModuleImpl::Create(