Correcting uninitialized members.

BUG=C-10344, C-10345, C-10346

Review URL: https://webrtc-codereview.appspot.com/345012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1525 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-01-24 07:49:33 +00:00
parent a5a5cbb992
commit ba09cf16ec
4 changed files with 28 additions and 23 deletions

View File

@ -33,6 +33,7 @@ bool StreamVideoFileRepeatedlyIntoCaptureDevice(void* data) {
ViEFakeCamera::ViEFakeCamera(webrtc::ViECapture* capture_interface)
: capture_interface_(capture_interface),
capture_id_(-1),
camera_thread_(NULL),
file_capture_device_(NULL) {
}

View File

@ -47,32 +47,33 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
rtp_rtcp_(*RtpRtcp::CreateRtpRtcp(ViEModuleId(engine_id, channel_id),
false)),
default_rtp_rtcp_(NULL),
#ifndef WEBRTC_EXTERNAL_TRANSPORT
socket_transport_(*UdpTransport::Create(
ViEModuleId(engine_id, channel_id), num_socket_threads_)),
#endif
vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
vie_receiver_(*(new ViEReceiver(engine_id, channel_id, rtp_rtcp_, vcm_))),
vie_sender_(*(new ViESender(engine_id, channel_id))),
vie_sync_(*(new ViESyncModule(ViEId(engine_id, channel_id), vcm_,
rtp_rtcp_))),
module_process_thread_(module_process_thread),
codec_observer_(NULL),
do_key_frame_callbackRequest_(false),
rtp_observer_(NULL),
rtcp_observer_(NULL),
networkObserver_(NULL),
rtp_packet_timeout_(false),
using_packet_spread_(false),
external_transport_(NULL),
decoder_reset_(true),
wait_for_key_frame_(false),
decode_thread_(NULL),
external_encryption_(NULL),
effect_filter_(NULL),
color_enhancement_(true),
vcm_rttreported_(TickTime::Now()),
file_recorder_(channel_id) {
vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
vie_receiver_(*(new ViEReceiver(engine_id, channel_id, rtp_rtcp_, vcm_))),
vie_sender_(*(new ViESender(engine_id, channel_id))),
vie_sync_(*(new ViESyncModule(ViEId(engine_id, channel_id), vcm_,
rtp_rtcp_))),
module_process_thread_(module_process_thread),
codec_observer_(NULL),
do_key_frame_callbackRequest_(false),
rtp_observer_(NULL),
rtcp_observer_(NULL),
networkObserver_(NULL),
rtp_packet_timeout_(false),
using_packet_spread_(false),
external_transport_(NULL),
decoder_reset_(true),
wait_for_key_frame_(false),
decode_thread_(NULL),
external_encryption_(NULL),
effect_filter_(NULL),
color_enhancement_(true),
vcm_rttreported_(TickTime::Now()),
file_recorder_(channel_id) {
WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, channel_id),
"ViEChannel::ViEChannel(channel_id: %d, engine_id: %d)",
channel_id, engine_id);

View File

@ -61,6 +61,8 @@ ViEFilePlayer::ViEFilePlayer(int Id,
decode_thread_(NULL),
decode_event_(NULL),
decoded_audio_length_(0) {
memset(file_name_, 0, FileWrapper::kMaxFileNameSize);
memset(decoded_audio_, 0, kMaxDecodedAudioLength);
}
ViEFilePlayer::~ViEFilePlayer() {

View File

@ -93,6 +93,7 @@ class ViEFilePlayer
virtual void RecordFileEnded(const WebRtc_Word32 /*id*/) {}
private:
static const int kMaxDecodedAudioLength = 320;
bool play_back_started_;
ViEInputManager& input_manager_;
@ -121,7 +122,7 @@ class ViEFilePlayer
// Thread for decoding video (and audio if no audio clients connected).
ThreadWrapper* decode_thread_;
EventWrapper* decode_event_;
WebRtc_Word16 decoded_audio_[320];
WebRtc_Word16 decoded_audio_[kMaxDecodedAudioLength];
WebRtc_UWord32 decoded_audio_length_;
// Trick - list containing VoE buffer reading this file. Used if multiple