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Adding ViE NULL checks.

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BUG=C-10188, C-10246, C-10595

Review URL: https://webrtc-codereview.appspot.com/373013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1559 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-01-27 09:11:22 +00:00
parent cbe1de9aa6
commit 194a93ac66
3 changed files with 23 additions and 6 deletions
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13
src/video_engine/test/auto_test/source/tb_external_transport.cc
View File

@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
@ -350,7 +350,16 @@ bool TbExternalTransport::ViEExternalTransportProcess()
// Take first packet in queue // Take first packet in queue
_crit.Enter(); _crit.Enter();
packet = _rtcpPackets.front(); packet = _rtcpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs(); WebRtc_Word64 timeToReceive = 0;
if (packet)
{
timeToReceive = packet->receiveTime - NowMs();
}
else
{
// There should never be any empty packets in the list.
assert(false);
}
if (timeToReceive > 0) if (timeToReceive > 0)
{ {
// No packets to receive yet // No packets to receive yet

9
src/video_engine/vie_file_impl.cc
View File

@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
@ -376,8 +376,11 @@ int ViEFileImpl::StartRecordOutgoingVideo(const int video_channel,
VoiceEngine* ve_ptr = NULL; VoiceEngine* ve_ptr = NULL;
if (audio_source != NO_AUDIO) { if (audio_source != NO_AUDIO) {
ViEChannel* vie_channel = cs.Channel(video_channel); ViEChannel* vie_channel = cs.Channel(video_channel);
// Channel should exists since we have a ViEEncoder. if (!vie_channel) {
assert(vie_channel); // Channel should exists since we have a ViEEncoder above.
assert(false);
return -1;
}
ve_channel_id = vie_channel->VoiceChannel(); ve_channel_id = vie_channel->VoiceChannel();
ve_ptr = shared_data_->channel_manager()->GetVoiceEngine(); ve_ptr = shared_data_->channel_manager()->GetVoiceEngine();
if (!ve_ptr) { if (!ve_ptr) {

7
src/video_engine/vie_file_player.cc
View File

@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
@ -393,6 +393,11 @@ int ViEFilePlayer::SendAudioOnChannel(const int audio_channel,
int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) { int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) {
int result = 0; int result = 0;
if (!voe_file_interface_) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
"ViEFilePlayer::StopSendAudioOnChannel() - no VoE interface");
return -1;
}
std::set<int>::iterator it = audio_channels_sending_.find(audio_channel); std::set<int>::iterator it = audio_channels_sending_.find(audio_channel);
if (it == audio_channels_sending_.end()) { if (it == audio_channels_sending_.end()) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_), WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),