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Fix heap-use-after-free in WebRtcVideoEngine2.

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Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44779004

Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2015-03-20 14:27:49 +00:00
parent a3209a2b27
commit 8296ec518b
3 changed files with 8 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
talk/media/webrtc/webrtcvideoengine2.cc
View File

@ -577,7 +577,10 @@ WebRtcVideoChannel2::WebRtcVideoChannel2(
WebRtcVideoEncoderFactory* external_encoder_factory, WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory) WebRtcVideoDecoderFactory* external_decoder_factory)
: unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
voice_channel_(voice_channel), voice_channel_id_(voice_channel != nullptr
? static_cast<WebRtcVoiceMediaChannel*>(
voice_channel)->voe_channel()
: -1),
external_encoder_factory_(external_encoder_factory), external_encoder_factory_(external_encoder_factory),
external_decoder_factory_(external_decoder_factory) { external_decoder_factory_(external_decoder_factory) {
SetDefaultOptions(); SetDefaultOptions();
@ -909,10 +912,9 @@ bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
// the SSRC of the remote audio channel in order to sync the correct webrtc // the SSRC of the remote audio channel in order to sync the correct webrtc
// VoiceEngine channel. For now sync the first channel in non-conference to // VoiceEngine channel. For now sync the first channel in non-conference to
// match existing behavior in WebRtcVideoEngine. // match existing behavior in WebRtcVideoEngine.
if (voice_channel_ != NULL && receive_streams_.empty() && if (voice_channel_id_ != -1 && receive_streams_.empty() &&
!options_.conference_mode.GetWithDefaultIfUnset(false)) { !options_.conference_mode.GetWithDefaultIfUnset(false)) {
config.audio_channel_id = config.audio_channel_id = voice_channel_id_;
static_cast<WebRtcVoiceMediaChannel*>(voice_channel_)->voe_channel();
} }
receive_streams_[ssrc] = receive_streams_[ssrc] =

2
talk/media/webrtc/webrtcvideoengine2.h
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@ -485,7 +485,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
Settable<VideoCodecSettings> send_codec_; Settable<VideoCodecSettings> send_codec_;
std::vector<webrtc::RtpExtension> send_rtp_extensions_; std::vector<webrtc::RtpExtension> send_rtp_extensions_;
VoiceMediaChannel* const voice_channel_; const int voice_channel_id_;
WebRtcVideoEncoderFactory* const external_encoder_factory_; WebRtcVideoEncoderFactory* const external_encoder_factory_;
WebRtcVideoDecoderFactory* const external_decoder_factory_; WebRtcVideoDecoderFactory* const external_decoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_; std::vector<VideoCodecSettings> recv_codecs_;

2
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -323,7 +323,7 @@ class WebRtcMediaChannel : public T, public webrtc::Transport {
private: private:
E *engine_; E *engine_;
int voe_channel_; const int voe_channel_;
}; };
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses