Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(165)

Side by Side Diff: webrtc/test/call_test.cc

Issue 1238033003: Prevent OOB reads for truncated H264 STAP-A packets. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: added logging + todos Created 5 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 #include "webrtc/test/call_test.h" 10 #include "webrtc/test/call_test.h"
(...skipping 114 matching lines...) Expand 10 before | Expand all | Expand 10 after
125 125
126 void CallTest::CreateFrameGeneratorCapturer() { 126 void CallTest::CreateFrameGeneratorCapturer() {
127 VideoStream stream = encoder_config_.streams.back(); 127 VideoStream stream = encoder_config_.streams.back();
128 frame_generator_capturer_.reset( 128 frame_generator_capturer_.reset(
129 test::FrameGeneratorCapturer::Create(send_stream_->Input(), 129 test::FrameGeneratorCapturer::Create(send_stream_->Input(),
130 stream.width, 130 stream.width,
131 stream.height, 131 stream.height,
132 stream.max_framerate, 132 stream.max_framerate,
133 clock_)); 133 clock_));
134 } 134 }
135
135 void CallTest::CreateStreams() { 136 void CallTest::CreateStreams() {
136 assert(send_stream_ == NULL); 137 assert(send_stream_ == NULL);
137 assert(receive_streams_.empty()); 138 assert(receive_streams_.empty());
138 139
139 send_stream_ = 140 send_stream_ =
140 sender_call_->CreateVideoSendStream(send_config_, encoder_config_); 141 sender_call_->CreateVideoSendStream(send_config_, encoder_config_);
141 142
142 for (size_t i = 0; i < receive_configs_.size(); ++i) { 143 for (size_t i = 0; i < receive_configs_.size(); ++i) {
143 receive_streams_.push_back( 144 receive_streams_.push_back(
144 receiver_call_->CreateVideoReceiveStream(receive_configs_[i])); 145 receiver_call_->CreateVideoReceiveStream(receive_configs_[i]));
(...skipping 85 matching lines...) Expand 10 before | Expand all | Expand 10 after
230 const FakeNetworkPipe::Config& config) 231 const FakeNetworkPipe::Config& config)
231 : BaseTest(timeout_ms, config) { 232 : BaseTest(timeout_ms, config) {
232 } 233 }
233 234
234 bool EndToEndTest::ShouldCreateReceivers() const { 235 bool EndToEndTest::ShouldCreateReceivers() const {
235 return true; 236 return true;
236 } 237 }
237 238
238 } // namespace test 239 } // namespace test
239 } // namespace webrtc 240 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/video_coding/main/source/session_info.cc ('k') | webrtc/video/packet_injection_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698