a96e2d77cb
I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
96 lines
3.2 KiB
C++
96 lines
3.2 KiB
C++
/*
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* libjingle
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* Copyright 2013 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/remotevideocapturer.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/media/base/videoframe.h"
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namespace webrtc {
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RemoteVideoCapturer::RemoteVideoCapturer() {}
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RemoteVideoCapturer::~RemoteVideoCapturer() {}
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cricket::CaptureState RemoteVideoCapturer::Start(
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const cricket::VideoFormat& capture_format) {
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if (capture_state() == cricket::CS_RUNNING) {
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LOG(LS_WARNING)
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<< "RemoteVideoCapturer::Start called when it's already started.";
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return capture_state();
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}
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LOG(LS_INFO) << "RemoteVideoCapturer::Start";
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SetCaptureFormat(&capture_format);
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return cricket::CS_RUNNING;
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}
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void RemoteVideoCapturer::Stop() {
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if (capture_state() == cricket::CS_STOPPED) {
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LOG(LS_WARNING)
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<< "RemoteVideoCapturer::Stop called when it's already stopped.";
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return;
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}
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LOG(LS_INFO) << "RemoteVideoCapturer::Stop";
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SetCaptureFormat(NULL);
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SetCaptureState(cricket::CS_STOPPED);
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}
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bool RemoteVideoCapturer::IsRunning() {
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return capture_state() == cricket::CS_RUNNING;
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}
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bool RemoteVideoCapturer::GetPreferredFourccs(std::vector<uint32_t>* fourccs) {
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if (!fourccs)
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return false;
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fourccs->push_back(cricket::FOURCC_I420);
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return true;
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}
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bool RemoteVideoCapturer::GetBestCaptureFormat(
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const cricket::VideoFormat& desired, cricket::VideoFormat* best_format) {
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if (!best_format) {
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return false;
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}
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// RemoteVideoCapturer does not support capability enumeration.
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// Use the desired format as the best format.
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best_format->width = desired.width;
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best_format->height = desired.height;
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best_format->fourcc = cricket::FOURCC_I420;
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best_format->interval = desired.interval;
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return true;
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}
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bool RemoteVideoCapturer::IsScreencast() const {
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// TODO(ronghuawu): what about remote screencast stream.
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return false;
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}
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} // namespace webrtc
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