Move talk/app/webrtc to webrtc/api
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.
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
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc 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/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
new file mode 100644
index 0000000..961d869
--- /dev/null
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -0,0 +1,66 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// This file contains interfaces for RtpReceivers
+// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
+
+#ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
+#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
+
+#include <string>
+
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/proxy.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class RtpReceiverInterface : public rtc::RefCountInterface {
+ public:
+ virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+ // Not to be confused with "mid", this is a field we can temporarily use
+ // to uniquely identify a receiver until we implement Unified Plan SDP.
+ virtual std::string id() const = 0;
+
+ virtual void Stop() = 0;
+
+ protected:
+ virtual ~RtpReceiverInterface() {}
+};
+
+// Define proxy for RtpReceiverInterface.
+BEGIN_PROXY_MAP(RtpReceiver)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(std::string, id)
+PROXY_METHOD0(void, Stop)
+END_PROXY()
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_