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/datachannelinterface.h b/webrtc/api/datachannelinterface.h
new file mode 100644
index 0000000..e291328
--- /dev/null
+++ b/webrtc/api/datachannelinterface.h
@@ -0,0 +1,159 @@
+/*
+ * libjingle
+ * Copyright 2012 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 DataChannels
+// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
+
+#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_
+#define WEBRTC_API_DATACHANNELINTERFACE_H_
+
+#include <string>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/refcount.h"
+
+
+namespace webrtc {
+
+struct DataChannelInit {
+ DataChannelInit()
+ : reliable(false),
+ ordered(true),
+ maxRetransmitTime(-1),
+ maxRetransmits(-1),
+ negotiated(false),
+ id(-1) {
+ }
+
+ bool reliable; // Deprecated.
+ bool ordered; // True if ordered delivery is required.
+ int maxRetransmitTime; // The max period of time in milliseconds in which
+ // retransmissions will be sent. After this time, no
+ // more retransmissions will be sent. -1 if unset.
+ int maxRetransmits; // The max number of retransmissions. -1 if unset.
+ std::string protocol; // This is set by the application and opaque to the
+ // WebRTC implementation.
+ bool negotiated; // True if the channel has been externally negotiated
+ // and we do not send an in-band signalling in the
+ // form of an "open" message.
+ int id; // The stream id, or SID, for SCTP data channels. -1
+ // if unset.
+};
+
+struct DataBuffer {
+ DataBuffer(const rtc::Buffer& data, bool binary)
+ : data(data),
+ binary(binary) {
+ }
+ // For convenience for unit tests.
+ explicit DataBuffer(const std::string& text)
+ : data(text.data(), text.length()),
+ binary(false) {
+ }
+ size_t size() const { return data.size(); }
+
+ rtc::Buffer data;
+ // Indicates if the received data contains UTF-8 or binary data.
+ // Note that the upper layers are left to verify the UTF-8 encoding.
+ // TODO(jiayl): prefer to use an enum instead of a bool.
+ bool binary;
+};
+
+class DataChannelObserver {
+ public:
+ // The data channel state have changed.
+ virtual void OnStateChange() = 0;
+ // A data buffer was successfully received.
+ virtual void OnMessage(const DataBuffer& buffer) = 0;
+ // The data channel's buffered_amount has changed.
+ virtual void OnBufferedAmountChange(uint64_t previous_amount){};
+
+ protected:
+ virtual ~DataChannelObserver() {}
+};
+
+class DataChannelInterface : public rtc::RefCountInterface {
+ public:
+ // Keep in sync with DataChannel.java:State and
+ // RTCDataChannel.h:RTCDataChannelState.
+ enum DataState {
+ kConnecting,
+ kOpen, // The DataChannel is ready to send data.
+ kClosing,
+ kClosed
+ };
+
+ static const char* DataStateString(DataState state) {
+ switch (state) {
+ case kConnecting:
+ return "connecting";
+ case kOpen:
+ return "open";
+ case kClosing:
+ return "closing";
+ case kClosed:
+ return "closed";
+ }
+ RTC_CHECK(false) << "Unknown DataChannel state: " << state;
+ return "";
+ }
+
+ virtual void RegisterObserver(DataChannelObserver* observer) = 0;
+ virtual void UnregisterObserver() = 0;
+ // The label attribute represents a label that can be used to distinguish this
+ // DataChannel object from other DataChannel objects.
+ virtual std::string label() const = 0;
+ virtual bool reliable() const = 0;
+
+ // TODO(tommyw): Remove these dummy implementations when all classes have
+ // implemented these APIs. They should all just return the values the
+ // DataChannel was created with.
+ virtual bool ordered() const { return false; }
+ virtual uint16_t maxRetransmitTime() const { return 0; }
+ virtual uint16_t maxRetransmits() const { return 0; }
+ virtual std::string protocol() const { return std::string(); }
+ virtual bool negotiated() const { return false; }
+
+ virtual int id() const = 0;
+ virtual DataState state() const = 0;
+ // The buffered_amount returns the number of bytes of application data
+ // (UTF-8 text and binary data) that have been queued using SendBuffer but
+ // have not yet been transmitted to the network.
+ virtual uint64_t buffered_amount() const = 0;
+ virtual void Close() = 0;
+ // Sends |data| to the remote peer.
+ virtual bool Send(const DataBuffer& buffer) = 0;
+
+ protected:
+ virtual ~DataChannelInterface() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_DATACHANNELINTERFACE_H_
