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/sctputils.cc b/webrtc/api/sctputils.cc
new file mode 100644
index 0000000..84cb293
--- /dev/null
+++ b/webrtc/api/sctputils.cc
@@ -0,0 +1,205 @@
+/*
+ * libjingle
+ * Copyright 2013 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.
+ */
+
+#include "webrtc/api/sctputils.h"
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+// Format defined at
+// http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section
+
+static const uint8_t DATA_CHANNEL_OPEN_MESSAGE_TYPE = 0x03;
+static const uint8_t DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE = 0x02;
+
+enum DataChannelOpenMessageChannelType {
+ DCOMCT_ORDERED_RELIABLE = 0x00,
+ DCOMCT_ORDERED_PARTIAL_RTXS = 0x01,
+ DCOMCT_ORDERED_PARTIAL_TIME = 0x02,
+ DCOMCT_UNORDERED_RELIABLE = 0x80,
+ DCOMCT_UNORDERED_PARTIAL_RTXS = 0x81,
+ DCOMCT_UNORDERED_PARTIAL_TIME = 0x82,
+};
+
+bool IsOpenMessage(const rtc::Buffer& payload) {
+ // Format defined at
+ // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
+
+ rtc::ByteBuffer buffer(payload);
+ uint8_t message_type;
+ if (!buffer.ReadUInt8(&message_type)) {
+ LOG(LS_WARNING) << "Could not read OPEN message type.";
+ return false;
+ }
+ return message_type == DATA_CHANNEL_OPEN_MESSAGE_TYPE;
+}
+
+bool ParseDataChannelOpenMessage(const rtc::Buffer& payload,
+ std::string* label,
+ DataChannelInit* config) {
+ // Format defined at
+ // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
+
+ rtc::ByteBuffer buffer(payload);
+ uint8_t message_type;
+ if (!buffer.ReadUInt8(&message_type)) {
+ LOG(LS_WARNING) << "Could not read OPEN message type.";
+ return false;
+ }
+ if (message_type != DATA_CHANNEL_OPEN_MESSAGE_TYPE) {
+ LOG(LS_WARNING) << "Data Channel OPEN message of unexpected type: "
+ << message_type;
+ return false;
+ }
+
+ uint8_t channel_type;
+ if (!buffer.ReadUInt8(&channel_type)) {
+ LOG(LS_WARNING) << "Could not read OPEN message channel type.";
+ return false;
+ }
+
+ uint16_t priority;
+ if (!buffer.ReadUInt16(&priority)) {
+ LOG(LS_WARNING) << "Could not read OPEN message reliabilility prioirty.";
+ return false;
+ }
+ uint32_t reliability_param;
+ if (!buffer.ReadUInt32(&reliability_param)) {
+ LOG(LS_WARNING) << "Could not read OPEN message reliabilility param.";
+ return false;
+ }
+ uint16_t label_length;
+ if (!buffer.ReadUInt16(&label_length)) {
+ LOG(LS_WARNING) << "Could not read OPEN message label length.";
+ return false;
+ }
+ uint16_t protocol_length;
+ if (!buffer.ReadUInt16(&protocol_length)) {
+ LOG(LS_WARNING) << "Could not read OPEN message protocol length.";
+ return false;
+ }
+ if (!buffer.ReadString(label, (size_t) label_length)) {
+ LOG(LS_WARNING) << "Could not read OPEN message label";
+ return false;
+ }
+ if (!buffer.ReadString(&config->protocol, protocol_length)) {
+ LOG(LS_WARNING) << "Could not read OPEN message protocol.";
+ return false;
+ }
+
+ config->ordered = true;
+ switch (channel_type) {
+ case DCOMCT_UNORDERED_RELIABLE:
+ case DCOMCT_UNORDERED_PARTIAL_RTXS:
+ case DCOMCT_UNORDERED_PARTIAL_TIME:
+ config->ordered = false;
+ }
+
+ config->maxRetransmits = -1;
+ config->maxRetransmitTime = -1;
+ switch (channel_type) {
+ case DCOMCT_ORDERED_PARTIAL_RTXS:
+ case DCOMCT_UNORDERED_PARTIAL_RTXS:
+ config->maxRetransmits = reliability_param;
+ break;
+ case DCOMCT_ORDERED_PARTIAL_TIME:
+ case DCOMCT_UNORDERED_PARTIAL_TIME:
+ config->maxRetransmitTime = reliability_param;
+ break;
+ }
+ return true;
+}
+
+bool ParseDataChannelOpenAckMessage(const rtc::Buffer& payload) {
+ rtc::ByteBuffer buffer(payload);
+ uint8_t message_type;
+ if (!buffer.ReadUInt8(&message_type)) {
+ LOG(LS_WARNING) << "Could not read OPEN_ACK message type.";
+ return false;
+ }
+ if (message_type != DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE) {
+ LOG(LS_WARNING) << "Data Channel OPEN_ACK message of unexpected type: "
+ << message_type;
+ return false;
+ }
+ return true;
+}
+
+bool WriteDataChannelOpenMessage(const std::string& label,
+ const DataChannelInit& config,
+ rtc::Buffer* payload) {
+ // Format defined at
+ // http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-00#section-6.1
+ uint8_t channel_type = 0;
+ uint32_t reliability_param = 0;
+ uint16_t priority = 0;
+ if (config.ordered) {
+ if (config.maxRetransmits > -1) {
+ channel_type = DCOMCT_ORDERED_PARTIAL_RTXS;
+ reliability_param = config.maxRetransmits;
+ } else if (config.maxRetransmitTime > -1) {
+ channel_type = DCOMCT_ORDERED_PARTIAL_TIME;
+ reliability_param = config.maxRetransmitTime;
+ } else {
+ channel_type = DCOMCT_ORDERED_RELIABLE;
+ }
+ } else {
+ if (config.maxRetransmits > -1) {
+ channel_type = DCOMCT_UNORDERED_PARTIAL_RTXS;
+ reliability_param = config.maxRetransmits;
+ } else if (config.maxRetransmitTime > -1) {
+ channel_type = DCOMCT_UNORDERED_PARTIAL_TIME;
+ reliability_param = config.maxRetransmitTime;
+ } else {
+ channel_type = DCOMCT_UNORDERED_RELIABLE;
+ }
+ }
+
+ rtc::ByteBuffer buffer(
+ NULL, 20 + label.length() + config.protocol.length(),
+ rtc::ByteBuffer::ORDER_NETWORK);
+ buffer.WriteUInt8(DATA_CHANNEL_OPEN_MESSAGE_TYPE);
+ buffer.WriteUInt8(channel_type);
+ buffer.WriteUInt16(priority);
+ buffer.WriteUInt32(reliability_param);
+ buffer.WriteUInt16(static_cast<uint16_t>(label.length()));
+ buffer.WriteUInt16(static_cast<uint16_t>(config.protocol.length()));
+ buffer.WriteString(label);
+ buffer.WriteString(config.protocol);
+ payload->SetData(buffer.Data(), buffer.Length());
+ return true;
+}
+
+void WriteDataChannelOpenAckMessage(rtc::Buffer* payload) {
+ rtc::ByteBuffer buffer(rtc::ByteBuffer::ORDER_NETWORK);
+ buffer.WriteUInt8(DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE);
+ payload->SetData(buffer.Data(), buffer.Length());
+}
+} // namespace webrtc
