Update talk to 54898858.
TEST=try bots
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/2414004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/mediastreamsignaling.cc b/talk/app/webrtc/mediastreamsignaling.cc
index a23799f..771a4e8 100644
--- a/talk/app/webrtc/mediastreamsignaling.cc
+++ b/talk/app/webrtc/mediastreamsignaling.cc
@@ -916,144 +916,6 @@
}
-// Format defined at
-// http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
-const uint8 DATA_CHANNEL_OPEN_MESSAGE_TYPE = 0x03;
-
-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 MediaStreamSignaling::ParseDataChannelOpenMessage(
- const talk_base::Buffer& payload,
- std::string* label,
- DataChannelInit* config) {
- // Format defined at
- // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
-
- talk_base::ByteBuffer buffer(payload.data(), payload.length());
-
- uint8 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 channel_type;
- if (!buffer.ReadUInt8(&channel_type)) {
- LOG(LS_WARNING) << "Could not read OPEN message channel type.";
- return false;
- }
- uint16 priority;
- if (!buffer.ReadUInt16(&priority)) {
- LOG(LS_WARNING) << "Could not read OPEN message reliabilility prioirty.";
- return false;
- }
- uint32 reliability_param;
- if (!buffer.ReadUInt32(&reliability_param)) {
- LOG(LS_WARNING) << "Could not read OPEN message reliabilility param.";
- return false;
- }
- uint16 label_length;
- if (!buffer.ReadUInt16(&label_length)) {
- LOG(LS_WARNING) << "Could not read OPEN message label length.";
- return false;
- }
- uint16 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;
-
- case DCOMCT_ORDERED_PARTIAL_TIME:
- case DCOMCT_UNORDERED_PARTIAL_TIME:
- config->maxRetransmitTime = reliability_param;
- }
-
- return true;
-}
-
-bool MediaStreamSignaling::WriteDataChannelOpenMessage(
- const std::string& label,
- const DataChannelInit& config,
- talk_base::Buffer* payload) {
- // Format defined at
- // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
- // TODO(pthatcher)
-
- uint8 channel_type = 0;
- uint32 reliability_param = 0;
- uint16 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;
- }
- }
-
- talk_base::ByteBuffer buffer(
- NULL, 20 + label.length() + config.protocol.length(),
- talk_base::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>(label.length()));
- buffer.WriteUInt16(static_cast<uint16>(config.protocol.length()));
- buffer.WriteString(label);
- buffer.WriteString(config.protocol);
- payload->SetData(buffer.Data(), buffer.Length());
- return true;
-}
-
void MediaStreamSignaling::UpdateLocalSctpDataChannels() {
DataChannels::iterator it = data_channels_.begin();
for (; it != data_channels_.end(); ++it) {
diff --git a/talk/app/webrtc/mediastreamsignaling.h b/talk/app/webrtc/mediastreamsignaling.h
index f64bf97..067ed2f 100644
--- a/talk/app/webrtc/mediastreamsignaling.h
+++ b/talk/app/webrtc/mediastreamsignaling.h
@@ -196,12 +196,6 @@
// After we receive an OPEN message, create a data channel and add it.
bool AddDataChannelFromOpenMessage(
const std::string& label, const DataChannelInit& config);
- bool ParseDataChannelOpenMessage(
- const talk_base::Buffer& payload, std::string* label,
- DataChannelInit* config);
- bool WriteDataChannelOpenMessage(
- const std::string& label, const DataChannelInit& config,
- talk_base::Buffer* payload);
// Returns a MediaSessionOptions struct with options decided by |constraints|,
// the local MediaStreams and DataChannels.
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index c7805c1..74d1488 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -42,6 +42,7 @@
#include "talk/base/stringencode.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/videocapturer.h"
+#include "talk/media/sctp/sctputils.h"
#include "talk/session/media/channel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
@@ -1031,8 +1032,7 @@
}
if (!config->negotiated) {
talk_base::Buffer *payload = new talk_base::Buffer;
- if (!mediastream_signaling_->WriteDataChannelOpenMessage(
- label, *config, payload)) {
+ if (!cricket::WriteDataChannelOpenMessage(label, *config, payload)) {
LOG(LS_WARNING) << "Could not write data channel OPEN message";
}
// SendControl may queue the message until the data channel's set up,
@@ -1368,8 +1368,8 @@
if (!data_channel_.get()) {
return false;
}
- data_channel_->SignalDataReceived.connect(
- this, &WebRtcSession::OnDataReceived);
+ data_channel_->SignalNewStreamReceived.connect(
+ this, &WebRtcSession::OnNewDataChannelReceived);
return true;
}
@@ -1386,27 +1386,11 @@
saved_candidates_.clear();
}
-// Look for OPEN messages and set up data channels in response.
-void WebRtcSession::OnDataReceived(
- cricket::DataChannel* channel,
- const cricket::ReceiveDataParams& params,
- const talk_base::Buffer& payload) {
- if (params.type != cricket::DMT_CONTROL) {
- return;
- }
-
- std::string label;
- DataChannelInit config;
- if (!mediastream_signaling_->ParseDataChannelOpenMessage(
- payload, &label, &config)) {
- LOG(LS_WARNING) << "Failed to parse data channel OPEN message.";
- return;
- }
-
- config.negotiated = true; // This is the negotiation.
-
+void WebRtcSession::OnNewDataChannelReceived(
+ const std::string& label, const DataChannelInit& init) {
+ ASSERT(data_channel_type_ == cricket::DCT_SCTP);
if (!mediastream_signaling_->AddDataChannelFromOpenMessage(
- label, config)) {
+ label, init)) {
LOG(LS_WARNING) << "Failed to create data channel from OPEN message.";
return;
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index be89798..12e2291 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -271,10 +271,8 @@
// The |saved_candidates_| will be cleared after this function call.
void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
- void OnDataReceived(
- cricket::DataChannel* channel,
- const cricket::ReceiveDataParams& params,
- const talk_base::Buffer& payload);
+ void OnNewDataChannelReceived(const std::string& label,
+ const DataChannelInit& init);
bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index f9edd83..16044b5 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -841,6 +841,8 @@
# TODO(ronghuawu): Enable when SCTP is ready.
# 'media/sctp/sctpdataengine.cc',
# 'media/sctp/sctpdataengine.h',
+ 'media/sctp/sctputils.cc',
+ 'media/sctp/sctputils.h',
'media/webrtc/webrtccommon.h',
'media/webrtc/webrtcexport.h',
'media/webrtc/webrtcmediaengine.h',
diff --git a/talk/libjingle.scons b/talk/libjingle.scons
index cfc0249..87b43f5 100644
--- a/talk/libjingle.scons
+++ b/talk/libjingle.scons
@@ -290,6 +290,7 @@
"media/base/videoframe.cc",
"media/devices/devicemanager.cc",
"media/devices/filevideocapturer.cc",
+ "media/sctp/sctputils.cc",
"session/media/audiomonitor.cc",
"session/media/call.cc",
"session/media/channel.cc",
@@ -676,6 +677,7 @@
"media/base/videocommon_unittest.cc",
"media/devices/devicemanager_unittest.cc",
"media/devices/filevideocapturer_unittest.cc",
+ "media/sctp/sctputils_unittest.cc",
"session/media/channel_unittest.cc",
"session/media/channelmanager_unittest.cc",
"session/media/currentspeakermonitor_unittest.cc",
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 7431bc1..f90265c 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -50,6 +50,10 @@
class Timing;
}
+namespace webrtc {
+struct DataChannelInit;
+}
+
namespace cricket {
class AudioRenderer;
@@ -980,6 +984,11 @@
// Signal when the media channel is ready to send the stream. Arguments are:
// writable(bool)
sigslot::signal1<bool> SignalReadyToSend;
+ // Signal for notifying when a new stream is added from the remote side. Used
+ // for the in-band negotioation through the OPEN message for SCTP data
+ // channel.
+ sigslot::signal2<const std::string&, const webrtc::DataChannelInit&>
+ SignalNewStreamReceived;
};
} // namespace cricket
diff --git a/talk/media/sctp/sctpdataengine.cc b/talk/media/sctp/sctpdataengine.cc
index 2e8e90c..2b86001 100644
--- a/talk/media/sctp/sctpdataengine.cc
+++ b/talk/media/sctp/sctpdataengine.cc
@@ -31,12 +31,14 @@
#include <stdio.h>
#include <vector>
+#include "talk/app/webrtc/datachannelinterface.h"
#include "talk/base/buffer.h"
#include "talk/base/helpers.h"
#include "talk/base/logging.h"
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
+#include "talk/media/sctp/sctputils.h"
#include "usrsctplib/usrsctp.h"
namespace cricket {
@@ -584,7 +586,23 @@
StreamParams found_stream;
if (!GetStreamBySsrc(streams_, params.ssrc, &found_stream)) {
if (params.type == DMT_CONTROL) {
- SignalDataReceived(params, buffer->data(), buffer->length());
+ std::string label;
+ webrtc::DataChannelInit config;
+ if (ParseDataChannelOpenMessage(*buffer, &label, &config)) {
+ config.id = params.ssrc;
+ // Do not send the OPEN message for this data channel.
+ config.negotiated = true;
+ SignalNewStreamReceived(label, config);
+
+ // Add the stream immediately.
+ cricket::StreamParams sparams =
+ cricket::StreamParams::CreateLegacy(params.ssrc);
+ AddSendStream(sparams);
+ AddRecvStream(sparams);
+ } else {
+ LOG(LS_ERROR) << debug_name_ << "->OnDataFromSctpToChannel(...): "
+ << "Received malformed control message";
+ }
} else {
LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): "
<< "Received packet for unknown ssrc: " << params.ssrc;
diff --git a/talk/media/sctp/sctpdataengine_unittest.cc b/talk/media/sctp/sctpdataengine_unittest.cc
index 363e7df..cab60b0 100644
--- a/talk/media/sctp/sctpdataengine_unittest.cc
+++ b/talk/media/sctp/sctpdataengine_unittest.cc
@@ -30,6 +30,7 @@
#include <stdio.h>
#include <string>
+#include "talk/app/webrtc/datachannelinterface.h"
#include "talk/base/buffer.h"
#include "talk/base/criticalsection.h"
#include "talk/base/gunit.h"
@@ -41,6 +42,7 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/sctp/sctpdataengine.h"
+#include "talk/media/sctp/sctputils.h"
enum {
MSG_PACKET = 1,
@@ -161,7 +163,8 @@
};
// SCTP Data Engine testing framework.
-class SctpDataMediaChannelTest : public testing::Test {
+class SctpDataMediaChannelTest : public testing::Test,
+ public sigslot::has_slots<> {
protected:
virtual void SetUp() {
engine_.reset(new cricket::SctpDataEngine());
@@ -212,6 +215,8 @@
// When data is received, pass it to the SctpFakeDataReceiver.
channel->SignalDataReceived.connect(
recv, &SctpFakeDataReceiver::OnDataReceived);
+ channel->SignalNewStreamReceived.connect(
+ this, &SctpDataMediaChannelTest::OnNewStreamReceived);
return channel;
}
@@ -246,6 +251,14 @@
SctpFakeDataReceiver* receiver1() { return recv1_.get(); }
SctpFakeDataReceiver* receiver2() { return recv2_.get(); }
+ void OnNewStreamReceived(const std::string& label,
+ const webrtc::DataChannelInit& init) {
+ last_label_ = label;
+ last_dc_init_ = init;
+ }
+ std::string last_label() { return last_label_; }
+ webrtc::DataChannelInit last_dc_init() { return last_dc_init_; }
+
private:
talk_base::scoped_ptr<cricket::SctpDataEngine> engine_;
talk_base::scoped_ptr<SctpFakeNetworkInterface> net1_;
@@ -254,6 +267,8 @@
talk_base::scoped_ptr<SctpFakeDataReceiver> recv2_;
talk_base::scoped_ptr<cricket::SctpDataMediaChannel> chan1_;
talk_base::scoped_ptr<cricket::SctpDataMediaChannel> chan2_;
+ std::string last_label_;
+ webrtc::DataChannelInit last_dc_init_;
};
// Verifies that SignalReadyToSend is fired.
@@ -316,3 +331,30 @@
channel2()->SetSend(false);
LOG(LS_VERBOSE) << "Cleaning up. -----------------------------";
}
+
+TEST_F(SctpDataMediaChannelTest, SendReceiveOpenMessage) {
+ SetupConnectedChannels();
+
+ std::string label("x");
+ webrtc::DataChannelInit config;
+ config.id = 10;
+
+ // Send the OPEN message on a unknown ssrc.
+ channel1()->AddSendStream(cricket::StreamParams::CreateLegacy(config.id));
+ cricket::SendDataParams params;
+ params.ssrc = config.id;
+ params.type = cricket::DMT_CONTROL;
+ cricket::SendDataResult result;
+ talk_base::Buffer buffer;
+ ASSERT_TRUE(cricket::WriteDataChannelOpenMessage(label, config, &buffer));
+ ASSERT_TRUE(channel1()->SendData(params, buffer, &result));
+ // Send data on the new ssrc immediately after sending the OPEN message.
+ ASSERT_TRUE(SendData(channel1(), config.id, "hi chan2", &result));
+
+ // Verifies the received OPEN message.
+ EXPECT_TRUE_WAIT(last_label() == label, 1000);
+ EXPECT_EQ(config.id, last_dc_init().id);
+ EXPECT_EQ(true, last_dc_init().negotiated);
+ // Verifies the received data.
+ EXPECT_TRUE_WAIT(ReceivedData(receiver2(), config.id, "hi chan2"), 1000);
+}
diff --git a/talk/media/sctp/sctputils.cc b/talk/media/sctp/sctputils.cc
new file mode 100644
index 0000000..c33c64e
--- /dev/null
+++ b/talk/media/sctp/sctputils.cc
@@ -0,0 +1,176 @@
+/*
+ * 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 "talk/media/sctp/sctputils.h"
+
+#include "talk/app/webrtc/datachannelinterface.h"
+#include "talk/base/buffer.h"
+#include "talk/base/bytebuffer.h"
+#include "talk/base/logging.h"
+
+namespace cricket {
+
+// Format defined at
+// http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
+
+static const uint8 DATA_CHANNEL_OPEN_MESSAGE_TYPE = 0x03;
+
+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 ParseDataChannelOpenMessage(
+ const talk_base::Buffer& payload,
+ std::string* label,
+ webrtc::DataChannelInit* config) {
+ // Format defined at
+ // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
+
+ talk_base::ByteBuffer buffer(payload.data(), payload.length());
+
+ uint8 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 channel_type;
+ if (!buffer.ReadUInt8(&channel_type)) {
+ LOG(LS_WARNING) << "Could not read OPEN message channel type.";
+ return false;
+ }
+ uint16 reliability_param;
+ if (!buffer.ReadUInt16(&reliability_param)) {
+ LOG(LS_WARNING) << "Could not read OPEN message reliabilility param.";
+ return false;
+ }
+ uint16 priority;
+ if (!buffer.ReadUInt16(&priority)) {
+ LOG(LS_WARNING) << "Could not read OPEN message reliabilility prioirty.";
+ return false;
+ }
+ uint16 label_length;
+ if (!buffer.ReadUInt16(&label_length)) {
+ LOG(LS_WARNING) << "Could not read OPEN message label length.";
+ return false;
+ }
+ uint16 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;
+
+ case DCOMCT_ORDERED_PARTIAL_TIME:
+ case DCOMCT_UNORDERED_PARTIAL_TIME:
+ config->maxRetransmitTime = reliability_param;
+ }
+
+ return true;
+}
+
+bool WriteDataChannelOpenMessage(
+ const std::string& label,
+ const webrtc::DataChannelInit& config,
+ talk_base::Buffer* payload) {
+ // Format defined at
+ // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
+ // TODO(pthatcher)
+
+ uint8 channel_type = 0;
+ uint16 reliability_param = 0;
+ uint16 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;
+ }
+ }
+
+ talk_base::ByteBuffer buffer(
+ NULL, 20 + label.length() + config.protocol.length(),
+ talk_base::ByteBuffer::ORDER_NETWORK);
+ buffer.WriteUInt8(DATA_CHANNEL_OPEN_MESSAGE_TYPE);
+ buffer.WriteUInt8(channel_type);
+ buffer.WriteUInt16(reliability_param);
+ buffer.WriteUInt16(priority);
+ buffer.WriteUInt16(static_cast<uint16>(label.length()));
+ buffer.WriteUInt16(static_cast<uint16>(config.protocol.length()));
+ buffer.WriteString(label);
+ buffer.WriteString(config.protocol);
+ payload->SetData(buffer.Data(), buffer.Length());
+ return true;
+}
+
+} // namespace cricket
diff --git a/talk/media/sctp/sctputils.h b/talk/media/sctp/sctputils.h
new file mode 100644
index 0000000..d349274
--- /dev/null
+++ b/talk/media/sctp/sctputils.h
@@ -0,0 +1,53 @@
+/*
+ * 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.
+ */
+
+#ifndef TALK_MEDIA_BASE_SCTPUTILS_H_
+#define TALK_MEDIA_BASE_SCTPUTILS_H_
+
+#include <string>
+
+namespace talk_base {
+class Buffer;
+} // namespace talk_base
+
+namespace webrtc {
+struct DataChannelInit;
+} // namespace webrtc
+
+namespace cricket {
+
+bool ParseDataChannelOpenMessage(const talk_base::Buffer& payload,
+ std::string* label,
+ webrtc::DataChannelInit* config);
+
+bool WriteDataChannelOpenMessage(const std::string& label,
+ const webrtc::DataChannelInit& config,
+ talk_base::Buffer* payload);
+
+} // namespace cricket
+
+#endif // TALK_MEDIA_BASE_SCTPUTILS_H_
diff --git a/talk/media/sctp/sctputils_unittest.cc b/talk/media/sctp/sctputils_unittest.cc
new file mode 100644
index 0000000..1cc9a70
--- /dev/null
+++ b/talk/media/sctp/sctputils_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ * 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 "talk/app/webrtc/datachannelinterface.h"
+#include "talk/base/bytebuffer.h"
+#include "talk/base/gunit.h"
+#include "talk/media/sctp/sctputils.h"
+
+class SctpUtilsTest : public testing::Test {
+ public:
+ void VerifyOpenMessageFormat(const talk_base::Buffer& packet,
+ const std::string& label,
+ const webrtc::DataChannelInit& config) {
+ uint8 message_type;
+ uint8 channel_type;
+ uint16 reliability;
+ uint16 priority;
+ uint16 label_length;
+ uint16 protocol_length;
+
+ talk_base::ByteBuffer buffer(packet.data(), packet.length());
+ ASSERT_TRUE(buffer.ReadUInt8(&message_type));
+ EXPECT_EQ(0x03, message_type);
+
+ ASSERT_TRUE(buffer.ReadUInt8(&channel_type));
+ if (config.ordered) {
+ EXPECT_EQ(config.maxRetransmits > -1 ?
+ 0x01 : (config.maxRetransmitTime > -1 ? 0x02 : 0),
+ channel_type);
+ } else {
+ EXPECT_EQ(config.maxRetransmits > -1 ?
+ 0x81 : (config.maxRetransmitTime > -1 ? 0x82 : 0x80),
+ channel_type);
+ }
+
+ ASSERT_TRUE(buffer.ReadUInt16(&reliability));
+ if (config.maxRetransmits > -1 || config.maxRetransmitTime > -1) {
+ EXPECT_EQ(config.maxRetransmits > -1 ?
+ config.maxRetransmits : config.maxRetransmitTime,
+ reliability);
+ }
+
+ ASSERT_TRUE(buffer.ReadUInt16(&priority));
+
+ ASSERT_TRUE(buffer.ReadUInt16(&label_length));
+ ASSERT_TRUE(buffer.ReadUInt16(&protocol_length));
+ EXPECT_EQ(label.size(), label_length);
+ EXPECT_EQ(config.protocol.size(), protocol_length);
+
+ std::string label_output;
+ ASSERT_TRUE(buffer.ReadString(&label_output, label_length));
+ EXPECT_EQ(label, label_output);
+ std::string protocol_output;
+ ASSERT_TRUE(buffer.ReadString(&protocol_output, protocol_length));
+ EXPECT_EQ(config.protocol, protocol_output);
+ }
+};
+
+TEST_F(SctpUtilsTest, WriteParseMessageWithOrderedReliable) {
+ std::string input_label = "abc";
+ webrtc::DataChannelInit config;
+ config.protocol = "y";
+
+ talk_base::Buffer packet;
+ ASSERT(cricket::WriteDataChannelOpenMessage(input_label, config, &packet));
+
+ VerifyOpenMessageFormat(packet, input_label, config);
+
+ std::string output_label;
+ webrtc::DataChannelInit output_config;
+ ASSERT(cricket::ParseDataChannelOpenMessage(
+ packet, &output_label, &output_config));
+
+ EXPECT_EQ(input_label, output_label);
+ EXPECT_EQ(config.protocol, output_config.protocol);
+ EXPECT_EQ(config.ordered, output_config.ordered);
+ EXPECT_EQ(config.maxRetransmitTime, output_config.maxRetransmitTime);
+ EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
+}
+
+TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmitTime) {
+ std::string input_label = "abc";
+ webrtc::DataChannelInit config;
+ config.ordered = false;
+ config.maxRetransmitTime = 10;
+ config.protocol = "y";
+
+ talk_base::Buffer packet;
+ ASSERT(cricket::WriteDataChannelOpenMessage(input_label, config, &packet));
+
+ VerifyOpenMessageFormat(packet, input_label, config);
+
+ std::string output_label;
+ webrtc::DataChannelInit output_config;
+ ASSERT(cricket::ParseDataChannelOpenMessage(
+ packet, &output_label, &output_config));
+
+ EXPECT_EQ(input_label, output_label);
+ EXPECT_EQ(config.protocol, output_config.protocol);
+ EXPECT_EQ(config.ordered, output_config.ordered);
+ EXPECT_EQ(config.maxRetransmitTime, output_config.maxRetransmitTime);
+}
+
+TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmits) {
+ std::string input_label = "abc";
+ webrtc::DataChannelInit config;
+ config.maxRetransmits = 10;
+ config.protocol = "y";
+
+ talk_base::Buffer packet;
+ ASSERT(cricket::WriteDataChannelOpenMessage(input_label, config, &packet));
+
+ VerifyOpenMessageFormat(packet, input_label, config);
+
+ std::string output_label;
+ webrtc::DataChannelInit output_config;
+ ASSERT(cricket::ParseDataChannelOpenMessage(
+ packet, &output_label, &output_config));
+
+ EXPECT_EQ(input_label, output_label);
+ EXPECT_EQ(config.protocol, output_config.protocol);
+ EXPECT_EQ(config.ordered, output_config.ordered);
+ EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
+}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5c16d6e..83cbdaf 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -204,11 +204,30 @@
}
return false;
}
+
static bool IsNackEnabled(const AudioCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
}
+// Gets the default set of options applied to the engine. Historically, these
+// were supplied as a combination of flags from the channel manager (ec, agc,
+// ns, and highpass) and the rest hardcoded in InitInternal.
+static AudioOptions GetDefaultEngineOptions() {
+ AudioOptions options;
+ options.echo_cancellation.Set(true);
+ options.auto_gain_control.Set(true);
+ options.noise_suppression.Set(true);
+ options.highpass_filter.Set(true);
+ options.stereo_swapping.Set(false);
+ options.typing_detection.Set(true);
+ options.conference_mode.Set(false);
+ options.adjust_agc_delta.Set(0);
+ options.experimental_agc.Set(false);
+ options.experimental_aec.Set(false);
+ options.aec_dump.Set(false);
+ return options;
+}
class WebRtcSoundclipMedia : public SoundclipMedia {
public:
@@ -353,6 +372,7 @@
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
kRtpAudioLevelHeaderExtensionId));
+ options_ = GetDefaultEngineOptions();
}
static bool IsOpus(const AudioCodec& codec) {
@@ -482,24 +502,6 @@
return res;
}
-// Gets the default set of optoins applied to the engine. Historically, these
-// were supplied as a combination of flags from the channel manager (ec, agc,
-// ns, and highpass) and the rest hardcoded in InitInternal.
-static AudioOptions GetDefaultEngineOptions() {
- AudioOptions options;
- options.echo_cancellation.Set(true);
- options.auto_gain_control.Set(true);
- options.noise_suppression.Set(true);
- options.highpass_filter.Set(true);
- options.typing_detection.Set(true);
- options.conference_mode.Set(false);
- options.adjust_agc_delta.Set(0);
- options.experimental_agc.Set(false);
- options.experimental_aec.Set(false);
- options.aec_dump.Set(false);
- return options;
-}
-
bool WebRtcVoiceEngine::InitInternal() {
// Temporarily turn logging level up for the Init call
int old_filter = log_filter_;
@@ -1524,6 +1526,9 @@
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->voe()->base()->CreateChannel()),
+ send_bw_setting_(false),
+ send_autobw_(false),
+ send_bw_bps_(0),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
@@ -1828,6 +1833,10 @@
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
+ if (send_bw_setting_) {
+ SetSendBandwidthInternal(send_autobw_, send_bw_bps_);
+ }
+
return true;
}
@@ -2731,9 +2740,20 @@
bool WebRtcVoiceMediaChannel::SetSendBandwidth(bool autobw, int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth.";
+ send_bw_setting_ = true;
+ send_autobw_ = autobw;
+ send_bw_bps_ = bps;
+
+ return SetSendBandwidthInternal(send_autobw_, send_bw_bps_);
+}
+
+bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(bool autobw, int bps) {
+ LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidthInternal.";
+
if (!send_codec_) {
- LOG(LS_INFO) << "The send codec has not been set up yet.";
- return false;
+ LOG(LS_INFO) << "The send codec has not been set up yet. "
+ << "The send bandwidth setting will be applied later.";
+ return true;
}
// Bandwidth is auto by default.
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 7809706..b8b50b0 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -392,12 +392,16 @@
return channel_id == voe_channel();
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
+ bool SetSendBandwidthInternal(bool autobw, int bps);
talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
talk_base::scoped_ptr<webrtc::CodecInst> send_codec_;
+ bool send_bw_setting_;
+ bool send_autobw_;
+ int send_bw_bps_;
AudioOptions options_;
bool dtmf_allowed_;
bool desired_playout_;
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 8dc0dff..acefc38 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -212,13 +212,10 @@
codecs.push_back(codec);
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
- webrtc::CodecInst temp_codec;
- EXPECT_FALSE(voe_.GetSendCodec(channel_num, temp_codec));
- EXPECT_EQ(default_bitrate, temp_codec.rate);
-
bool result = channel_->SetSendBandwidth(auto_bitrate, desired_bitrate);
EXPECT_EQ(expected_result, result);
+ webrtc::CodecInst temp_codec;
EXPECT_FALSE(voe_.GetSendCodec(channel_num, temp_codec));
if (result) {
@@ -589,7 +586,7 @@
TestSendBandwidth(kOpusCodec, 64000, true, 96000, true);
}
-TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthFixedMultiRate) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthFixedMultiRateAsCaller) {
EXPECT_TRUE(SetupEngine());
EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
@@ -606,6 +603,24 @@
TestSendBandwidth(kOpusCodec, 64000, false, 96000, true);
}
+TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthFixedMultiRateAsCallee) {
+ EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
+ channel_ = engine_.CreateChannel();
+ EXPECT_TRUE(channel_ != NULL);
+ EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
+
+ int desired_bitrate = 128000;
+ EXPECT_TRUE(channel_->SetSendBandwidth(false, desired_bitrate));
+
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrc1)));
+
+ int channel_num = voe_.GetLastChannel();
+ webrtc::CodecInst codec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
+ EXPECT_EQ(desired_bitrate, codec.rate);
+}
+
// Test that bitrate cannot be set for CBR codecs.
// Bitrate is ignored if it is higher than the fixed bitrate.
// Bitrate less then the fixed bitrate is an error.
@@ -2606,6 +2621,16 @@
EXPECT_FALSE(ns_enabled);
}
+TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
+ cricket::WebRtcVoiceEngine engine;
+ cricket::AudioOptions options = engine.GetOptions();
+ // The default options should have at least a few things set. We purposefully
+ // don't check the option values here, though.
+ EXPECT_TRUE(options.echo_cancellation.IsSet());
+ EXPECT_TRUE(options.auto_gain_control.IsSet());
+ EXPECT_TRUE(options.noise_suppression.IsSet());
+}
+
// Test that GetReceiveChannelNum returns the default channel for the first
// recv stream in 1-1 calls.
TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelNumIn1To1Calls) {
diff --git a/talk/p2p/base/fakesession.h b/talk/p2p/base/fakesession.h
index 9a8fada..6a8ab4b 100644
--- a/talk/p2p/base/fakesession.h
+++ b/talk/p2p/base/fakesession.h
@@ -424,7 +424,7 @@
NULL, "", "", initiator),
fail_create_channel_(false) {
}
- FakeSession(bool initiator, talk_base::Thread* worker_thread)
+ FakeSession(talk_base::Thread* worker_thread, bool initiator)
: BaseSession(talk_base::Thread::Current(),
worker_thread,
NULL, "", "", initiator),
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 1cb5992..db169be 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -81,6 +81,7 @@
MSG_SETSCREENCASTFACTORY,
MSG_FIRSTPACKETRECEIVED,
MSG_SESSION_ERROR,
+ MSG_NEWSTREAMRECEIVED,
};
// Value specified in RFC 5764.
@@ -2500,6 +2501,8 @@
this, &DataChannel::OnDataChannelError);
media_channel()->SignalReadyToSend.connect(
this, &DataChannel::OnDataChannelReadyToSend);
+ media_channel()->SignalNewStreamReceived.connect(
+ this, &DataChannel::OnDataChannelNewStreamReceived);
srtp_filter()->SignalSrtpError.connect(
this, &DataChannel::OnSrtpError);
return true;
@@ -2722,6 +2725,13 @@
delete data;
break;
}
+ case MSG_NEWSTREAMRECEIVED: {
+ DataChannelNewStreamReceivedMessageData* data =
+ static_cast<DataChannelNewStreamReceivedMessageData*>(pmsg->pdata);
+ SignalNewStreamReceived(data->label, data->init);
+ delete data;
+ break;
+ }
default:
BaseChannel::OnMessage(pmsg);
break;
@@ -2777,6 +2787,14 @@
new DataChannelReadyToSendMessageData(writable));
}
+void DataChannel::OnDataChannelNewStreamReceived(
+ const std::string& label, const webrtc::DataChannelInit& init) {
+ signaling_thread()->Post(
+ this,
+ MSG_NEWSTREAMRECEIVED,
+ new DataChannelNewStreamReceivedMessageData(label, init));
+}
+
void DataChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
SrtpFilter::Error error) {
switch (error) {
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 6ee2e8cd..3a627ed 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -31,6 +31,7 @@
#include <string>
#include <vector>
+#include "talk/app/webrtc/datachannelinterface.h"
#include "talk/base/asyncudpsocket.h"
#include "talk/base/criticalsection.h"
#include "talk/base/network.h"
@@ -627,6 +628,11 @@
// That occurs when the channel is enabled, the transport is writable,
// both local and remote descriptions are set, and the channel is unblocked.
sigslot::signal1<bool> SignalReadyToSendData;
+ // Signal for notifying when a new stream is added from the remote side. Used
+ // for the in-band negotioation through the OPEN message for SCTP data
+ // channel.
+ sigslot::signal2<const std::string&, const webrtc::DataChannelInit&>
+ SignalNewStreamReceived;
protected:
// downcasts a MediaChannel.
@@ -666,6 +672,17 @@
typedef talk_base::TypedMessageData<bool> DataChannelReadyToSendMessageData;
+ struct DataChannelNewStreamReceivedMessageData
+ : public talk_base::MessageData {
+ DataChannelNewStreamReceivedMessageData(
+ const std::string& label, const webrtc::DataChannelInit& init)
+ : label(label),
+ init(init) {
+ }
+ const std::string label;
+ const webrtc::DataChannelInit init;
+ };
+
// overrides from BaseChannel
virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
// If data_channel_type_ is DCT_NONE, set it. Otherwise, check that
@@ -694,6 +711,8 @@
const ReceiveDataParams& params, const char* data, size_t len);
void OnDataChannelError(uint32 ssrc, DataMediaChannel::Error error);
void OnDataChannelReadyToSend(bool writable);
+ void OnDataChannelNewStreamReceived(const std::string& label,
+ const webrtc::DataChannelInit& init);
void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
talk_base::scoped_ptr<DataMediaMonitor> media_monitor_;
diff --git a/talk/session/media/channelmanager_unittest.cc b/talk/session/media/channelmanager_unittest.cc
index d0ba8b9..d0d380d 100644
--- a/talk/session/media/channelmanager_unittest.cc
+++ b/talk/session/media/channelmanager_unittest.cc
@@ -154,7 +154,7 @@
EXPECT_TRUE(cm_->set_worker_thread(&worker_));
EXPECT_TRUE(cm_->Init());
delete session_;
- session_ = new cricket::FakeSession(true, &worker_);
+ session_ = new cricket::FakeSession(&worker_, true);
cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
session_, cricket::CN_AUDIO, false);
EXPECT_TRUE(voice_channel != NULL);
