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Update talk to 54898858.

Browse Source 
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
This commit is contained in:
wu@webrtc.org 2013-10-16 18:12:02 +00:00
parent 83e9c89eeb
commit 1d1ffc9ad2
19 changed files with 575 additions and 201 deletions
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138
talk/app/webrtc/mediastreamsignaling.cc
View File

@ -916,144 +916,6 @@ void MediaStreamSignaling::CreateRemoteDataChannel(const std::string& label,
}
// 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) {

6
talk/app/webrtc/mediastreamsignaling.h
View File

@ -196,12 +196,6 @@ class MediaStreamSignaling {
// 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.

32
talk/app/webrtc/webrtcsession.cc
View File

@ -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 @@ talk_base::scoped_refptr<DataChannel> WebRtcSession::CreateDataChannel(
}
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 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
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 @@ void WebRtcSession::CopySavedCandidates(
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;
}

6
talk/app/webrtc/webrtcsession.h
View File

@ -271,10 +271,8 @@ class WebRtcSession : public cricket::BaseSession,
// 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);

2
talk/libjingle.gyp
View File

@ -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',

2
talk/libjingle.scons
View File

@ -290,6 +290,7 @@ talk.Library(env, name = "jingle",
"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 @@ talk.Unittest(env, name = "media",
"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",

9
talk/media/base/mediachannel.h
View File

@ -50,6 +50,10 @@ class RateLimiter;
class Timing;
}
namespace webrtc {
struct DataChannelInit;
}
namespace cricket {
class AudioRenderer;
@ -980,6 +984,11 @@ class DataMediaChannel : public MediaChannel {
// 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

20
talk/media/sctp/sctpdataengine.cc
View File

@ -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 @@ void SctpDataMediaChannel::OnDataFromSctpToChannel(
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;

44
talk/media/sctp/sctpdataengine_unittest.cc
View File

@ -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 @@ class SignalReadyToSendObserver : public sigslot::has_slots<> {
};
// 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 @@ class SctpDataMediaChannelTest : public testing::Test {
// 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 @@ class SctpDataMediaChannelTest : public testing::Test {
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 @@ class SctpDataMediaChannelTest : public testing::Test {
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 @@ TEST_F(SctpDataMediaChannelTest, SendData) {
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);
}

176
talk/media/sctp/sctputils.cc Normal file
View File

@ -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

53
talk/media/sctp/sctputils.h Normal file
View File

@ -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_

148
talk/media/sctp/sctputils_unittest.cc Normal file
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@ -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);
}

60
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -204,11 +204,30 @@ static bool FindCodec(const std::vector<AudioCodec>& codecs,
}
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 @@ void WebRtcVoiceEngine::Construct() {
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
kRtpAudioLevelHeaderExtensionId));
options_ = GetDefaultEngineOptions();
}
static bool IsOpus(const AudioCodec& codec) {
@ -482,24 +502,6 @@ bool WebRtcVoiceEngine::Init(talk_base::Thread* worker_thread) {
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 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: 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 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// 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::MuteStream(uint32 ssrc, bool muted) {
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.

4
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -392,12 +392,16 @@ class WebRtcVoiceMediaChannel
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_;

35
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -212,13 +212,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthFixedMultiRate) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
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) {

2
talk/p2p/base/fakesession.h
View File

@ -424,7 +424,7 @@ class FakeSession : public BaseSession {
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),

18
talk/session/media/channel.cc
View File

@ -81,6 +81,7 @@ enum {
MSG_SETSCREENCASTFACTORY,
MSG_FIRSTPACKETRECEIVED,
MSG_SESSION_ERROR,
MSG_NEWSTREAMRECEIVED,
};
// Value specified in RFC 5764.
@ -2500,6 +2501,8 @@ bool DataChannel::Init() {
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 @@ void DataChannel::OnMessage(talk_base::Message *pmsg) {
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 @@ void DataChannel::OnDataChannelReadyToSend(bool writable) {
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) {

19
talk/session/media/channel.h
View File

@ -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 @@ class DataChannel : public BaseChannel {
// 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 @@ class DataChannel : public BaseChannel {
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 @@ class DataChannel : public BaseChannel {
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_;

2
talk/session/media/channelmanager_unittest.cc
View File

@ -154,7 +154,7 @@ TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
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);