Update stable to r5065.
git-svn-id: http://webrtc.googlecode.com/svn/stable/talk@5066 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index 97d5cf6..ba6b09b 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -110,6 +110,8 @@
// TODO(perkj): Remove kEnableSctpDataChannels once Chrome use
// PeerConnectionFactory::SetOptions.
static const char kEnableSctpDataChannels[]; // Enable SCTP DataChannels
+ // Temporary pseudo-constraint for enabling DSCP through JS.
+ static const char kEnableDscp[];
// The prefix of internal-only constraints whose JS set values should be
// stripped by Chrome before passed down to Libjingle.
diff --git a/app/webrtc/webrtcsession.cc b/app/webrtc/webrtcsession.cc
index 3f14268..a96c2da 100644
--- a/app/webrtc/webrtcsession.cc
+++ b/app/webrtc/webrtcsession.cc
@@ -57,6 +57,8 @@
const char MediaConstraintsInterface::kInternalConstraintPrefix[] = "internal";
// Supported MediaConstraints.
+// DSCP constraints.
+const char MediaConstraintsInterface::kEnableDscp[] = "googDscp";
// DTLS-SRTP pseudo-constraints.
const char MediaConstraintsInterface::kEnableDtlsSrtp[] =
"DtlsSrtpKeyAgreement";
@@ -430,6 +432,7 @@
ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
older_version_remote_peer_(false),
dtls_enabled_(false),
+ dscp_enabled_(false),
data_channel_type_(cricket::DCT_NONE),
ice_restart_latch_(new IceRestartAnswerLatch) {
}
@@ -491,6 +494,14 @@
mediastream_signaling_->SetDataChannelFactory(this);
}
+ // Find DSCP constraint.
+ if (FindConstraint(
+ constraints,
+ MediaConstraintsInterface::kEnableDscp,
+ &value, NULL)) {
+ dscp_enabled_ = value;
+ }
+
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,
@@ -1357,13 +1368,29 @@
bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
voice_channel_.reset(channel_manager_->CreateVoiceChannel(
this, content->name, true));
- return (voice_channel_ != NULL);
+ if (!voice_channel_.get())
+ return false;
+
+ if (dscp_enabled_) {
+ cricket::AudioOptions options;
+ options.dscp.Set(true);
+ voice_channel_->SetChannelOptions(options);
+ }
+ return true;
}
bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content) {
video_channel_.reset(channel_manager_->CreateVideoChannel(
this, content->name, true, voice_channel_.get()));
- return (video_channel_ != NULL);
+ if (!video_channel_.get())
+ return false;
+
+ if (dscp_enabled_) {
+ cricket::VideoOptions options;
+ options.dscp.Set(true);
+ video_channel_->SetChannelOptions(options);
+ }
+ return true;
}
bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
diff --git a/app/webrtc/webrtcsession.h b/app/webrtc/webrtcsession.h
index c4f6055..893ab3f 100644
--- a/app/webrtc/webrtcsession.h
+++ b/app/webrtc/webrtcsession.h
@@ -313,6 +313,8 @@
// If the remote peer is using a older version of implementation.
bool older_version_remote_peer_;
bool dtls_enabled_;
+ // Flag will be set based on the constraint value.
+ bool dscp_enabled_;
// Specifies which kind of data channel is allowed. This is controlled
// by the chrome command-line flag and constraints:
// 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
diff --git a/app/webrtc/webrtcsession_unittest.cc b/app/webrtc/webrtcsession_unittest.cc
index ecbba18..0ddc16c 100644
--- a/app/webrtc/webrtcsession_unittest.cc
+++ b/app/webrtc/webrtcsession_unittest.cc
@@ -2791,6 +2791,32 @@
offer);
}
+// This test verifies DSCP is properly applied on the media channels.
+TEST_F(WebRtcSessionTest, TestDscpConstraint) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kEnableDscp, true);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer(NULL);
+
+ SetLocalDescriptionWithoutError(offer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(video_channel_ != NULL);
+ ASSERT_TRUE(voice_channel_ != NULL);
+ cricket::AudioOptions audio_options;
+ EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
+ cricket::VideoOptions video_options;
+ EXPECT_TRUE(video_channel_->GetOptions(&video_options));
+ EXPECT_TRUE(audio_options.dscp.IsSet());
+ EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
+ EXPECT_TRUE(video_options.dscp.IsSet());
+ EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
+}
+
// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
// currently fails because upon disconnection and reconnection OnIceComplete is
// called more than once without returning to IceGatheringGathering.
diff --git a/media/base/fakenetworkinterface.h b/media/base/fakenetworkinterface.h
index d0f277e..37679eb 100644
--- a/media/base/fakenetworkinterface.h
+++ b/media/base/fakenetworkinterface.h
@@ -34,6 +34,7 @@
#include "talk/base/buffer.h"
#include "talk/base/byteorder.h"
#include "talk/base/criticalsection.h"
+#include "talk/base/dscp.h"
#include "talk/base/messagehandler.h"
#include "talk/base/messagequeue.h"
#include "talk/base/thread.h"
@@ -51,7 +52,8 @@
dest_(NULL),
conf_(false),
sendbuf_size_(-1),
- recvbuf_size_(-1) {
+ recvbuf_size_(-1),
+ dscp_(talk_base::DSCP_NO_CHANGE) {
}
void SetDestination(MediaChannel* dest) { dest_ = dest; }
@@ -128,6 +130,7 @@
int sendbuf_size() const { return sendbuf_size_; }
int recvbuf_size() const { return recvbuf_size_; }
+ talk_base::DiffServCodePoint dscp() const { return dscp_; }
protected:
virtual bool SendPacket(talk_base::Buffer* packet,
@@ -182,6 +185,8 @@
sendbuf_size_ = option;
} else if (opt == talk_base::Socket::OPT_RCVBUF) {
recvbuf_size_ = option;
+ } else if (opt == talk_base::Socket::OPT_DSCP) {
+ dscp_ = static_cast<talk_base::DiffServCodePoint>(option);
}
return 0;
}
@@ -244,6 +249,7 @@
std::vector<talk_base::Buffer> rtcp_packets_;
int sendbuf_size_;
int recvbuf_size_;
+ talk_base::DiffServCodePoint dscp_;
};
} // namespace cricket
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 49b3336..919248f 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -183,6 +183,7 @@
rx_agc_limiter.SetFrom(change.rx_agc_limiter);
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
+ dscp.SetFrom(change.dscp);
}
bool operator==(const AudioOptions& o) const {
@@ -206,7 +207,8 @@
rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
- playout_sample_rate == o.playout_sample_rate;
+ playout_sample_rate == o.playout_sample_rate &&
+ dscp == o.dscp;
}
std::string ToString() const {
@@ -235,6 +237,7 @@
ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
+ ost << ToStringIfSet("dscp", dscp);
ost << "}";
return ost.str();
}
@@ -269,6 +272,8 @@
Settable<bool> rx_agc_limiter;
Settable<uint32> recording_sample_rate;
Settable<uint32> playout_sample_rate;
+ // Set DSCP value for packet sent from audio channel.
+ Settable<bool> dscp;
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
@@ -307,6 +312,7 @@
change.system_high_adaptation_threshhold);
buffered_mode_latency.SetFrom(change.buffered_mode_latency);
lower_min_bitrate.SetFrom(change.lower_min_bitrate);
+ dscp.SetFrom(change.dscp);
}
bool operator==(const VideoOptions& o) const {
@@ -331,7 +337,8 @@
system_high_adaptation_threshhold ==
o.system_high_adaptation_threshhold &&
buffered_mode_latency == o.buffered_mode_latency &&
- lower_min_bitrate == o.lower_min_bitrate;
+ lower_min_bitrate == o.lower_min_bitrate &&
+ dscp == o.dscp;
}
std::string ToString() const {
@@ -359,6 +366,7 @@
ost << ToStringIfSet("high", system_high_adaptation_threshhold);
ost << ToStringIfSet("buffered mode latency", buffered_mode_latency);
ost << ToStringIfSet("lower min bitrate", lower_min_bitrate);
+ ost << ToStringIfSet("dscp", dscp);
ost << "}";
return ost.str();
}
@@ -405,6 +413,8 @@
Settable<int> buffered_mode_latency;
// Make minimum configured send bitrate even lower than usual, at 30kbit.
Settable<bool> lower_min_bitrate;
+ // Set DSCP value for packet sent from video channel.
+ Settable<bool> dscp;
};
// A class for playing out soundclips.
@@ -543,6 +553,21 @@
return network_interface_->SetOption(type, opt, option);
}
+ protected:
+ // This method sets DSCP |value| on both RTP and RTCP channels.
+ int SetDscp(talk_base::DiffServCodePoint value) {
+ int ret;
+ ret = SetOption(NetworkInterface::ST_RTP,
+ talk_base::Socket::OPT_DSCP,
+ value);
+ if (ret == 0) {
+ ret = SetOption(NetworkInterface::ST_RTCP,
+ talk_base::Socket::OPT_DSCP,
+ value);
+ }
+ return ret;
+ }
+
private:
bool DoSendPacket(talk_base::Buffer* packet, bool rtcp) {
talk_base::CritScope cs(&network_interface_crit_);
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index f2827db..6067507 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -156,6 +156,11 @@
static const char kRtpAbsoluteSendTimeHeaderExtension[] =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
static const int kRtpAbsoluteSendTimeExtensionId = 3;
+// Default video dscp value.
+// See http://tools.ietf.org/html/rfc2474 for details
+// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
+static const talk_base::DiffServCodePoint kVideoDscpValue =
+ talk_base::DSCP_AF41;
static bool IsNackEnabled(const VideoCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
@@ -2612,6 +2617,8 @@
bool cpu_overuse_detection_changed = options.cpu_overuse_detection.IsSet() &&
(options_.cpu_overuse_detection != options.cpu_overuse_detection);
+ bool dscp_option_changed = (options_.dscp != options.dscp);
+
bool conference_mode_turned_off = false;
if (options_.conference_mode.IsSet() && options.conference_mode.IsSet() &&
options_.conference_mode.GetWithDefaultIfUnset(false) &&
@@ -2710,6 +2717,14 @@
send_channel->SetCpuOveruseDetection(cpu_overuse_detection);
}
}
+ if (dscp_option_changed) {
+ talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
+ if (options.dscp.GetWithDefaultIfUnset(false))
+ dscp = kVideoDscpValue;
+ if (MediaChannel::SetDscp(dscp) != 0) {
+ LOG(LS_WARNING) << "Failed to set DSCP settings for video channel";
+ }
+ }
return true;
}
diff --git a/media/webrtc/webrtcvideoengine_unittest.cc b/media/webrtc/webrtcvideoengine_unittest.cc
index 0b8bdda..05fac8e 100644
--- a/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/media/webrtc/webrtcvideoengine_unittest.cc
@@ -1508,6 +1508,21 @@
Base::AddRemoveCapturerMultipleSources();
}
+// This test verifies DSCP settings are properly applied on video media channel.
+TEST_F(WebRtcVideoMediaChannelTest, TestSetDscpOptions) {
+ talk_base::scoped_ptr<cricket::FakeNetworkInterface> network_interface(
+ new cricket::FakeNetworkInterface);
+ channel_->SetInterface(network_interface.get());
+ cricket::VideoOptions options;
+ options.dscp.Set(true);
+ EXPECT_TRUE(channel_->SetOptions(options));
+ EXPECT_EQ(talk_base::DSCP_AF41, network_interface->dscp());
+ options.dscp.Set(false);
+ EXPECT_TRUE(channel_->SetOptions(options));
+ EXPECT_EQ(talk_base::DSCP_DEFAULT, network_interface->dscp());
+ channel_->SetInterface(NULL);
+}
+
TEST_F(WebRtcVideoMediaChannelTest, SetOptionsSucceedsWhenSending) {
cricket::VideoOptions options;
diff --git a/media/webrtc/webrtcvideoframe.cc b/media/webrtc/webrtcvideoframe.cc
index 584aac0..0938ae6 100644
--- a/media/webrtc/webrtcvideoframe.cc
+++ b/media/webrtc/webrtcvideoframe.cc
@@ -42,45 +42,38 @@
static const int kWatermarkOffsetFromBottom = 8;
static const unsigned char kWatermarkMaxYValue = 64;
-FrameBuffer::FrameBuffer() : length_(0) {}
+FrameBuffer::FrameBuffer() {}
-FrameBuffer::FrameBuffer(size_t length) : length_(0) {
+FrameBuffer::FrameBuffer(size_t length) {
char* buffer = new char[length];
SetData(buffer, length);
}
-FrameBuffer::~FrameBuffer() {
- // Make sure that the video_frame_ doesn't delete the buffer as it may be
- // shared between multiple WebRtcVideoFrame.
- uint8_t* new_memory = NULL;
- uint32_t new_length = 0;
- uint32_t new_size = 0;
- video_frame_.Swap(new_memory, new_length, new_size);
-}
+FrameBuffer::~FrameBuffer() {}
void FrameBuffer::SetData(char* data, size_t length) {
- data_.reset(data);
- length_ = length;
uint8_t* new_memory = reinterpret_cast<uint8_t*>(data);
- uint32_t new_length = static_cast<int>(length);
- uint32_t new_size = static_cast<int>(length);
+ uint32_t new_length = static_cast<uint32_t>(length);
+ uint32_t new_size = static_cast<uint32_t>(length);
video_frame_.Swap(new_memory, new_length, new_size);
}
void FrameBuffer::ReturnData(char** data, size_t* length) {
- uint8_t* old_memory = NULL;
+ *data = NULL;
uint32_t old_length = 0;
uint32_t old_size = 0;
- video_frame_.Swap(old_memory, old_length, old_size);
- data_.release();
- length_ = 0;
+ video_frame_.Swap(reinterpret_cast<uint8_t*&>(*data),
+ old_length, old_size);
*length = old_length;
- *data = reinterpret_cast<char*>(old_memory);
}
-char* FrameBuffer::data() { return data_.get(); }
+char* FrameBuffer::data() {
+ return reinterpret_cast<char*>(video_frame_.Buffer());
+}
-size_t FrameBuffer::length() const { return length_; }
+size_t FrameBuffer::length() const {
+ return static_cast<size_t>(video_frame_.Length());
+}
webrtc::VideoFrame* FrameBuffer::frame() { return &video_frame_; }
diff --git a/media/webrtc/webrtcvideoframe.h b/media/webrtc/webrtcvideoframe.h
index e023234..ce13170 100644
--- a/media/webrtc/webrtcvideoframe.h
+++ b/media/webrtc/webrtcvideoframe.h
@@ -55,8 +55,6 @@
const webrtc::VideoFrame* frame() const;
private:
- talk_base::scoped_ptr<char[]> data_;
- size_t length_;
webrtc::VideoFrame video_frame_;
};
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 7f06009..ac8e067 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -125,6 +125,10 @@
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
+// Default audio dscp value.
+// See http://tools.ietf.org/html/rfc2474 for details.
+// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
+static const talk_base::DiffServCodePoint kAudioDscpValue = talk_base::DSCP_EF;
// Ensure we open the file in a writeable path on ChromeOS and Android. This
// workaround can be removed when it's possible to specify a filename for audio
@@ -1637,6 +1641,9 @@
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
+ // Check if DSCP value is changed from previous.
+ bool dscp_option_changed = (options_.dscp != options.dscp);
+
// TODO(xians): Add support to set different options for different send
// streams after we support multiple APMs.
@@ -1704,6 +1711,14 @@
return false;
}
}
+ if (dscp_option_changed) {
+ talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
+ if (options.dscp.GetWithDefaultIfUnset(false))
+ dscp = kAudioDscpValue;
+ if (MediaChannel::SetDscp(dscp) != 0) {
+ LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
+ }
+ }
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 2e52c8f..d0b06bb 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -12,6 +12,7 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/fakemediaprocessor.h"
+#include "talk/media/base/fakenetworkinterface.h"
#include "talk/media/base/fakertp.h"
#include "talk/media/webrtc/fakewebrtcvoiceengine.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
@@ -2703,6 +2704,23 @@
EXPECT_FALSE(ns_enabled);
}
+// This test verifies DSCP settings are properly applied on voice media channel.
+TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
+ EXPECT_TRUE(SetupEngine());
+ talk_base::scoped_ptr<cricket::VoiceMediaChannel> channel(
+ engine_.CreateChannel());
+ talk_base::scoped_ptr<cricket::FakeNetworkInterface> network_interface(
+ new cricket::FakeNetworkInterface);
+ channel->SetInterface(network_interface.get());
+ cricket::AudioOptions options;
+ options.dscp.Set(true);
+ EXPECT_TRUE(channel->SetOptions(options));
+ EXPECT_EQ(talk_base::DSCP_EF, network_interface->dscp());
+ options.dscp.Set(false);
+ EXPECT_TRUE(channel->SetOptions(options));
+ EXPECT_EQ(talk_base::DSCP_DEFAULT, network_interface->dscp());
+}
+
TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
cricket::WebRtcVoiceEngine engine;
cricket::AudioOptions options = engine.GetOptions();
diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc
index 8bdaa9a..e8f53ad 100644
--- a/p2p/base/p2ptransportchannel.cc
+++ b/p2p/base/p2ptransportchannel.cc
@@ -800,6 +800,7 @@
error_ = EWOULDBLOCK;
return -1;
}
+
int sent = best_connection_->Send(data, len, dscp);
if (sent <= 0) {
ASSERT(sent < 0);
diff --git a/session/media/channel.cc b/session/media/channel.cc
index 9c9847f..1a3063d 100644
--- a/session/media/channel.cc
+++ b/session/media/channel.cc
@@ -439,7 +439,6 @@
return false;
}
- media_channel_->SetInterface(this);
transport_channel_->SignalWritableState.connect(
this, &BaseChannel::OnWritableState);
transport_channel_->SignalReadPacket.connect(
@@ -453,6 +452,9 @@
this, &BaseChannel::OnNewRemoteDescription);
set_rtcp_transport_channel(rtcp_transport_channel);
+ // Both RTP and RTCP channels are set, we can call SetInterface on
+ // media channel and it can set network options.
+ media_channel_->SetInterface(this);
return true;
}
