Merge from Chromium at DEPS revision 278856
This commit was generated by merge_to_master.py.
Change-Id: I7dc08db577ee87432079e1e7508ea87dd8d0bc6c
diff --git a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
index b171b58..240e996 100644
--- a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
+++ b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
@@ -525,7 +525,7 @@
private void doTest() throws Exception {
CountDownLatch testDone = new CountDownLatch(1);
System.gc(); // Encourage any GC-related threads to start up.
- //TreeSet<String> threadsBeforeTest = allThreads();
+ TreeSet<String> threadsBeforeTest = allThreads();
PeerConnectionFactory factory = new PeerConnectionFactory();
// Uncomment to get ALL WebRTC tracing and SENSITIVE libjingle logging.
@@ -742,11 +742,8 @@
factory.dispose();
System.gc();
- // TODO(ldixon): the usrsctp threads are not cleaned up (issue 2749) and
- // caused the assert to fail. We should reenable the assert once issue 2749
- // is fixed.
- //TreeSet<String> threadsAfterTest = allThreads();
- //assertEquals(threadsBeforeTest, threadsAfterTest);
+ TreeSet<String> threadsAfterTest = allThreads();
+ assertEquals(threadsBeforeTest, threadsAfterTest);
Thread.sleep(100);
}
diff --git a/app/webrtc/mediaconstraintsinterface.cc b/app/webrtc/mediaconstraintsinterface.cc
index b19604b..0ecadd6 100644
--- a/app/webrtc/mediaconstraintsinterface.cc
+++ b/app/webrtc/mediaconstraintsinterface.cc
@@ -118,6 +118,9 @@
"googHighBitrate";
const char MediaConstraintsInterface::kVeryHighBitrate[] =
"googVeryHighBitrate";
+const char MediaConstraintsInterface::kPayloadPadding[] = "googPayloadPadding";
+const char MediaConstraintsInterface::kOpusFec[] = "googOpusFec";
+
// Set |value| to the value associated with the first appearance of |key|, or
// return false if |key| is not found.
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index 36cf20d..36257db 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -132,6 +132,12 @@
static const char kHighStartBitrate[]; // googHighStartBitrate
static const char kHighBitrate[]; // googHighBitrate
static const char kVeryHighBitrate[]; // googVeryHighBitrate
+ static const char kPayloadPadding[]; // googPayloadPadding
+
+ // PeerConnection codec constraint keys. This should be combined with the
+ // values above.
+ // kOpusFec controls whether we ask the other side to turn on FEC for Opus.
+ static const char kOpusFec[]; // googOpusFec
// 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/mediastreamsignaling.cc b/app/webrtc/mediastreamsignaling.cc
index c7fa673..99f627a 100644
--- a/app/webrtc/mediastreamsignaling.cc
+++ b/app/webrtc/mediastreamsignaling.cc
@@ -274,6 +274,23 @@
return true;
}
+void MediaStreamSignaling::RemoveSctpDataChannel(int sid) {
+ for (SctpDataChannels::iterator iter = sctp_data_channels_.begin();
+ iter != sctp_data_channels_.end();
+ ++iter) {
+ if ((*iter)->id() == sid) {
+ sctp_data_channels_.erase(iter);
+
+ if (talk_base::IsEven(sid) && sid <= last_allocated_sctp_even_sid_) {
+ last_allocated_sctp_even_sid_ = sid - 2;
+ } else if (talk_base::IsOdd(sid) && sid <= last_allocated_sctp_odd_sid_) {
+ last_allocated_sctp_odd_sid_ = sid - 2;
+ }
+ return;
+ }
+ }
+}
+
bool MediaStreamSignaling::AddLocalStream(MediaStreamInterface* local_stream) {
if (local_streams_->find(local_stream->label()) != NULL) {
LOG(LS_WARNING) << "MediaStream with label " << local_stream->label()
@@ -481,12 +498,19 @@
}
void MediaStreamSignaling::OnDataChannelClose() {
- RtpDataChannels::iterator it1 = rtp_data_channels_.begin();
- for (; it1 != rtp_data_channels_.end(); ++it1) {
+ // Use a temporary copy of the RTP/SCTP DataChannel list because the
+ // DataChannel may callback to us and try to modify the list.
+ RtpDataChannels temp_rtp_dcs;
+ temp_rtp_dcs.swap(rtp_data_channels_);
+ RtpDataChannels::iterator it1 = temp_rtp_dcs.begin();
+ for (; it1 != temp_rtp_dcs.end(); ++it1) {
it1->second->OnDataEngineClose();
}
- SctpDataChannels::iterator it2 = sctp_data_channels_.begin();
- for (; it2 != sctp_data_channels_.end(); ++it2) {
+
+ SctpDataChannels temp_sctp_dcs;
+ temp_sctp_dcs.swap(sctp_data_channels_);
+ SctpDataChannels::iterator it2 = temp_sctp_dcs.begin();
+ for (; it2 != temp_sctp_dcs.end(); ++it2) {
(*it2)->OnDataEngineClose();
}
}
diff --git a/app/webrtc/mediastreamsignaling.h b/app/webrtc/mediastreamsignaling.h
index 8051289..7378166 100644
--- a/app/webrtc/mediastreamsignaling.h
+++ b/app/webrtc/mediastreamsignaling.h
@@ -198,6 +198,7 @@
// After we receive an OPEN message, create a data channel and add it.
bool AddDataChannelFromOpenMessage(const cricket::ReceiveDataParams& params,
const talk_base::Buffer& payload);
+ void RemoveSctpDataChannel(int sid);
// Returns a MediaSessionOptions struct with options decided by |constraints|,
// the local MediaStreams and DataChannels.
diff --git a/app/webrtc/mediastreamsignaling_unittest.cc b/app/webrtc/mediastreamsignaling_unittest.cc
index 14b68e9..150058e 100644
--- a/app/webrtc/mediastreamsignaling_unittest.cc
+++ b/app/webrtc/mediastreamsignaling_unittest.cc
@@ -1141,6 +1141,47 @@
EXPECT_NE(old_id, new_id);
}
+// Verifies that SCTP ids of removed DataChannels can be reused.
+TEST_F(MediaStreamSignalingTest, SctpIdReusedForRemovedDataChannel) {
+ int odd_id = 1;
+ int even_id = 0;
+ AddDataChannel(cricket::DCT_SCTP, "a", odd_id);
+ AddDataChannel(cricket::DCT_SCTP, "a", even_id);
+
+ int allocated_id = -1;
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_SERVER,
+ &allocated_id));
+ EXPECT_EQ(odd_id + 2, allocated_id);
+ AddDataChannel(cricket::DCT_SCTP, "a", allocated_id);
+
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_CLIENT,
+ &allocated_id));
+ EXPECT_EQ(even_id + 2, allocated_id);
+ AddDataChannel(cricket::DCT_SCTP, "a", allocated_id);
+
+ signaling_->RemoveSctpDataChannel(odd_id);
+ signaling_->RemoveSctpDataChannel(even_id);
+
+ // Verifies that removed DataChannel ids are reused.
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_SERVER,
+ &allocated_id));
+ EXPECT_EQ(odd_id, allocated_id);
+
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_CLIENT,
+ &allocated_id));
+ EXPECT_EQ(even_id, allocated_id);
+
+ // Verifies that used higher DataChannel ids are not reused.
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_SERVER,
+ &allocated_id));
+ EXPECT_NE(odd_id + 2, allocated_id);
+
+ ASSERT_TRUE(signaling_->AllocateSctpSid(talk_base::SSL_CLIENT,
+ &allocated_id));
+ EXPECT_NE(even_id + 2, allocated_id);
+
+}
+
// Verifies that duplicated label is not allowed for RTP data channel.
TEST_F(MediaStreamSignalingTest, RtpDuplicatedLabelNotAllowed) {
AddDataChannel(cricket::DCT_RTP, "a", -1);
diff --git a/app/webrtc/webrtcsession.cc b/app/webrtc/webrtcsession.cc
index 17177bd..1b1c287 100644
--- a/app/webrtc/webrtcsession.cc
+++ b/app/webrtc/webrtcsession.cc
@@ -371,14 +371,15 @@
// Set |option| to the highest-priority value of |key| in the optional
// constraints if the key is found and has a valid value.
+template<typename T>
static void SetOptionFromOptionalConstraint(
const MediaConstraintsInterface* constraints,
- const std::string& key, cricket::Settable<int>* option) {
+ const std::string& key, cricket::Settable<T>* option) {
if (!constraints) {
return;
}
std::string string_value;
- int value;
+ T value;
if (constraints->GetOptional().FindFirst(key, &string_value)) {
if (talk_base::FromString(string_value, &value)) {
option->Set(value);
@@ -565,26 +566,22 @@
MediaConstraintsInterface::kCpuOveruseThreshold,
&video_options_.cpu_overuse_threshold);
SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kCpuOveruseDetection,
+ &video_options_.cpu_overuse_detection);
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kCpuOveruseEncodeUsage,
+ &video_options_.cpu_overuse_encode_usage);
+ SetOptionFromOptionalConstraint(constraints,
MediaConstraintsInterface::kCpuUnderuseEncodeRsdThreshold,
&video_options_.cpu_underuse_encode_rsd_threshold);
SetOptionFromOptionalConstraint(constraints,
MediaConstraintsInterface::kCpuOveruseEncodeRsdThreshold,
&video_options_.cpu_overuse_encode_rsd_threshold);
- if (FindConstraint(
- constraints,
- MediaConstraintsInterface::kCpuOveruseDetection,
- &value,
- NULL)) {
- video_options_.cpu_overuse_detection.Set(value);
- }
- if (FindConstraint(
- constraints,
- MediaConstraintsInterface::kCpuOveruseEncodeUsage,
- &value,
- NULL)) {
- video_options_.cpu_overuse_encode_usage.Set(value);
- }
+ // Find payload padding constraint.
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kPayloadPadding,
+ &video_options_.use_payload_padding);
// Find improved wifi bwe constraint.
if (FindConstraint(
@@ -598,13 +595,9 @@
video_options_.use_improved_wifi_bandwidth_estimator.Set(true);
}
- if (FindConstraint(
- constraints,
- MediaConstraintsInterface::kHighStartBitrate,
- &value,
- NULL)) {
- video_options_.video_start_bitrate.Set(cricket::kHighStartBitrate);
- }
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kHighStartBitrate,
+ &video_options_.video_start_bitrate);
if (FindConstraint(
constraints,
@@ -622,6 +615,10 @@
cricket::VideoOptions::HIGH);
}
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kOpusFec,
+ &audio_options_.opus_fec);
+
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,
@@ -1151,6 +1148,8 @@
}
void WebRtcSession::RemoveSctpDataStream(uint32 sid) {
+ mediastream_signaling_->RemoveSctpDataChannel(static_cast<int>(sid));
+
if (!data_channel_.get()) {
LOG(LS_ERROR) << "RemoveDataChannelStreams called when data_channel_ is "
<< "NULL.";
diff --git a/base/common.cc b/base/common.cc
index 40755dd..9f63aa4 100644
--- a/base/common.cc
+++ b/base/common.cc
@@ -78,4 +78,12 @@
}
}
+bool IsOdd(int n) {
+ return (n & 0x1);
+}
+
+bool IsEven(int n) {
+ return !IsOdd(n);
+}
+
} // namespace talk_base
diff --git a/base/common.h b/base/common.h
index 381b6b5..ed7d59e 100644
--- a/base/common.h
+++ b/base/common.h
@@ -116,6 +116,10 @@
// only by one component.
void SetCustomAssertLogger(AssertLogger logger);
+bool IsOdd(int n);
+
+bool IsEven(int n);
+
} // namespace talk_base
#if ENABLE_DEBUG
diff --git a/base/profiler_unittest.cc b/base/profiler_unittest.cc
index f451e5f..a39b32c 100644
--- a/base/profiler_unittest.cc
+++ b/base/profiler_unittest.cc
@@ -47,13 +47,15 @@
TEST(ProfilerTest, TestFunction) {
ASSERT_TRUE(Profiler::Instance()->Clear());
+
// Profile a long-running function.
const char* function_name = TestFunc();
const ProfilerEvent* event = Profiler::Instance()->GetEvent(function_name);
ASSERT_TRUE(event != NULL);
EXPECT_FALSE(event->is_started());
EXPECT_EQ(1, event->event_count());
- EXPECT_NEAR(kWaitSec, event->mean(), kTolerance);
+ EXPECT_NEAR(kWaitSec, event->mean(), kTolerance * 3);
+
// Run it a second time.
TestFunc();
EXPECT_FALSE(event->is_started());
@@ -95,7 +97,9 @@
// Check the result.
EXPECT_FALSE(event2->is_started());
EXPECT_EQ(1, event2->event_count());
- EXPECT_NEAR(kEvent2WaitSec, event2->mean(), kTolerance);
+
+ // The difference here can be as much as 0.33, so we need high tolerance.
+ EXPECT_NEAR(kEvent2WaitSec, event2->mean(), kTolerance * 4);
// Make sure event1 is unchanged.
EXPECT_FALSE(event1->is_started());
EXPECT_EQ(1, event1->event_count());
diff --git a/libjingle_tests.gyp b/libjingle_tests.gyp
index 00e2230..016f0a5 100755
--- a/libjingle_tests.gyp
+++ b/libjingle_tests.gyp
@@ -287,9 +287,7 @@
'media/base/videoengine_unittest.h',
'media/devices/dummydevicemanager_unittest.cc',
'media/devices/filevideocapturer_unittest.cc',
- # TODO(jiayl): Enable the SCTP test once the memcheck and tsan bots
- # failures are fixed (issue 2846).
- #'media/sctp/sctpdataengine_unittest.cc',
+ 'media/sctp/sctpdataengine_unittest.cc',
'media/webrtc/webrtcpassthroughrender_unittest.cc',
'media/webrtc/webrtcvideocapturer_unittest.cc',
# Omitted because depends on non-open-source testdata files.
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 49902ee..34d2def 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -182,6 +182,7 @@
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
dscp.SetFrom(change.dscp);
+ opus_fec.SetFrom(change.opus_fec);
}
bool operator==(const AudioOptions& o) const {
@@ -207,7 +208,8 @@
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
- dscp == o.dscp;
+ dscp == o.dscp &&
+ opus_fec == o.opus_fec;
}
std::string ToString() const {
@@ -238,6 +240,7 @@
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("dscp", dscp);
+ ost << ToStringIfSet("opus_fec", opus_fec);
ost << "}";
return ost.str();
}
@@ -275,6 +278,8 @@
Settable<uint32> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.
Settable<bool> dscp;
+ // Set Opus FEC
+ Settable<bool> opus_fec;
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
@@ -335,6 +340,7 @@
screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
use_improved_wifi_bandwidth_estimator.SetFrom(
change.use_improved_wifi_bandwidth_estimator);
+ use_payload_padding.SetFrom(change.use_payload_padding);
}
bool operator==(const VideoOptions& o) const {
@@ -374,7 +380,8 @@
skip_encoding_unused_streams == o.skip_encoding_unused_streams &&
screencast_min_bitrate == o.screencast_min_bitrate &&
use_improved_wifi_bandwidth_estimator ==
- o.use_improved_wifi_bandwidth_estimator;
+ o.use_improved_wifi_bandwidth_estimator &&
+ use_payload_padding == o.use_payload_padding;
}
std::string ToString() const {
@@ -421,6 +428,7 @@
ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
ost << ToStringIfSet("improved wifi bwe",
use_improved_wifi_bandwidth_estimator);
+ ost << ToStringIfSet("payload padding", use_payload_padding);
ost << "}";
return ost.str();
}
@@ -501,6 +509,8 @@
Settable<int> screencast_min_bitrate;
// Enable improved bandwidth estiamtor on wifi.
Settable<bool> use_improved_wifi_bandwidth_estimator;
+ // Enable payload padding.
+ Settable<bool> use_payload_padding;
};
// A class for playing out soundclips.
diff --git a/media/base/videocapturer_unittest.cc b/media/base/videocapturer_unittest.cc
index 75da236..9f025e3 100644
--- a/media/base/videocapturer_unittest.cc
+++ b/media/base/videocapturer_unittest.cc
@@ -31,6 +31,12 @@
const int kMinHdHeight = 720;
const uint32 kTimeout = 5000U;
+void NormalizeVideoSize(int* expected_width, int* expected_height) {
+ // WebRtcVideoFrame truncates the frame size to a multiple of 4.
+ *expected_width = *expected_width & ~3;
+ *expected_height = *expected_height & ~3;
+}
+
} // namespace
// Sets the elapsed time in the video frame to 0.
@@ -228,6 +234,59 @@
EXPECT_EQ(2, renderer_.num_rendered_frames());
}
+TEST_F(VideoCapturerTest, ScreencastScaledOddWidth) {
+ capturer_.SetScreencast(true);
+
+ int kWidth = 1281;
+ int kHeight = 720;
+
+ std::vector<cricket::VideoFormat> formats;
+ formats.push_back(cricket::VideoFormat(kWidth, kHeight,
+ cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
+ capturer_.ResetSupportedFormats(formats);
+
+ EXPECT_EQ(cricket::CS_RUNNING, capturer_.Start(cricket::VideoFormat(
+ kWidth,
+ kHeight,
+ cricket::VideoFormat::FpsToInterval(30),
+ cricket::FOURCC_ARGB)));
+ EXPECT_TRUE(capturer_.IsRunning());
+ EXPECT_EQ(0, renderer_.num_rendered_frames());
+ int expected_width = kWidth;
+ int expected_height = kHeight;
+ NormalizeVideoSize(&expected_width, &expected_height);
+ renderer_.SetSize(expected_width, expected_height, 0);
+ EXPECT_TRUE(capturer_.CaptureFrame());
+ EXPECT_EQ(1, renderer_.num_rendered_frames());
+}
+
+TEST_F(VideoCapturerTest, ScreencastScaledSuperLarge) {
+ capturer_.SetScreencast(true);
+
+ const int kMaxWidth = 4096;
+ const int kMaxHeight = 3072;
+ int kWidth = kMaxWidth + 4;
+ int kHeight = kMaxHeight + 4;
+
+ std::vector<cricket::VideoFormat> formats;
+ formats.push_back(cricket::VideoFormat(kWidth, kHeight,
+ cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
+ capturer_.ResetSupportedFormats(formats);
+
+ EXPECT_EQ(cricket::CS_RUNNING, capturer_.Start(cricket::VideoFormat(
+ kWidth,
+ kHeight,
+ cricket::VideoFormat::FpsToInterval(30),
+ cricket::FOURCC_ARGB)));
+ EXPECT_TRUE(capturer_.IsRunning());
+ EXPECT_EQ(0, renderer_.num_rendered_frames());
+ int expected_width = 2050;
+ int expected_height = 1538;
+ NormalizeVideoSize(&expected_width, &expected_height);
+ renderer_.SetSize(expected_width, expected_height, 0);
+ EXPECT_TRUE(capturer_.CaptureFrame());
+ EXPECT_EQ(1, renderer_.num_rendered_frames());
+}
TEST_F(VideoCapturerTest, TestFourccMatch) {
cricket::VideoFormat desired(640, 480,
diff --git a/media/sctp/sctpdataengine.cc b/media/sctp/sctpdataengine.cc
index 46b2ece..3647d21 100644
--- a/media/sctp/sctpdataengine.cc
+++ b/media/sctp/sctpdataengine.cc
@@ -277,18 +277,20 @@
}
SctpDataEngine::~SctpDataEngine() {
- // TODO(ldixon): There is currently a bug in teardown of usrsctp that blocks
- // indefintely if a finish call made too soon after close calls. So teardown
- // has been skipped. Once the bug is fixed, retest and enable teardown.
- // Tracked in webrtc issue 2749.
- //
- // usrsctp_engines_count--;
- // LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
- // if (usrsctp_engines_count == 0) {
- // if (usrsctp_finish() != 0) {
- // LOG(LS_WARNING) << "usrsctp_finish.";
- // }
- // }
+ usrsctp_engines_count--;
+ LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
+
+ if (usrsctp_engines_count == 0) {
+ // usrsctp_finish() may fail if it's called too soon after the channels are
+ // closed. Wait and try again until it succeeds for up to 3 seconds.
+ for (size_t i = 0; i < 300; ++i) {
+ if (usrsctp_finish() == 0)
+ return;
+
+ talk_base::Thread::SleepMs(10);
+ }
+ LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
+ }
}
DataMediaChannel* SctpDataEngine::CreateChannel(
diff --git a/media/sctp/sctpdataengine_unittest.cc b/media/sctp/sctpdataengine_unittest.cc
index 092524b..cf410e5 100644
--- a/media/sctp/sctpdataengine_unittest.cc
+++ b/media/sctp/sctpdataengine_unittest.cc
@@ -81,13 +81,13 @@
// an SCTP packet.
virtual void OnMessage(talk_base::Message* msg) {
LOG(LS_VERBOSE) << "SctpFakeNetworkInterface::OnMessage";
- talk_base::Buffer* buffer =
+ talk_base::scoped_ptr<talk_base::Buffer> buffer(
static_cast<talk_base::TypedMessageData<talk_base::Buffer*>*>(
- msg->pdata)->data();
+ msg->pdata)->data());
if (dest_) {
- dest_->OnPacketReceived(buffer, talk_base::PacketTime());
+ dest_->OnPacketReceived(buffer.get(), talk_base::PacketTime());
}
- delete buffer;
+ delete msg->pdata;
}
// Unsupported functions required to exist by NetworkInterface.
@@ -295,7 +295,7 @@
params.ssrc = ssrc;
return chan->SendData(params, talk_base::Buffer(
- msg.data(), msg.length()), result);
+ &msg[0], msg.length()), result);
}
bool ReceivedData(const SctpFakeDataReceiver* recv, uint32 ssrc,
@@ -364,26 +364,26 @@
EXPECT_EQ(cricket::SDR_SUCCESS, result);
EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
LOG(LS_VERBOSE) << "recv2.received=" << receiver2()->received()
- << "recv2.last_params.ssrc="
+ << ", recv2.last_params.ssrc="
<< receiver2()->last_params().ssrc
- << "recv2.last_params.timestamp="
+ << ", recv2.last_params.timestamp="
<< receiver2()->last_params().ssrc
- << "recv2.last_params.seq_num="
+ << ", recv2.last_params.seq_num="
<< receiver2()->last_params().seq_num
- << "recv2.last_data=" << receiver2()->last_data();
+ << ", recv2.last_data=" << receiver2()->last_data();
LOG(LS_VERBOSE) << "chan2 sending: 'hi chan1' -----------------------------";
ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
EXPECT_EQ(cricket::SDR_SUCCESS, result);
EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
LOG(LS_VERBOSE) << "recv1.received=" << receiver1()->received()
- << "recv1.last_params.ssrc="
+ << ", recv1.last_params.ssrc="
<< receiver1()->last_params().ssrc
- << "recv1.last_params.timestamp="
+ << ", recv1.last_params.timestamp="
<< receiver1()->last_params().ssrc
- << "recv1.last_params.seq_num="
+ << ", recv1.last_params.seq_num="
<< receiver1()->last_params().seq_num
- << "recv1.last_data=" << receiver1()->last_data();
+ << ", recv1.last_data=" << receiver1()->last_data();
}
// Sends a lot of large messages at once and verifies SDR_BLOCK is returned.
@@ -398,7 +398,7 @@
for (size_t i = 0; i < 100; ++i) {
channel1()->SendData(
- params, talk_base::Buffer(buffer.data(), buffer.size()), &result);
+ params, talk_base::Buffer(&buffer[0], buffer.size()), &result);
if (result == cricket::SDR_BLOCK)
break;
}
diff --git a/media/webrtc/fakewebrtcvideoengine.h b/media/webrtc/fakewebrtcvideoengine.h
index 3eba47b..85c59d8 100644
--- a/media/webrtc/fakewebrtcvideoengine.h
+++ b/media/webrtc/fakewebrtcvideoengine.h
@@ -1000,6 +1000,10 @@
return 0;
}
+#ifdef USE_WEBRTC_DEV_BRANCH
+ WEBRTC_STUB(SetPadWithRedundantPayloads, (int, bool));
+#endif
+
WEBRTC_FUNC(SetRtxReceivePayloadType, (const int channel,
const uint8 payload_type)) {
WEBRTC_CHECK_CHANNEL(channel);
@@ -1111,7 +1115,7 @@
channels_[channel]->transmission_smoothing_ = enable;
return 0;
}
- WEBRTC_FUNC(SetReservedTransmitBitrate, (int channel,
+ WEBRTC_FUNC(SetReservedTransmitBitrate, (int channel,
unsigned int reserved_transmit_bitrate_bps)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->reserved_transmit_bitrate_bps_ =
diff --git a/media/webrtc/fakewebrtcvoiceengine.h b/media/webrtc/fakewebrtcvoiceengine.h
index ff2079b..25c952d 100644
--- a/media/webrtc/fakewebrtcvoiceengine.h
+++ b/media/webrtc/fakewebrtcvoiceengine.h
@@ -536,6 +536,7 @@
}
WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
webrtc::VadModes& mode, bool& disabledDTX));
+#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->codec_fec = enable;
@@ -546,6 +547,7 @@
enable = channels_[channel]->codec_fec;
return 0;
}
+#endif // USE_WEBRTC_DEV_BRANCH
// webrtc::VoEDtmf
WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index f1810bf..fd609e9 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -558,6 +558,7 @@
}
void Enable(bool enable) {
+ LOG(LS_INFO) << "WebRtcOveruseObserver enable: " << enable;
talk_base::CritScope cs(&crit_);
enabled_ = enable;
}
@@ -586,8 +587,7 @@
external_capture_(external_capture),
capturer_updated_(false),
interval_(0),
- cpu_monitor_(cpu_monitor),
- overuse_observer_enabled_(false) {
+ cpu_monitor_(cpu_monitor) {
}
int channel_id() const { return channel_id_; }
@@ -679,7 +679,8 @@
vie_wrapper->base()->RegisterCpuOveruseObserver(channel_id_,
overuse_observer_.get());
// (Dis)connect the video adapter from the cpu monitor as appropriate.
- SetCpuOveruseDetection(overuse_observer_enabled_);
+ SetCpuOveruseDetection(
+ video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
SignalCpuAdaptationUnable.repeat(adapter->SignalCpuAdaptationUnable);
}
@@ -698,10 +699,18 @@
}
void ApplyCpuOptions(const VideoOptions& video_options) {
+ bool cpu_overuse_detection_changed =
+ video_options.cpu_overuse_detection.IsSet() &&
+ (video_options.cpu_overuse_detection.GetWithDefaultIfUnset(false) !=
+ video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
// Use video_options_.SetAll() instead of assignment so that unset value in
// video_options will not overwrite the previous option value.
video_options_.SetAll(video_options);
UpdateAdapterCpuOptions();
+ if (cpu_overuse_detection_changed) {
+ SetCpuOveruseDetection(
+ video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
+ }
}
void UpdateAdapterCpuOptions() {
@@ -709,15 +718,19 @@
return;
}
- bool cpu_adapt, cpu_smoothing, adapt_third;
+ bool cpu_smoothing, adapt_third;
float low, med, high;
+ bool cpu_adapt =
+ video_options_.adapt_input_to_cpu_usage.GetWithDefaultIfUnset(false);
+ bool cpu_overuse_detection =
+ video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
// TODO(thorcarpenter): Have VideoAdapter be responsible for setting
// all these video options.
CoordinatedVideoAdapter* video_adapter = video_capturer_->video_adapter();
- if (video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt) ||
- overuse_observer_enabled_) {
- video_adapter->set_cpu_adaptation(cpu_adapt || overuse_observer_enabled_);
+ if (video_options_.adapt_input_to_cpu_usage.IsSet() ||
+ video_options_.cpu_overuse_detection.IsSet()) {
+ video_adapter->set_cpu_adaptation(cpu_adapt || cpu_overuse_detection);
}
if (video_options_.adapt_cpu_with_smoothing.Get(&cpu_smoothing)) {
video_adapter->set_cpu_smoothing(cpu_smoothing);
@@ -737,8 +750,6 @@
}
void SetCpuOveruseDetection(bool enable) {
- overuse_observer_enabled_ = enable;
-
if (overuse_observer_) {
overuse_observer_->Enable(enable);
}
@@ -747,10 +758,6 @@
// it will be signaled by cpu monitor.
CoordinatedVideoAdapter* adapter = video_adapter();
if (adapter) {
- bool cpu_adapt = false;
- video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt);
- adapter->set_cpu_adaptation(
- adapter->cpu_adaptation() || cpu_adapt || enable);
if (cpu_monitor_) {
if (enable) {
cpu_monitor_->SignalUpdate.disconnect(adapter);
@@ -815,7 +822,6 @@
talk_base::CpuMonitor* cpu_monitor_;
talk_base::scoped_ptr<WebRtcOveruseObserver> overuse_observer_;
- bool overuse_observer_enabled_;
VideoOptions video_options_;
};
@@ -2967,9 +2973,6 @@
bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
(options_.buffered_mode_latency != options.buffered_mode_latency);
- 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 suspend_below_min_bitrate_changed =
@@ -2988,6 +2991,11 @@
options_.use_improved_wifi_bandwidth_estimator !=
options.use_improved_wifi_bandwidth_estimator;
+#ifdef USE_WEBRTC_DEV_BRANCH
+ bool payload_padding_changed = options.use_payload_padding.IsSet() &&
+ options_.use_payload_padding != options.use_payload_padding;
+#endif
+
// Save the options, to be interpreted where appropriate.
// Use options_.SetAll() instead of assignment so that unset value in options
@@ -3076,17 +3084,6 @@
}
}
}
- if (cpu_overuse_detection_changed) {
- bool cpu_overuse_detection =
- options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
- LOG(LS_INFO) << "CPU overuse detection is enabled? "
- << cpu_overuse_detection;
- for (SendChannelMap::iterator iter = send_channels_.begin();
- iter != send_channels_.end(); ++iter) {
- WebRtcVideoChannelSendInfo* send_channel = iter->second;
- send_channel->SetCpuOveruseDetection(cpu_overuse_detection);
- }
- }
if (dscp_option_changed) {
talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
@@ -3120,6 +3117,17 @@
it->second->channel_id(), config);
}
}
+#ifdef USE_WEBRTC_DEV_BRANCH
+ if (payload_padding_changed) {
+ LOG(LS_INFO) << "Payload-based padding called.";
+ for (SendChannelMap::iterator it = send_channels_.begin();
+ it != send_channels_.end(); ++it) {
+ engine()->vie()->rtp()->SetPadWithRedundantPayloads(
+ it->second->channel_id(),
+ options_.use_payload_padding.GetWithDefaultIfUnset(false));
+ }
+ }
+#endif
webrtc::CpuOveruseOptions overuse_options;
if (GetCpuOveruseOptions(options_, &overuse_options)) {
for (SendChannelMap::iterator it = send_channels_.begin();
@@ -3560,10 +3568,6 @@
send_channel->SignalCpuAdaptationUnable.connect(this,
&WebRtcVideoMediaChannel::OnCpuAdaptationUnable);
- if (options_.cpu_overuse_detection.GetWithDefaultIfUnset(false)) {
- send_channel->SetCpuOveruseDetection(true);
- }
-
webrtc::CpuOveruseOptions overuse_options;
if (GetCpuOveruseOptions(options_, &overuse_options)) {
if (engine()->vie()->base()->SetCpuOveruseOptions(channel_id,
@@ -3703,7 +3707,7 @@
target_codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
bool enable_denoising =
- options_.video_noise_reduction.GetWithDefaultIfUnset(false);
+ options_.video_noise_reduction.GetWithDefaultIfUnset(true);
target_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
}
@@ -3976,17 +3980,21 @@
// Turn off VP8 frame dropping when screensharing as the current model does
// not work well at low fps.
bool vp8_frame_dropping = !is_screencast;
- // Disable denoising for screencasting.
+ // TODO(pbos): Remove |video_noise_reduction| and enable it for all
+ // non-screencast.
bool enable_denoising =
- options_.video_noise_reduction.GetWithDefaultIfUnset(false);
+ options_.video_noise_reduction.GetWithDefaultIfUnset(true);
+ // Disable denoising for screencasting.
+ if (is_screencast) {
+ enable_denoising = false;
+ }
int screencast_min_bitrate =
options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
bool leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(true);
- bool denoising = !is_screencast && enable_denoising;
bool reset_send_codec =
target_width != cur_width || target_height != cur_height ||
automatic_resize != vie_codec.codecSpecific.VP8.automaticResizeOn ||
- denoising != vie_codec.codecSpecific.VP8.denoisingOn ||
+ enable_denoising != vie_codec.codecSpecific.VP8.denoisingOn ||
vp8_frame_dropping != vie_codec.codecSpecific.VP8.frameDroppingOn;
if (reset_send_codec) {
@@ -3999,7 +4007,7 @@
vie_codec.maxBitrate = target_codec.maxBitrate;
vie_codec.targetBitrate = 0;
vie_codec.codecSpecific.VP8.automaticResizeOn = automatic_resize;
- vie_codec.codecSpecific.VP8.denoisingOn = denoising;
+ vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
MaybeChangeBitrates(channel_id, &vie_codec);
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index d6d1354..716c5a8 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -278,6 +278,13 @@
video_codecs_ = DefaultVideoCodecs();
default_codec_format_ = VideoFormat(kDefaultVideoFormat);
+
+ rtp_header_extensions_.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
+ kRtpTimestampOffsetHeaderExtensionDefaultId));
+ rtp_header_extensions_.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+ kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@@ -774,6 +781,20 @@
return true;
}
+static std::string RtpExtensionsToString(
+ const std::vector<RtpHeaderExtension>& extensions) {
+ std::stringstream out;
+ out << '{';
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
+ if (i != extensions.size() - 1) {
+ out << ", ";
+ }
+ }
+ out << '}';
+ return out.str();
+}
+
} // namespace
bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
@@ -967,6 +988,8 @@
config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
}
+ config.rtp.extensions = send_rtp_extensions_;
+
if (IsNackEnabled(codec_settings.codec)) {
config.rtp.nack.rtp_history_ms = kNackHistoryMs;
}
@@ -1047,6 +1070,7 @@
config.rtp.nack.rtp_history_ms = kNackHistoryMs;
}
config.rtp.remb = true;
+ config.rtp.extensions = recv_rtp_extensions_;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
// corresponding sender SSRC. Figure out what to do when we don't have
@@ -1280,15 +1304,31 @@
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
- // TODO(pbos): Implement.
- LOG(LS_VERBOSE) << "SetRecvRtpHeaderExtensions()";
+ LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
+ << RtpExtensionsToString(extensions);
+ std::vector<webrtc::RtpExtension> webrtc_extensions;
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ // TODO(pbos): Make sure we don't pass unsupported extensions!
+ webrtc::RtpExtension webrtc_extension(extensions[i].uri.c_str(),
+ extensions[i].id);
+ webrtc_extensions.push_back(webrtc_extension);
+ }
+ recv_rtp_extensions_ = webrtc_extensions;
return true;
}
bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
- // TODO(pbos): Implement.
- LOG(LS_VERBOSE) << "SetSendRtpHeaderExtensions()";
+ LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
+ << RtpExtensionsToString(extensions);
+ std::vector<webrtc::RtpExtension> webrtc_extensions;
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ // TODO(pbos): Make sure we don't pass unsupported extensions!
+ webrtc::RtpExtension webrtc_extension(extensions[i].uri.c_str(),
+ extensions[i].id);
+ webrtc_extensions.push_back(webrtc_extension);
+ }
+ send_rtp_extensions_ = webrtc_extensions;
return true;
}
diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h
index d1a784d..81466eb 100644
--- a/media/webrtc/webrtcvideoengine2.h
+++ b/media/webrtc/webrtcvideoengine2.h
@@ -236,6 +236,9 @@
OVERRIDE;
virtual void OnReadyToSend(bool ready) OVERRIDE;
virtual bool MuteStream(uint32 ssrc, bool mute) OVERRIDE;
+
+ // Set send/receive RTP header extensions. This must be done before creating
+ // streams as it only has effect on future streams.
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetSendRtpHeaderExtensions(
@@ -351,8 +354,11 @@
std::map<uint32, webrtc::VideoReceiveStream*> receive_streams_;
Settable<VideoCodecSettings> send_codec_;
+ std::vector<webrtc::RtpExtension> send_rtp_extensions_;
+
WebRtcVideoEncoderFactory2* const encoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_;
+ std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
VideoOptions options_;
};
diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc
index c9ff182..6886300 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -396,6 +396,31 @@
FAIL() << "No RTX codec found among default codecs.";
}
+TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
+ std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
+ ASSERT_FALSE(extensions.empty());
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ if (extensions[i].uri == kRtpTimestampOffsetHeaderExtension) {
+ EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extensions[i].id);
+ return;
+ }
+ }
+ FAIL() << "Timestamp offset extension not in header-extension list.";
+}
+
+TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
+ std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
+ ASSERT_FALSE(extensions.empty());
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
+ EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
+ extensions[i].id);
+ return;
+ }
+ }
+ FAIL() << "Absolute Sender Time extension not in header-extension list.";
+}
+
class WebRtcVideoChannel2BaseTest
: public VideoMediaChannelTest<WebRtcVideoEngine2, WebRtcVideoChannel2> {
protected:
@@ -598,6 +623,67 @@
EXPECT_EQ(video_codec.height, webrtc_codec.height);
EXPECT_EQ(video_codec.framerate, webrtc_codec.maxFramerate);
}
+
+ void TestSetSendRtpHeaderExtensions(const std::string& cricket_ext,
+ const std::string& webrtc_ext) {
+ // Enable extension.
+ const int id = 1;
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ extensions.push_back(cricket::RtpHeaderExtension(cricket_ext, id));
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+
+ FakeVideoSendStream* send_stream =
+ AddSendStream(cricket::StreamParams::CreateLegacy(123));
+
+ // Verify the send extension id.
+ ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
+ EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
+ EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name);
+ // Verify call with same set of extensions returns true.
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ // Verify that SetSendRtpHeaderExtensions doesn't implicitly add them for
+ // receivers.
+ EXPECT_TRUE(AddRecvStream(cricket::StreamParams::CreateLegacy(123))
+ ->GetConfig()
+ .rtp.extensions.empty());
+
+ // Remove the extension id, verify that this doesn't reset extensions as
+ // they should be set before creating channels.
+ std::vector<cricket::RtpHeaderExtension> empty_extensions;
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
+ EXPECT_FALSE(send_stream->GetConfig().rtp.extensions.empty());
+ }
+
+ void TestSetRecvRtpHeaderExtensions(const std::string& cricket_ext,
+ const std::string& webrtc_ext) {
+ // Enable extension.
+ const int id = 1;
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ extensions.push_back(cricket::RtpHeaderExtension(cricket_ext, id));
+ EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+
+ FakeVideoReceiveStream* recv_stream =
+ AddRecvStream(cricket::StreamParams::CreateLegacy(123));
+
+ // Verify the recv extension id.
+ ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
+ EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
+ EXPECT_EQ(webrtc_ext, recv_stream->GetConfig().rtp.extensions[0].name);
+ // Verify call with same set of extensions returns true.
+ EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+ // Verify that SetRecvRtpHeaderExtensions doesn't implicitly add them for
+ // senders.
+ EXPECT_TRUE(AddSendStream(cricket::StreamParams::CreateLegacy(123))
+ ->GetConfig()
+ .rtp.extensions.empty());
+
+ // Remove the extension id, verify that this doesn't reset extensions as
+ // they should be set before creating channels.
+ std::vector<cricket::RtpHeaderExtension> empty_extensions;
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
+ EXPECT_FALSE(recv_stream->GetConfig().rtp.extensions.empty());
+ }
+
talk_base::scoped_ptr<VideoMediaChannel> channel_;
FakeWebRtcVideoChannel2* fake_channel_;
uint32 last_ssrc_;
@@ -723,12 +809,34 @@
ASSERT_TRUE(recv_stream->GetConfig().rtp.rtx.empty());
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_RtpTimestampOffsetHeaderExtensions) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+TEST_F(WebRtcVideoChannel2Test, NoHeaderExtesionsByDefault) {
+ FakeVideoSendStream* send_stream =
+ AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
+ ASSERT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
+
+ FakeVideoReceiveStream* recv_stream =
+ AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
+ ASSERT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_AbsoluteSendTimeHeaderExtensions) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+// Test support for RTP timestamp offset header extension.
+TEST_F(WebRtcVideoChannel2Test, SendRtpTimestampOffsetHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension,
+ webrtc::RtpExtension::kTOffset);
+}
+TEST_F(WebRtcVideoChannel2Test, RecvRtpTimestampOffsetHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension,
+ webrtc::RtpExtension::kTOffset);
+}
+
+// Test support for absolute send time header extension.
+TEST_F(WebRtcVideoChannel2Test, SendAbsoluteSendTimeHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
+ webrtc::RtpExtension::kAbsSendTime);
+}
+TEST_F(WebRtcVideoChannel2Test, RecvAbsoluteSendTimeHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
+ webrtc::RtpExtension::kAbsSendTime);
}
TEST_F(WebRtcVideoChannel2Test, DISABLED_LeakyBucketTest) {
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 785cdf1..f1460a6 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -237,6 +237,7 @@
options.experimental_aec.Set(false);
options.experimental_ns.Set(false);
options.aec_dump.Set(false);
+ options.opus_fec.Set(false);
return options;
}
@@ -399,12 +400,8 @@
// True if params["stereo"] == "1"
static bool IsOpusStereoEnabled(const AudioCodec& codec) {
- CodecParameterMap::const_iterator param =
- codec.params.find(kCodecParamStereo);
- if (param == codec.params.end()) {
- return false;
- }
- return param->second == kParamValueTrue;
+ int value;
+ return codec.GetParam(kCodecParamStereo, &value) && value == 1;
}
static bool IsValidOpusBitrate(int bitrate) {
@@ -426,14 +423,20 @@
return bitrate;
}
-// True if params["useinbandfec"] == "1"
+// Return true params[kCodecParamUseInbandFec] == kParamValueTrue, false
+// otherwise.
static bool IsOpusFecEnabled(const AudioCodec& codec) {
- CodecParameterMap::const_iterator param =
- codec.params.find(kCodecParamUseInbandFec);
- if (param == codec.params.end())
- return false;
+ int value;
+ return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
+}
- return param->second == kParamValueTrue;
+// Set params[kCodecParamUseInbandFec]. Caller should make sure codec is Opus.
+static void SetOpusFec(AudioCodec *codec, bool opus_fec) {
+ if (opus_fec) {
+ codec->params[kCodecParamUseInbandFec] = kParamValueTrue;
+ } else {
+ codec->params.erase(kCodecParamUseInbandFec);
+ }
}
void WebRtcVoiceEngine::ConstructCodecs() {
@@ -480,6 +483,7 @@
}
// TODO(hellner): Add ptime, sprop-stereo, stereo and useinbandfec
// when they can be set to values other than the default.
+ SetOpusFec(&codec, false);
}
codecs_.push_back(codec);
} else {
@@ -905,6 +909,16 @@
}
}
+ bool opus_fec = false;
+ if (options.opus_fec.Get(&opus_fec)) {
+ LOG(LS_INFO) << "Opus FEC is enabled? " << opus_fec;
+ for (std::vector<AudioCodec>::iterator it = codecs_.begin();
+ it != codecs_.end(); ++it) {
+ if (IsOpus(*it))
+ SetOpusFec(&(*it), opus_fec);
+ }
+ }
+
return true;
}
@@ -2022,7 +2036,7 @@
voe_codec.rate = bitrate_from_params;
}
- // If FEC is enabled.
+ // For Opus, we also enable inband FEC if it is requested.
if (IsOpusFecEnabled(*it)) {
LOG(LS_INFO) << "Enabling Opus FEC on channel " << channel;
#ifdef USE_WEBRTC_DEV_BRANCH
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 80a50c5..58893b9 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -1146,7 +1146,7 @@
#ifdef USE_WEBRTC_DEV_BRANCH
// Test that without useinbandfec, Opus FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFEC) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
@@ -1157,7 +1157,7 @@
}
// Test that with useinbandfec=0, Opus FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusDisableFEC) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusDisableFec) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
@@ -1174,7 +1174,7 @@
}
// Test that with useinbandfec=1, Opus FEC is on.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFEC) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFec) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
@@ -1191,7 +1191,7 @@
}
// Test that with useinbandfec=1, stereo=1, Opus FEC is on.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFECStereo) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFecStereo) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
@@ -1209,7 +1209,7 @@
}
// Test that with non-Opus, codec FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacNoFEC) {
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacNoFec) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
@@ -1219,6 +1219,31 @@
}
#endif // USE_WEBRTC_DEV_BRANCH
+// Test AudioOptions controls whether opus FEC is supported in codec list.
+TEST_F(WebRtcVoiceEngineTestFake, OpusFecViaOptions) {
+ EXPECT_TRUE(SetupEngine());
+ std::vector<cricket::AudioCodec> codecs = engine_.codecs();
+ int value;
+ for (std::vector<cricket::AudioCodec>::const_iterator it = codecs.begin();
+ it != codecs.end(); ++it) {
+ if (_stricmp(it->name.c_str(), cricket::kOpusCodecName) == 0) {
+ EXPECT_FALSE(it->GetParam(cricket::kCodecParamUseInbandFec, &value));
+ }
+ }
+
+ cricket::AudioOptions options;
+ options.opus_fec.Set(true);
+ EXPECT_TRUE(engine_.SetOptions(options));
+ codecs = engine_.codecs();
+ for (std::vector<cricket::AudioCodec>::const_iterator it = codecs.begin();
+ it != codecs.end(); ++it) {
+ if (_stricmp(it->name.c_str(), cricket::kOpusCodecName) == 0) {
+ EXPECT_TRUE(it->GetParam(cricket::kCodecParamUseInbandFec, &value));
+ EXPECT_EQ(1, value);
+ }
+ }
+}
+
// Test that we can apply CELT with stereo mode but fail with mono mode.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) {
EXPECT_TRUE(SetupEngine());
diff --git a/session/media/channelmanager.cc b/session/media/channelmanager.cc
index 3461a9c..88316b5 100644
--- a/session/media/channelmanager.cc
+++ b/session/media/channelmanager.cc
@@ -578,6 +578,29 @@
return ret;
}
+// Sets Engine-specific audio options according to enabled experiments.
+bool ChannelManager::SetEngineAudioOptions(const AudioOptions& options) {
+ // If we're initialized, pass the settings to the media engine.
+ bool ret = false;
+ if (initialized_) {
+ ret = worker_thread_->Invoke<bool>(
+ Bind(&ChannelManager::SetEngineAudioOptions_w, this, options));
+ }
+
+ // If all worked well, save the audio options.
+ if (ret) {
+ audio_options_ = options;
+ }
+ return ret;
+}
+
+bool ChannelManager::SetEngineAudioOptions_w(const AudioOptions& options) {
+ ASSERT(worker_thread_ == talk_base::Thread::Current());
+ ASSERT(initialized_);
+
+ return media_engine_->SetAudioOptions(options);
+}
+
bool ChannelManager::GetOutputVolume(int* level) {
if (!initialized_) {
return false;
diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h
index e811bb7..e8d6c0e 100644
--- a/session/media/channelmanager.h
+++ b/session/media/channelmanager.h
@@ -143,6 +143,8 @@
bool SetAudioOptions(const std::string& wave_in_device,
const std::string& wave_out_device,
const AudioOptions& options);
+ // Sets Engine-specific audio options according to enabled experiments.
+ bool SetEngineAudioOptions(const AudioOptions& options);
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
bool IsSameCapturer(const std::string& capturer_name,
@@ -266,6 +268,7 @@
void DestroySoundclip_w(Soundclip* soundclip);
bool SetAudioOptions_w(const AudioOptions& options, int delay_offset,
const Device* in_dev, const Device* out_dev);
+ bool SetEngineAudioOptions_w(const AudioOptions& options);
bool SetCaptureDevice_w(const Device* cam_device);
void OnVideoCaptureStateChange(VideoCapturer* capturer,
CaptureState result);
diff --git a/session/media/channelmanager_unittest.cc b/session/media/channelmanager_unittest.cc
index 055e1ad..cbf19f8 100644
--- a/session/media/channelmanager_unittest.cc
+++ b/session/media/channelmanager_unittest.cc
@@ -321,6 +321,25 @@
EXPECT_FALSE(cm_->SetAudioOptions("audio-in9", "audio-out2", options));
}
+TEST_F(ChannelManagerTest, SetEngineAudioOptions) {
+ EXPECT_TRUE(cm_->Init());
+ // Test setting specific values.
+ AudioOptions options;
+ options.experimental_ns.Set(true);
+ EXPECT_TRUE(cm_->SetEngineAudioOptions(options));
+ bool experimental_ns = false;
+ EXPECT_TRUE(fme_->audio_options().experimental_ns.Get(&experimental_ns));
+ EXPECT_TRUE(experimental_ns);
+}
+
+TEST_F(ChannelManagerTest, SetEngineAudioOptionsBeforeInitFails) {
+ // Test that values that we set before Init are not applied.
+ AudioOptions options;
+ options.experimental_ns.Set(true);
+ EXPECT_FALSE(cm_->SetEngineAudioOptions(options));
+ EXPECT_FALSE(fme_->audio_options().experimental_ns.IsSet());
+}
+
TEST_F(ChannelManagerTest, SetCaptureDeviceBeforeInit) {
// Test that values that we set before Init are applied.
EXPECT_TRUE(cm_->SetCaptureDevice("video-in2"));