Allow sending abs-send-time for audio streams.
Bug: webrtc:10742
Change-Id: I088c8221e04e84152cfce925051bf6bc23d5fe68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149061
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28861}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index b27e29c..4ee5109 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -207,6 +207,8 @@
for (const auto& extension : extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
ids.audio_level = extension.id;
+ } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+ ids.abs_send_time = extension.id;
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
ids.transport_sequence_number = extension.id;
} else if (extension.uri == RtpExtension::kMidUri) {
@@ -273,6 +275,16 @@
channel_send->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
new_ids.audio_level);
}
+
+ if (first_time || new_ids.abs_send_time != old_ids.abs_send_time) {
+ channel_send->GetRtpRtcp()->DeregisterSendRtpHeaderExtension(
+ kRtpExtensionAbsoluteSendTime);
+ if (new_ids.abs_send_time) {
+ channel_send->GetRtpRtcp()->RegisterSendRtpHeaderExtension(
+ kRtpExtensionAbsoluteSendTime, new_ids.abs_send_time);
+ }
+ }
+
bool transport_seq_num_id_changed =
new_ids.transport_sequence_number != old_ids.transport_sequence_number;
if (first_time || (transport_seq_num_id_changed &&
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 37eb89a..3649ddf 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -183,6 +183,7 @@
// So it should be safe to use 0 here to indicate "not configured".
struct ExtensionIds {
int audio_level = 0;
+ int abs_send_time = 0;
int transport_sequence_number = 0;
int mid = 0;
int rid = 0;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 540623e..7e62bc6 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -516,6 +516,8 @@
int id = 1;
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, id++));
+ capabilities.header_extensions.push_back(
+ webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, id++));
capabilities.header_extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri, id++));
return capabilities;
diff --git a/test/network/network_emulation.cc b/test/network/network_emulation.cc
index 8a19486..7d3ed44 100644
--- a/test/network/network_emulation.cc
+++ b/test/network/network_emulation.cc
@@ -102,6 +102,9 @@
void NetworkRouterNode::OnPacketReceived(EmulatedIpPacket packet) {
RTC_DCHECK_RUN_ON(task_queue_);
+ if (watcher_) {
+ watcher_(packet);
+ }
auto receiver_it = routing_.find(packet.to.ipaddr());
if (receiver_it == routing_.end()) {
return;
@@ -128,6 +131,14 @@
routing_.erase(dest_ip);
}
+void NetworkRouterNode::SetWatcher(
+ std::function<void(const EmulatedIpPacket&)> watcher) {
+ task_queue_->PostTask([=] {
+ RTC_DCHECK_RUN_ON(task_queue_);
+ watcher_ = watcher;
+ });
+}
+
EmulatedNetworkNode::EmulatedNetworkNode(
Clock* clock,
rtc::TaskQueue* task_queue,
diff --git a/test/network/network_emulation.h b/test/network/network_emulation.h
index 24e2fd9..c5ed539 100644
--- a/test/network/network_emulation.h
+++ b/test/network/network_emulation.h
@@ -102,11 +102,14 @@
void SetReceiver(rtc::IPAddress dest_ip,
EmulatedNetworkReceiverInterface* receiver);
void RemoveReceiver(rtc::IPAddress dest_ip);
+ void SetWatcher(std::function<void(const EmulatedIpPacket&)> watcher);
private:
rtc::TaskQueue* const task_queue_;
std::map<rtc::IPAddress, EmulatedNetworkReceiverInterface*> routing_
RTC_GUARDED_BY(task_queue_);
+ std::function<void(const EmulatedIpPacket&)> watcher_
+ RTC_GUARDED_BY(task_queue_);
};
// Represents node in the emulated network. Nodes can be connected with each
diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc
index 4509197..c72b9d2 100644
--- a/test/peer_scenario/peer_scenario_client.cc
+++ b/test/peer_scenario/peer_scenario_client.cc
@@ -241,7 +241,7 @@
SdpCreateObserver([=](SessionDescriptionInterface* offer) {
std::string sdp_offer;
offer->ToString(&sdp_offer);
- printf("%s\n", sdp_offer.c_str());
+ RTC_LOG(LS_INFO) << sdp_offer;
peer_connection_->SetLocalDescription(
SdpSetObserver([sdp_offer, offer_handler]() {
offer_handler(std::move(sdp_offer));
@@ -261,7 +261,7 @@
SdpCreateObserver([=](SessionDescriptionInterface* answer) {
std::string sdp_answer;
answer->ToString(&sdp_answer);
- printf("%s\n", sdp_answer.c_str());
+ RTC_LOG(LS_INFO) << sdp_answer;
peer_connection_->SetLocalDescription(
SdpSetObserver([answer_handler, sdp_answer]() {
answer_handler(sdp_answer);
diff --git a/test/peer_scenario/tests/BUILD.gn b/test/peer_scenario/tests/BUILD.gn
index 6c1c75b..d799d2c 100644
--- a/test/peer_scenario/tests/BUILD.gn
+++ b/test/peer_scenario/tests/BUILD.gn
@@ -17,7 +17,9 @@
]
deps = [
"..:peer_scenario",
+ "../../:field_trial",
"../../:test_support",
+ "../../../modules/rtp_rtcp:rtp_rtcp",
"../../../pc:rtc_pc_base",
]
}
diff --git a/test/peer_scenario/tests/remote_estimate_test.cc b/test/peer_scenario/tests/remote_estimate_test.cc
index 05addc2..81d788c 100644
--- a/test/peer_scenario/tests/remote_estimate_test.cc
+++ b/test/peer_scenario/tests/remote_estimate_test.cc
@@ -8,12 +8,37 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "pc/media_session.h"
#include "pc/session_description.h"
+#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/peer_scenario/peer_scenario.h"
namespace webrtc {
namespace test {
+namespace {
+RtpHeaderExtensionMap AudioExtensions(
+ const SessionDescriptionInterface& session) {
+ auto* audio_desc =
+ cricket::GetFirstAudioContentDescription(session.description());
+ return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
+}
+
+absl::optional<RTPHeaderExtension> GetRtpPacketExtensions(
+ const rtc::ArrayView<const uint8_t> packet,
+ const RtpHeaderExtensionMap& extension_map) {
+ RtpUtility::RtpHeaderParser rtp_parser(packet.data(), packet.size());
+ if (!rtp_parser.RTCP()) {
+ RTPHeader header;
+ if (rtp_parser.Parse(&header, &extension_map, true)) {
+ return header.extension;
+ }
+ }
+ return absl::nullopt;
+}
+
+} // namespace
TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
PeerScenario s;
@@ -45,5 +70,45 @@
EXPECT_TRUE(s.WaitAndProcess(&offer_exchange_done));
}
+TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
+ ScopedFieldTrials trials("WebRTC-KeepAbsSendTimeExtension/Enabled/");
+ PeerScenario s;
+
+ auto* caller = s.CreateClient(PeerScenarioClient::Config());
+ auto* callee = s.CreateClient(PeerScenarioClient::Config());
+
+ auto send_node = s.net()->NodeBuilder().Build().node;
+ auto ret_node = s.net()->NodeBuilder().Build().node;
+
+ s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
+ s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
+
+ auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
+ caller->CreateAudio("AUDIO", cricket::AudioOptions());
+ signaling.StartIceSignaling();
+ RtpHeaderExtensionMap extension_map;
+ rtc::Event offer_exchange_done;
+ signaling.NegotiateSdp(
+ [&extension_map](SessionDescriptionInterface* offer) {
+ extension_map = AudioExtensions(*offer);
+ EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
+ },
+ [&](const SessionDescriptionInterface& answer) {
+ EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
+ kRtpExtensionAbsoluteSendTime));
+ offer_exchange_done.Set();
+ });
+ EXPECT_TRUE(s.WaitAndProcess(&offer_exchange_done));
+ rtc::Event received_abs_send_time;
+ send_node->router()->SetWatcher(
+ [extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
+ auto extensions = GetRtpPacketExtensions(packet.data, extension_map);
+ if (extensions) {
+ EXPECT_TRUE(extensions->hasAbsoluteSendTime);
+ received_abs_send_time.Set();
+ }
+ });
+ EXPECT_TRUE(s.WaitAndProcess(&received_abs_send_time));
+}
} // namespace test
} // namespace webrtc
