Delete PacketReceiver::DeliverPacket from all implementations
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc
index afa3136e..6dedc59 100644
--- a/call/bitrate_estimator_tests.cc
+++ b/call/bitrate_estimator_tests.cc
@@ -110,6 +110,11 @@
virtual void SetUp() {
SendTask(task_queue(), [this]() {
+ RegisterRtpExtension(
+ RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
+ RegisterRtpExtension(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+
CreateCalls();
CreateSendTransport(BuiltInNetworkBehaviorConfig(), /*observer=*/nullptr);
diff --git a/call/call.cc b/call/call.cc
index 218505c..e676d7a 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -241,11 +241,6 @@
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
- // Implements PacketReceiver.
- DeliveryStatus DeliverPacket(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) override;
-
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override;
void DeliverRtpPacket(
@@ -339,9 +334,6 @@
void DeliverRtcp(MediaType media_type, rtc::CopyOnWriteBuffer packet)
RTC_RUN_ON(network_thread_);
- DeliveryStatus DeliverRtp(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) RTC_RUN_ON(worker_thread_);
AudioReceiveStreamImpl* FindAudioStreamForSyncGroup(
absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
@@ -351,7 +343,6 @@
MediaType media_type)
RTC_RUN_ON(worker_thread_);
- bool IdentifyReceivedPacket(RtpPacketReceived& packet);
bool RegisterReceiveStream(uint32_t ssrc, ReceiveStreamInterface* stream);
bool UnregisterReceiveStream(uint32_t ssrc);
@@ -1475,57 +1466,6 @@
}
}
-PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- // TODO(perkj, https://bugs.webrtc.org/7135): Deprecate this method and
- // direcly use DeliverRtpPacket.
- TRACE_EVENT0("webrtc", "Call::DeliverRtp");
- RTC_DCHECK_NE(media_type, MediaType::ANY);
-
- RtpPacketReceived parsed_packet;
- if (!parsed_packet.Parse(std::move(packet)))
- return DELIVERY_PACKET_ERROR;
-
- if (packet_time_us != -1) {
- parsed_packet.set_arrival_time(Timestamp::Micros(packet_time_us));
- } else {
- parsed_packet.set_arrival_time(clock_->CurrentTime());
- }
-
- if (!IdentifyReceivedPacket(parsed_packet))
- return DELIVERY_UNKNOWN_SSRC;
- if (media_type == MediaType::VIDEO) {
- parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
- }
- DeliverRtpPacket(media_type, std::move(parsed_packet),
- [](const webrtc::RtpPacketReceived& packet) {
- // If IdentifyReceivedPacket returns true, a packet is
- // expected to be demuxable.
- RTC_DCHECK_NOTREACHED();
- return false;
- });
- return DELIVERY_OK;
-}
-
-PacketReceiver::DeliveryStatus Call::DeliverPacket(
- MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- if (IsRtcpPacket(packet)) {
- RTC_DCHECK_RUN_ON(network_thread_);
- worker_thread_->PostTask(SafeTask(
- task_safety_.flag(), [this, packet = std::move(packet)]() mutable {
- RTC_DCHECK_RUN_ON(worker_thread_);
- DeliverRtcpPacket(std::move(packet));
- }));
- return DELIVERY_OK;
- }
-
- RTC_DCHECK_RUN_ON(worker_thread_);
- return DeliverRtp(media_type, std::move(packet), packet_time_us);
-}
-
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type) {
RTC_DCHECK_RUN_ON(worker_thread_);
@@ -1549,19 +1489,6 @@
}
}
-bool Call::IdentifyReceivedPacket(RtpPacketReceived& packet) {
- RTC_DCHECK_RUN_ON(&receive_11993_checker_);
- auto it = receive_rtp_config_.find(packet.Ssrc());
- if (it == receive_rtp_config_.end()) {
- RTC_DLOG(LS_WARNING) << "receive_rtp_config_ lookup failed for ssrc "
- << packet.Ssrc();
- return false;
- }
-
- packet.IdentifyExtensions(it->second->GetRtpExtensionMap());
- return true;
-}
-
bool Call::RegisterReceiveStream(uint32_t ssrc,
ReceiveStreamInterface* stream) {
RTC_DCHECK_RUN_ON(&receive_11993_checker_);
diff --git a/call/degraded_call.cc b/call/degraded_call.cc
index 50349c1..fc76c7b 100644
--- a/call/degraded_call.cc
+++ b/call/degraded_call.cc
@@ -346,25 +346,6 @@
call_->OnSentPacket(sent_packet);
}
-PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
- MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- RTC_DCHECK_RUN_ON(&received_packet_sequence_checker_);
- PacketReceiver::DeliveryStatus status = receive_pipe_->DeliverPacket(
- media_type, std::move(packet), packet_time_us);
- // This is not optimal, but there are many places where there are thread
- // checks that fail if we're not using the worker thread call into this
- // method. If we want to fix this we probably need a task queue to do handover
- // of all overriden methods, which feels like overkill for the current use
- // case.
- // By just having this thread call out via the Process() method we work around
- // that, with the tradeoff that a non-zero delay may become a little larger
- // than anticipated at very low packet rates.
- receive_pipe_->Process();
- return status;
-}
-
void DegradedCall::DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
diff --git a/call/degraded_call.h b/call/degraded_call.h
index 6a22b69..98e7891 100644
--- a/call/degraded_call.h
+++ b/call/degraded_call.h
@@ -113,9 +113,6 @@
protected:
// Implements PacketReceiver.
- DeliveryStatus DeliverPacket(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) override;
void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 76adfe3..8879927 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -191,16 +191,6 @@
return true;
}
-PacketReceiver::DeliveryStatus FakeNetworkPipe::DeliverPacket(
- MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- return EnqueuePacket(std::move(packet), absl::nullopt, false, media_type,
- packet_time_us)
- ? PacketReceiver::DELIVERY_OK
- : PacketReceiver::DELIVERY_PACKET_ERROR;
-}
-
void FakeNetworkPipe::DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
@@ -393,10 +383,6 @@
<< packet.Ssrc() << " seq : " << packet.SequenceNumber();
return false;
});
- } else {
- receiver_->DeliverPacket(packet->media_type(),
- std::move(*packet->raw_packet()),
- packet_time_us);
}
}
}
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index 2649a00..ba4c89e 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -162,12 +162,6 @@
OnUndemuxablePacketHandler undemuxable_packet_handler) override;
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override;
- // TODO(perkj, https://bugs.webrtc.org/7135): Remove once implementations
- // dont use it.
- PacketReceiver::DeliveryStatus DeliverPacket(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) override;
-
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void Process() override;
diff --git a/call/fake_network_pipe_unittest.cc b/call/fake_network_pipe_unittest.cc
index d3f7734..31f97fc 100644
--- a/call/fake_network_pipe_unittest.cc
+++ b/call/fake_network_pipe_unittest.cc
@@ -31,10 +31,6 @@
namespace webrtc {
class MockReceiver : public PacketReceiver {
public:
- MOCK_METHOD(DeliveryStatus,
- DeliverPacket,
- (MediaType, rtc::CopyOnWriteBuffer, int64_t),
- (override));
MOCK_METHOD(void,
DeliverRtcpPacket,
(rtc::CopyOnWriteBuffer packet),
diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index a36ab44..c7f55ac 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -20,26 +20,8 @@
class PacketReceiver {
public:
- enum DeliveryStatus {
- DELIVERY_OK,
- DELIVERY_UNKNOWN_SSRC,
- DELIVERY_PACKET_ERROR,
- };
-
- // TODO(perkj, https://bugs.webrtc.org/7135): Remove this method. This method
- // is no longer used by PeerConnections. Some tests still use it.
- virtual DeliveryStatus DeliverPacket(MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- RTC_CHECK_NOTREACHED();
- }
-
// Demux RTCP packets. Must be called on the worker thread.
- virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
- // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
- // FakeNetworkPipe.
- RTC_CHECK_NOTREACHED();
- }
+ virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) = 0;
// Invoked once when a packet packet is received that can not be demuxed.
// If the method returns true, a new attempt is made to demux the packet.
@@ -50,11 +32,7 @@
virtual void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
- OnUndemuxablePacketHandler undemuxable_packet_handler) {
- // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
- // FakeNetworkPipe.
- RTC_CHECK_NOTREACHED();
- }
+ OnUndemuxablePacketHandler undemuxable_packet_handler) = 0;
protected:
virtual ~PacketReceiver() {}
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index a20b826..6408e4e 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -665,21 +665,6 @@
return this;
}
-webrtc::PacketReceiver::DeliveryStatus FakeCall::DeliverPacket(
- webrtc::MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) {
- RTC_DCHECK(webrtc::IsRtpPacket(packet));
- uint32_t ssrc = ParseRtpSsrc(packet);
- webrtc::Timestamp arrival_time =
- packet_time_us > -1 ? webrtc::Timestamp::Micros(packet_time_us)
- : webrtc::Timestamp::Zero();
- if (DeliverPacketInternal(media_type, ssrc, packet, arrival_time)) {
- return DELIVERY_OK;
- }
- return DELIVERY_UNKNOWN_SSRC;
-}
-
void FakeCall::DeliverRtpPacket(
webrtc::MediaType media_type,
webrtc::RtpPacketReceived packet,
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index f7e3de5..954bd16 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -442,10 +442,6 @@
webrtc::PacketReceiver* Receiver() override;
- DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
- rtc::CopyOnWriteBuffer packet,
- int64_t packet_time_us) override;
-
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override {}
void DeliverRtpPacket(
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index 3ae0216..2604979 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -44,18 +44,6 @@
TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
- const std::map<uint8_t, MediaType>& payload_type_map)
- : DirectTransport(task_queue,
- std::move(pipe),
- send_call,
- payload_type_map,
- {},
- {}) {}
-
-DirectTransport::DirectTransport(
- TaskQueueBase* task_queue,
- std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
- Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map,
rtc::ArrayView<const RtpExtension> audio_extensions,
rtc::ArrayView<const RtpExtension> video_extensions)
@@ -63,7 +51,6 @@
task_queue_(task_queue),
demuxer_(payload_type_map),
fake_network_(std::move(pipe)),
- use_legacy_send_(audio_extensions.empty() && video_extensions.empty()),
audio_extensions_(audio_extensions),
video_extensions_(video_extensions) {
Start();
@@ -89,30 +76,27 @@
send_call_->OnSentPacket(sent_packet);
}
- if (use_legacy_send_) {
- LegacySendPacket(data, length);
- } else {
- const RtpHeaderExtensionMap* extensions = nullptr;
- MediaType media_type = demuxer_.GetMediaType(data, length);
- switch (demuxer_.GetMediaType(data, length)) {
- case webrtc::MediaType::AUDIO:
- extensions = &audio_extensions_;
- break;
- case webrtc::MediaType::VIDEO:
- extensions = &video_extensions_;
- break;
- default:
- RTC_CHECK_NOTREACHED();
- }
- RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
- if (media_type == MediaType::VIDEO) {
- packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
- }
- RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
- fake_network_->DeliverRtpPacket(
- media_type, std::move(packet),
- [](const RtpPacketReceived& packet) { return false; });
+ const RtpHeaderExtensionMap* extensions = nullptr;
+ MediaType media_type = demuxer_.GetMediaType(data, length);
+ switch (demuxer_.GetMediaType(data, length)) {
+ case webrtc::MediaType::AUDIO:
+ extensions = &audio_extensions_;
+ break;
+ case webrtc::MediaType::VIDEO:
+ extensions = &video_extensions_;
+ break;
+ default:
+ RTC_CHECK_NOTREACHED();
}
+ RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
+ if (media_type == MediaType::VIDEO) {
+ packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+ }
+ RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
+ fake_network_->DeliverRtpPacket(
+ media_type, std::move(packet),
+ [](const RtpPacketReceived& packet) { return false; });
+
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
@@ -120,24 +104,13 @@
}
bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
- if (use_legacy_send_) {
- LegacySendPacket(data, length);
- } else {
- fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
- }
+ fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
return true;
}
-void DirectTransport::LegacySendPacket(const uint8_t* data, size_t length) {
- MediaType media_type = demuxer_.GetMediaType(data, length);
- int64_t send_time_us = rtc::TimeMicros();
- fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
- send_time_us);
-}
-
int DirectTransport::GetAverageDelayMs() {
return fake_network_->AverageDelay();
}
diff --git a/test/direct_transport.h b/test/direct_transport.h
index 4776084..468e339 100644
--- a/test/direct_transport.h
+++ b/test/direct_transport.h
@@ -44,14 +44,6 @@
// same task-queue - the one that's passed in via the constructor.
class DirectTransport : public Transport {
public:
- // TODO(perkj, https://bugs.webrtc.org/7135): Remove header once downstream
- // projects have been updated.
- [[deprecated("Use ctor that provide header extensions.")]] DirectTransport(
- TaskQueueBase* task_queue,
- std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
- Call* send_call,
- const std::map<uint8_t, MediaType>& payload_type_map);
-
DirectTransport(TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
@@ -85,7 +77,6 @@
const Demuxer demuxer_;
const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
- const bool use_legacy_send_;
const RtpHeaderExtensionMap audio_extensions_;
const RtpHeaderExtensionMap video_extensions_;
};
diff --git a/video/end_to_end_tests/ssrc_tests.cc b/video/end_to_end_tests/ssrc_tests.cc
index a3bce40..edacde1 100644
--- a/video/end_to_end_tests/ssrc_tests.cc
+++ b/video/end_to_end_tests/ssrc_tests.cc
@@ -108,7 +108,8 @@
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
- receiver_call_.get(), payload_type_map_);
+ receiver_call_.get(), payload_type_map_, GetRegisteredExtensions(),
+ GetRegisteredExtensions());
input_observer =
std::make_unique<PacketInputObserver>(receiver_call_->Receiver());
send_transport->SetReceiver(input_observer.get());
