commit f6ce1d39ee58629953aa0c043c301b51d70a6236
author Per K <perkj@webrtc.org> Tue Mar 28 15:27:47 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Mar 29 06:36:17 2023
tree 631275105753927ef43256b9e32eaf6a3dc4aa76
parent ae049f192488fca413e7ce85b0d604c8b56c0d95

Allow injecting packets of type Any to Call::DeliverRtpPacket

MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.

Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}

call/BUILD.gn

diff --git a/call/BUILD.gn b/call/BUILD.gn
index f4c3295..32e5ca6 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -490,6 +490,7 @@
         "../api/task_queue:default_task_queue_factory",
         "../api/test/video:function_video_factory",
         "../api/transport:field_trial_based_config",
+        "../api/units:timestamp",
         "../api/video:builtin_video_bitrate_allocator_factory",
         "../api/video:video_frame",
         "../api/video:video_rtp_headers",

call/call.cc

diff --git a/call/call.cc b/call/call.cc
index ccfed35..b55492e 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1413,16 +1413,10 @@
     packet.set_arrival_time(Timestamp::Micros(packet_time_us));
   }
 
-  // We might get RTP keep-alive packets in accordance with RFC6263 section 4.6.
-  // These are empty (zero length payload) RTP packets with an unsignaled
-  // payload type.
-  const bool is_keep_alive_packet = packet.payload_size() == 0;
-  RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO ||
-             is_keep_alive_packet);
   NotifyBweOfReceivedPacket(packet, media_type);
 
+  event_log_->Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
   if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) {
-    RTC_DCHECK(is_keep_alive_packet);
     return;
   }
 
@@ -1443,7 +1437,6 @@
       return;
     }
   }
-  event_log_->Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
 
   // RateCounters expect input parameter as int, save it as int,
   // instead of converting each time it is passed to RateCounter::Add below.

call/call_unittest.cc

diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index 5db3f59..944006d 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -18,11 +18,13 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/media_types.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/test/mock_audio_mixer.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/transport/field_trial_based_config.h"
+#include "api/units/timestamp.h"
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "audio/audio_receive_stream.h"
 #include "audio/audio_send_stream.h"
@@ -42,6 +44,7 @@
 
 using ::testing::_;
 using ::testing::Contains;
+using ::testing::MockFunction;
 using ::testing::NiceMock;
 using ::testing::StrictMock;
 
@@ -323,6 +326,45 @@
   }
 }
 
+TEST(CallTest,
+     DeliverRtpPacketOfTypeAudioTriggerOnUndemuxablePacketHandlerIfNotDemuxed) {
+  CallHelper call(/*use_null_audio_processing=*/false);
+  MockFunction<bool(const RtpPacketReceived& parsed_packet)>
+      un_demuxable_packet_handler;
+
+  RtpPacketReceived packet;
+  packet.set_arrival_time(Timestamp::Millis(1));
+  EXPECT_CALL(un_demuxable_packet_handler, Call);
+  call->Receiver()->DeliverRtpPacket(
+      MediaType::AUDIO, packet, un_demuxable_packet_handler.AsStdFunction());
+}
+
+TEST(CallTest,
+     DeliverRtpPacketOfTypeVideoTriggerOnUndemuxablePacketHandlerIfNotDemuxed) {
+  CallHelper call(/*use_null_audio_processing=*/false);
+  MockFunction<bool(const RtpPacketReceived& parsed_packet)>
+      un_demuxable_packet_handler;
+
+  RtpPacketReceived packet;
+  packet.set_arrival_time(Timestamp::Millis(1));
+  EXPECT_CALL(un_demuxable_packet_handler, Call);
+  call->Receiver()->DeliverRtpPacket(
+      MediaType::VIDEO, packet, un_demuxable_packet_handler.AsStdFunction());
+}
+
+TEST(CallTest,
+     DeliverRtpPacketOfTypeAnyDoesNotTriggerOnUndemuxablePacketHandler) {
+  CallHelper call(/*use_null_audio_processing=*/false);
+  MockFunction<bool(const RtpPacketReceived& parsed_packet)>
+      un_demuxable_packet_handler;
+
+  RtpPacketReceived packet;
+  packet.set_arrival_time(Timestamp::Millis(1));
+  EXPECT_CALL(un_demuxable_packet_handler, Call).Times(0);
+  call->Receiver()->DeliverRtpPacket(
+      MediaType::ANY, packet, un_demuxable_packet_handler.AsStdFunction());
+}
+
 TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
   constexpr uint32_t kSSRC = 12345;
   for (bool use_null_audio_processing : {false, true}) {

call/packet_receiver.h

diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index c7f55ac..cdcf7bf 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -28,7 +28,9 @@
   using OnUndemuxablePacketHandler =
       absl::AnyInvocable<bool(const RtpPacketReceived& parsed_packet)>;
 
-  // Demux RTP packets. Must be called on the worker thread.
+  // Must be called on the worker thread.
+  // If `media_type` is not Audio or Video, packets may be used for BWE
+  // calculations but are not demuxed.
   virtual void DeliverRtpPacket(
       MediaType media_type,
       RtpPacketReceived packet,