commit ba846ccf2487aa6b7123ad002245eed996cb8ed5
author Harald Alvestrand <hta@webrtc.org> Wed Feb 01 14:55:13 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Feb 01 16:38:38 2023
tree 035e3416c32499f7ada434729fd80995149aaab0
parent 84f75699c65384a961cdf9ebffefd6bc83e0d8e2

Add a test that shows when channel_receive fires RR

This seems to happen 2.5 seconds after initialization.
Written as part of debugging a different issue.

Bug: webrtc:13931
Change-Id: I3686cdbc39284505a437ebc0bfd8c74c483624c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291704
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39245}

audio/BUILD.gn

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 6ca47d4..a144540 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -228,13 +228,17 @@
       ":audio",
       "../api/crypto:frame_decryptor_interface",
       "../api/task_queue:default_task_queue_factory",
+      "../logging:mocks",
       "../modules/audio_device:audio_device_api",
       "../modules/audio_device:mock_audio_device",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
+      "../rtc_base:logging",
       "../rtc_base:threading",
       "../test:mock_transport",
       "../test:test_support",
       "../test/time_controller",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
   }
 
   if (rtc_enable_protobuf && !build_with_chromium) {

audio/channel_receive_unittest.cc

diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc
index 3d9baeb..b5654e2 100644
--- a/audio/channel_receive_unittest.cc
+++ b/audio/channel_receive_unittest.cc
@@ -12,8 +12,12 @@
 
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/task_queue/default_task_queue_factory.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -22,29 +26,70 @@
 
 namespace webrtc {
 namespace voe {
+namespace {
 
-TEST(ChannelReceiveTest, CreateAndDestroy) {
-  GlobalSimulatedTimeController time_controller(Timestamp::Seconds(5555));
-  uint32_t local_ssrc = 1111;
-  uint32_t remote_ssrc = 2222;
-  webrtc::CryptoOptions crypto_options;
-  rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_module =
-      test::MockAudioDeviceModule::CreateNice();
-  MockTransport transport;
-  auto channel = CreateChannelReceive(
-      time_controller.GetClock(),
-      /* neteq_factory= */ nullptr, audio_device_module.get(), &transport,
-      /* rtc_event_log= */ nullptr, local_ssrc, remote_ssrc,
-      /* jitter_buffer_max_packets= */ 0,
-      /* jitter_buffer_fast_playout= */ false,
-      /* jitter_buffer_min_delay_ms= */ 0,
-      /* enable_non_sender_rtt= */ false,
-      /* decoder_factory= */ nullptr,
-      /* codec_pair_id= */ absl::nullopt,
-      /* frame_decryptor_interface= */ nullptr, crypto_options,
-      /* frame_transformer= */ nullptr);
-  EXPECT_TRUE(!!channel);
+using ::testing::NiceMock;
+using ::testing::NotNull;
+using ::testing::Test;
+
+constexpr uint32_t kLocalSsrc = 1111;
+constexpr uint32_t kRemoteSsrc = 2222;
+
+class ChannelReceiveTest : public Test {
+ public:
+  ChannelReceiveTest()
+      : time_controller_(Timestamp::Seconds(5555)),
+        audio_device_module_(test::MockAudioDeviceModule::CreateStrict()) {}
+
+  std::unique_ptr<ChannelReceiveInterface> CreateTestChannelReceive() {
+    CryptoOptions crypto_options;
+    return CreateChannelReceive(
+        time_controller_.GetClock(),
+        /* neteq_factory= */ nullptr, audio_device_module_.get(), &transport_,
+        &event_log_, kLocalSsrc, kRemoteSsrc,
+        /* jitter_buffer_max_packets= */ 0,
+        /* jitter_buffer_fast_playout= */ false,
+        /* jitter_buffer_min_delay_ms= */ 0,
+        /* enable_non_sender_rtt= */ false,
+        /* decoder_factory= */ nullptr,
+        /* codec_pair_id= */ absl::nullopt,
+        /* frame_decryptor_interface= */ nullptr, crypto_options,
+        /* frame_transformer= */ nullptr);
+  }
+
+  NtpTime NtpNow() { return time_controller_.GetClock()->CurrentNtpTime(); }
+
+ protected:
+  GlobalSimulatedTimeController time_controller_;
+  rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_module_;
+  MockTransport transport_;
+  NiceMock<MockRtcEventLog> event_log_;
+};
+
+TEST_F(ChannelReceiveTest, CreateAndDestroy) {
+  auto channel = CreateTestChannelReceive();
+  EXPECT_THAT(channel, NotNull());
 }
 
+TEST_F(ChannelReceiveTest, ReceiveReportGeneratedOnTime) {
+  auto channel = CreateTestChannelReceive();
+  channel->SetReceiveCodecs({{10, {"L16", 44100, 1}}});
+
+  bool receiver_report_sent = false;
+  EXPECT_CALL(transport_, SendRtcp)
+      .WillRepeatedly([&](const uint8_t* packet, size_t length) {
+        if (length >= 2 && packet[1] == rtcp::ReceiverReport::kPacketType) {
+          receiver_report_sent = true;
+        }
+        return true;
+      });
+  // RFC 3550 section 6.2 mentions 5 seconds as a reasonable expectation
+  // for the interval between RTCP packets.
+  time_controller_.AdvanceTime(TimeDelta::Seconds(5));
+
+  EXPECT_TRUE(receiver_report_sent);
+}
+
+}  // namespace
 }  // namespace voe
 }  // namespace webrtc