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Add a test that shows when channel_receive fires RR

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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}
This commit is contained in:
Harald Alvestrand 2023-02-01 14:55:13 +00:00 committed by WebRTC LUCI CQ
parent 84f75699c6
commit ba846ccf24
2 changed files with 70 additions and 21 deletions
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4
audio/BUILD.gn
View File

@ -228,13 +228,17 @@ if (rtc_include_tests) {
":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) {

71
audio/channel_receive_unittest.cc
View File

@ -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"
@ -22,19 +26,27 @@
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,
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,
@ -43,8 +55,41 @@ TEST(ChannelReceiveTest, CreateAndDestroy) {
/* codec_pair_id= */ absl::nullopt,
/* frame_decryptor_interface= */ nullptr, crypto_options,
/* frame_transformer= */ nullptr);
EXPECT_TRUE(!!channel);
}
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