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This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).

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FakeRtpTransportController moves to a common header and its constructor is changed to take a SendSideCongestionController to enable injecting the mock.

BUG=webrtc:7395

Review-Url: https://codereview.webrtc.org/2834663003
Cr-Commit-Position: refs/heads/master@{#18055}
This commit is contained in:
zstein 2017-05-08 11:52:38 -07:00 committed by Commit bot
parent 966963ac7d
commit 7cb69d5cc7
14 changed files with 167 additions and 52 deletions
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1
webrtc/audio/BUILD.gn
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@ -81,6 +81,7 @@ if (rtc_include_tests) {
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/congestion_controller:congestion_controller",
"../modules/congestion_controller:mock_congestion_controller",
"../modules/pacing:pacing",
"../test:test_common",
"../test:test_support",

45
webrtc/audio/audio_send_stream_unittest.cc
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@ -15,7 +15,9 @@
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/call/fake_rtp_transport_controller_send.h"
#include "webrtc/call/rtp_transport_controller_send_interface.h"
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
@ -23,8 +25,8 @@
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_audio_encoder.h"
#include "webrtc/test/mock_audio_encoder_factory.h"
@ -124,36 +126,15 @@ rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
}
struct ConfigHelper {
class FakeRtpTransportController
: public RtpTransportControllerSendInterface {
public:
explicit FakeRtpTransportController(RtcEventLog* event_log)
: simulated_clock_(123456),
send_side_cc_(&simulated_clock_,
&bitrate_observer_,
event_log,
&packet_router_) {}
PacketRouter* packet_router() override { return &packet_router_; }
SendSideCongestionController* send_side_cc() override {
return &send_side_cc_;
}
TransportFeedbackObserver* transport_feedback_observer() override {
return &send_side_cc_;
}
RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); }
private:
SimulatedClock simulated_clock_;
testing::NiceMock<MockCongestionObserver> bitrate_observer_;
PacketRouter packet_router_;
SendSideCongestionController send_side_cc_;
};
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: stream_config_(nullptr),
fake_transport_(&event_log_),
simulated_clock_(123456),
send_side_cc_(rtc::MakeUnique<SendSideCongestionController>(
&simulated_clock_,
nullptr /* observer */,
&event_log_,
&packet_router_)),
fake_transport_(send_side_cc_.get()),
bitrate_allocator_(&limit_observer_),
worker_queue_("ConfigHelper_worker_queue") {
using testing::Invoke;
@ -322,11 +303,13 @@ struct ConfigHelper {
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
testing::NiceMock<MockCongestionObserver> bitrate_observer_;
MockAudioProcessing audio_processing_;
MockTransmitMixer transmit_mixer_;
AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
FakeRtpTransportController fake_transport_;
SimulatedClock simulated_clock_;
PacketRouter packet_router_;
std::unique_ptr<SendSideCongestionController> send_side_cc_;
FakeRtpTransportControllerSend fake_transport_;
MockRtcEventLog event_log_;
MockRtpRtcp rtp_rtcp_;
MockRtcpRttStats rtcp_rtt_stats_;

1
webrtc/call/BUILD.gn
View File

@ -93,6 +93,7 @@ if (rtc_include_tests) {
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/bitrate_controller",
"../modules/congestion_controller:mock_congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../system_wrappers",

20
webrtc/call/call.cc
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@ -26,6 +26,7 @@
#include "webrtc/base/location.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
@ -97,7 +98,7 @@ class Call : public webrtc::Call,
public BitrateAllocator::LimitObserver {
public:
Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSend> transport_send);
std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
virtual ~Call();
// Implements webrtc::Call.
@ -298,16 +299,21 @@ std::string Call::Stats::ToString(int64_t time_ms) const {
}
Call* Call::Create(const Call::Config& config) {
return new internal::Call(
config, std::unique_ptr<RtpTransportControllerSend>(
new RtpTransportControllerSend(Clock::GetRealTimeClock(),
config.event_log)));
return new internal::Call(config,
rtc::MakeUnique<RtpTransportControllerSend>(
Clock::GetRealTimeClock(), config.event_log));
}
Call* Call::Create(
const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
return new internal::Call(config, std::move(transport_send));
}
namespace internal {
Call::Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSend> transport_send)
std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
: clock_(Clock::GetRealTimeClock()),
num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
@ -342,7 +348,7 @@ Call::Call(const Call::Config& config,
config.bitrate_config.start_bitrate_bps);
}
Trace::CreateTrace();
transport_send->RegisterNetworkObserver(this);
transport_send->send_side_cc()->RegisterNetworkObserver(this);
transport_send_ = std::move(transport_send);
transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
transport_send_->send_side_cc()->SetBweBitrates(

7
webrtc/call/call.h
View File

@ -10,6 +10,7 @@
#ifndef WEBRTC_CALL_CALL_H_
#define WEBRTC_CALL_CALL_H_
#include <memory>
#include <string>
#include <vector>
@ -20,6 +21,7 @@
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/call/flexfec_receive_stream.h"
#include "webrtc/call/rtp_transport_controller_send_interface.h"
#include "webrtc/common_types.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@ -98,6 +100,11 @@ class Call {
static Call* Create(const Call::Config& config);
// Allows mocking |transport_send| for testing.
static Call* Create(
const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
virtual AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) = 0;
virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;

28
webrtc/call/call_unittest.cc
View File

@ -11,11 +11,15 @@
#include <list>
#include <map>
#include <memory>
#include <utility>
#include "webrtc/base/ptr_util.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/call/call.h"
#include "webrtc/call/fake_rtp_transport_controller_send.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_audio_decoder_factory.h"
#include "webrtc/test/mock_transport.h"
@ -305,4 +309,28 @@ TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
}
}
// TODO(zstein): This is just a motivating example for
// MockSendSideCongestionController. It should be deleted once we have more
// meaningful tests.
TEST(CallTest, MockSendSideCongestionControllerExample) {
RtcEventLogNullImpl event_log;
Call::Config config(&event_log);
SimulatedClock clock(123456);
PacketRouter packet_router;
testing::NiceMock<test::MockSendSideCongestionController> mock_cc(
&clock, &event_log, &packet_router);
auto transport_send =
rtc::MakeUnique<FakeRtpTransportControllerSend>(&mock_cc);
std::unique_ptr<Call> call(Call::Create(config, std::move(transport_send)));
Call::Config::BitrateConfig bitrate_config;
bitrate_config.min_bitrate_bps = 1;
bitrate_config.start_bitrate_bps = 2;
bitrate_config.max_bitrate_bps = 3;
EXPECT_CALL(mock_cc, SetBweBitrates(1, 2, 3));
call->SetBitrateConfig(bitrate_config);
}
} // namespace webrtc

48
webrtc/call/fake_rtp_transport_controller_send.h Normal file
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@ -0,0 +1,48 @@
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
#define WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
#include "webrtc/call/rtp_transport_controller_send_interface.h"
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/modules/pacing/packet_router.h"
namespace webrtc {
class FakeRtpTransportControllerSend
: public RtpTransportControllerSendInterface {
public:
explicit FakeRtpTransportControllerSend(
SendSideCongestionController* send_side_cc)
: send_side_cc_(send_side_cc) {
RTC_DCHECK(send_side_cc);
}
PacketRouter* packet_router() override { return &packet_router_; }
SendSideCongestionController* send_side_cc() override {
return send_side_cc_;
}
TransportFeedbackObserver* transport_feedback_observer() override {
return send_side_cc_;
}
RtpPacketSender* packet_sender() override { return send_side_cc_->pacer(); }
private:
PacketRouter packet_router_;
SendSideCongestionController* send_side_cc_;
};
} // namespace webrtc
#endif // WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_

6
webrtc/call/rtp_transport_controller_send.cc
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@ -18,10 +18,4 @@ RtpTransportControllerSend::RtpTransportControllerSend(
: send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) {
}
void RtpTransportControllerSend::RegisterNetworkObserver(
SendSideCongestionController::Observer* observer) {
// Must be called only once.
send_side_cc_.RegisterNetworkObserver(observer);
}
} // namespace webrtc

3
webrtc/call/rtp_transport_controller_send.h
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@ -26,9 +26,6 @@ class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
public:
RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
void RegisterNetworkObserver(
SendSideCongestionController::Observer* observer);
// Implements RtpTransportControllerSendInterface
PacketRouter* packet_router() override { return &packet_router_; }
SendSideCongestionController* send_side_cc() override {

13
webrtc/modules/congestion_controller/BUILD.gn
View File

@ -85,6 +85,7 @@ if (rtc_include_tests) {
]
deps = [
":congestion_controller",
":mock_congestion_controller",
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",
@ -101,4 +102,16 @@ if (rtc_include_tests) {
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_source_set("mock_congestion_controller") {
testonly = true
sources = [
"include/mock/mock_congestion_observer.h",
"include/mock/mock_send_side_congestion_controller.h",
]
deps = [
":congestion_controller",
"../../test:test_support",
]
}
}

2
webrtc/modules/congestion_controller/include/congestion_controller.h
View File

@ -58,7 +58,7 @@ class CongestionController : public CallStatsObserver,
receive_side_cc_(clock, packet_router) {}
CongestionController(const Clock* clock,
Observer* observer,
RemoteBitrateObserver* remote_bitrate_observer,
RemoteBitrateObserver* /* remote_bitrate_observer */,
RtcEventLog* event_log,
PacketRouter* packet_router,
std::unique_ptr<PacedSender> pacer)

3
webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h
View File

@ -11,9 +11,6 @@
#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_CONGESTION_OBSERVER_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_CONGESTION_OBSERVER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/socket.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/test/gmock.h"

39
webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h Normal file
View File

@ -0,0 +1,39 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
namespace test {
class MockSendSideCongestionController : public SendSideCongestionController {
public:
MockSendSideCongestionController(const Clock* clock,
RtcEventLog* event_log,
PacketRouter* packet_router)
: SendSideCongestionController(clock,
nullptr /* observer */,
event_log,
packet_router) {}
MOCK_METHOD3(SetBweBitrates,
void(int min_bitrate_bps,
int start_bitrate_bps,
int max_bitrate_bps));
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_

3
webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
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@ -57,7 +57,8 @@ class SendSideCongestionController : public CallStatsObserver,
protected:
virtual ~Observer() {}
};
// TODO(nisse): Consider deleting the |observer| argument to constructors.
// TODO(nisse): Consider deleting the |observer| argument to constructors
// once CongestionController is deleted.
SendSideCongestionController(const Clock* clock,
Observer* observer,
RtcEventLog* event_log,