Using mock transport controller in audio unit tests.
Using a mock of rtp transport controller send in audio send stream unit
tests. This reduces the dependencies and makes the tests more focused
on testing the functionality of audio send stream itself.
Bug: webrtc:8415
Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2
Reviewed-on: https://webrtc-review.googlesource.com/56600
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22161}diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 43e6c47..4130e68 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -16,12 +16,13 @@
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
-#include "call/rtp_transport_controller_send.h"
+#include "call/test/mock_rtp_transport_controller_send.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "rtc_base/fakeclock.h"
@@ -126,8 +127,6 @@
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: stream_config_(nullptr),
audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
- simulated_clock_(123456),
- rtp_transport_(&simulated_clock_, &event_log_, BitrateConstraints()),
bitrate_allocator_(&limit_observer_),
worker_queue_("ConfigHelper_worker_queue"),
audio_encoder_(nullptr) {
@@ -201,12 +200,15 @@
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
+ EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
+ .WillRepeatedly(Return(&bandwidth_observer_));
if (audio_bwe_enabled) {
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
- EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
- &rtp_transport_, Ne(nullptr)))
+ EXPECT_CALL(*channel_proxy_,
+ RegisterSenderCongestionControlObjects(
+ &rtp_transport_, Eq(&bandwidth_observer_)))
.Times(1);
} else {
EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
@@ -303,10 +305,10 @@
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
AudioProcessingStats audio_processing_stats_;
- SimulatedClock simulated_clock_;
TimeInterval active_lifetime_;
+ testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
testing::NiceMock<MockRtcEventLog> event_log_;
- RtpTransportControllerSend rtp_transport_;
+ testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
testing::NiceMock<MockRtpRtcp> rtp_rtcp_;
MockRtcpRttStats rtcp_rtt_stats_;
testing::NiceMock<MockLimitObserver> limit_observer_;