Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(326)

Unified Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 2752233002: Split CongestionController into send- and receive-side classes. (Closed)
Patch Set: Use variable names receive_side_cc and send_side_cc. Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/audio/audio_send_stream.cc ('k') | webrtc/call/call.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 318ea955cf79dd0906c0d17164235d9330ff6d16..98ab9ebca22bbf2c9060cf4369420f481955c001 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -18,8 +18,8 @@
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
-#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.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_rtcp_rtt_stats.h"
#include "webrtc/test/gtest.h"
@@ -72,11 +72,10 @@ struct ConfigHelper {
explicit ConfigHelper(bool audio_bwe_enabled)
: simulated_clock_(123456),
stream_config_(nullptr),
- congestion_controller_(&simulated_clock_,
- &bitrate_observer_,
- nullptr,
- &event_log_,
- &packet_router_),
+ send_side_cc_(&simulated_clock_,
+ &bitrate_observer_,
+ &event_log_,
+ &packet_router_),
bitrate_allocator_(&limit_observer_),
worker_queue_("ConfigHelper_worker_queue") {
using testing::Invoke;
@@ -126,9 +125,7 @@ struct ConfigHelper {
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
PacketRouter* packet_router() { return &packet_router_; }
- CongestionController* congestion_controller() {
- return &congestion_controller_;
- }
+ SendSideCongestionController* send_side_cc() { return &send_side_cc_; }
BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
rtc::TaskQueue* worker_queue() { return &worker_queue_; }
RtcEventLog* event_log() { return &event_log_; }
@@ -149,16 +146,14 @@ struct ConfigHelper {
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
- EXPECT_CALL(*channel_proxy_,
- RegisterSenderCongestionControlObjects(
- congestion_controller_.pacer(), &congestion_controller_,
- packet_router(), Ne(nullptr)))
+ EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
+ send_side_cc_.pacer(), &send_side_cc_,
+ packet_router(), Ne(nullptr)))
.Times(1);
} else {
- EXPECT_CALL(*channel_proxy_,
- RegisterSenderCongestionControlObjects(
- congestion_controller_.pacer(), &congestion_controller_,
- packet_router(), Eq(nullptr)))
+ EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
+ send_side_cc_.pacer(), &send_side_cc_,
+ packet_router(), Eq(nullptr)))
.Times(1);
}
EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()).Times(1);
@@ -249,7 +244,7 @@ struct ConfigHelper {
MockTransmitMixer transmit_mixer_;
AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
PacketRouter packet_router_;
- CongestionController congestion_controller_;
+ SendSideCongestionController send_side_cc_;
MockRtcEventLog event_log_;
MockRtcpRttStats rtcp_rtt_stats_;
testing::NiceMock<MockLimitObserver> limit_observer_;
@@ -296,16 +291,16 @@ TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency, kTelephoneEventCode,
@@ -316,8 +311,8 @@ TEST(AudioSendStreamTest, SetMuted) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
send_stream.SetMuted(true);
}
@@ -326,24 +321,24 @@ TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(true);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -373,8 +368,8 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
helper.SetupMockForGetStats();
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
@@ -427,8 +422,8 @@ TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config));
internal::AudioSendStream send_stream(
stream_config, helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
}
// VAD is applied when codec is mono and the CNG frequency matches the codec
@@ -444,16 +439,16 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
.WillOnce(Return(true));
internal::AudioSendStream send_stream(
stream_config, helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
}
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
EXPECT_CALL(*helper.channel_proxy(),
SetBitrate(helper.config().max_bitrate_bps, _));
send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
@@ -464,8 +459,8 @@ TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
ConfigHelper helper(false);
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.packet_router(), helper.congestion_controller(),
- helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+ helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+ helper.event_log(), helper.rtcp_rtt_stats());
EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
}
« no previous file with comments | « webrtc/audio/audio_send_stream.cc ('k') | webrtc/call/call.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698