From d8bbc7858622b6d9c278469aab701ca0b609cddf Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 15 May 2024 05:35:49 +0200 Subject: Merging upstream version 126.0. Signed-off-by: Daniel Baumann --- .../libwebrtc/audio/channel_send_unittest.cc | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'third_party/libwebrtc/audio/channel_send_unittest.cc') diff --git a/third_party/libwebrtc/audio/channel_send_unittest.cc b/third_party/libwebrtc/audio/channel_send_unittest.cc index 58d7c93c1e..c86dcefadc 100644 --- a/third_party/libwebrtc/audio/channel_send_unittest.cc +++ b/third_party/libwebrtc/audio/channel_send_unittest.cc @@ -14,7 +14,8 @@ #include "api/audio/audio_frame.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" -#include "api/rtc_event_log/rtc_event_log.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" @@ -53,18 +54,17 @@ class ChannelSendTest : public ::testing::Test { protected: ChannelSendTest() : time_controller_(Timestamp::Seconds(1)), + env_(CreateEnvironment(&field_trials_, + time_controller_.GetClock(), + time_controller_.CreateTaskQueueFactory())), transport_controller_( - time_controller_.GetClock(), - RtpTransportConfig{ - .bitrate_config = GetBitrateConfig(), - .event_log = &event_log_, - .task_queue_factory = time_controller_.GetTaskQueueFactory(), - .trials = &field_trials_, - }) { + RtpTransportConfig{.env = env_, + .bitrate_config = GetBitrateConfig()}) { channel_ = voe::CreateChannelSend( time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(), - &transport_, nullptr, &event_log_, nullptr, crypto_options_, false, - kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_, field_trials_); + &transport_, nullptr, &env_.event_log(), nullptr, crypto_options_, + false, kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_, + env_.field_trials()); encoder_factory_ = CreateBuiltinAudioEncoderFactory(); SdpAudioFormat opus = SdpAudioFormat("opus", kRtpRateHz, 2); std::unique_ptr encoder = @@ -94,7 +94,7 @@ class ChannelSendTest : public ::testing::Test { GlobalSimulatedTimeController time_controller_; webrtc::test::ScopedKeyValueConfig field_trials_; - RtcEventLogNull event_log_; + Environment env_; NiceMock transport_; CryptoOptions crypto_options_; RtpTransportControllerSend transport_controller_; -- cgit v1.2.3