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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
commit | d8bbc7858622b6d9c278469aab701ca0b609cddf (patch) | |
tree | eff41dc61d9f714852212739e6b3738b82a2af87 /third_party/libwebrtc/audio/channel_send_unittest.cc | |
parent | Releasing progress-linux version 125.0.3-1~progress7.99u1. (diff) | |
download | firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.tar.xz firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.zip |
Merging upstream version 126.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/channel_send_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/audio/channel_send_unittest.cc | 22 |
1 files changed, 11 insertions, 11 deletions
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<AudioEncoder> encoder = @@ -94,7 +94,7 @@ class ChannelSendTest : public ::testing::Test { GlobalSimulatedTimeController time_controller_; webrtc::test::ScopedKeyValueConfig field_trials_; - RtcEventLogNull event_log_; + Environment env_; NiceMock<MockTransport> transport_; CryptoOptions crypto_options_; RtpTransportControllerSend transport_controller_; |