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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/audio/channel_send_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
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 | 113 |
1 files changed, 113 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send_unittest.cc b/third_party/libwebrtc/audio/channel_send_unittest.cc new file mode 100644 index 0000000000..50d8368d4a --- /dev/null +++ b/third_party/libwebrtc/audio/channel_send_unittest.cc @@ -0,0 +1,113 @@ +/* + * Copyright 2023 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. + */ + +#include "audio/channel_send.h" + +#include <utility> + +#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/scoped_refptr.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "call/rtp_transport_controller_send.h" +#include "test/gtest.h" +#include "test/mock_transport.h" +#include "test/scoped_key_value_config.h" +#include "test/time_controller/simulated_time_controller.h" + +namespace webrtc { +namespace voe { +namespace { + +constexpr int kRtcpIntervalMs = 1000; +constexpr int kSsrc = 333; +constexpr int kPayloadType = 1; + +BitrateConstraints GetBitrateConfig() { + BitrateConstraints bitrate_config; + bitrate_config.min_bitrate_bps = 10000; + bitrate_config.start_bitrate_bps = 100000; + bitrate_config.max_bitrate_bps = 1000000; + return bitrate_config; +} + +std::unique_ptr<AudioFrame> CreateAudioFrame() { + auto frame = std::make_unique<AudioFrame>(); + frame->samples_per_channel_ = 480; + frame->sample_rate_hz_ = 48000; + frame->num_channels_ = 1; + return frame; +} + +class ChannelSendTest : public ::testing::Test { + protected: + ChannelSendTest() + : time_controller_(Timestamp::Seconds(1)), + transport_controller_( + time_controller_.GetClock(), + RtpTransportConfig{ + .bitrate_config = GetBitrateConfig(), + .event_log = &event_log_, + .task_queue_factory = time_controller_.GetTaskQueueFactory(), + .trials = &field_trials_, + }) { + transport_controller_.EnsureStarted(); + } + + std::unique_ptr<ChannelSendInterface> CreateChannelSend() { + return voe::CreateChannelSend( + time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(), + &transport_, nullptr, &event_log_, nullptr, crypto_options_, false, + kRtcpIntervalMs, kSsrc, nullptr, nullptr, field_trials_); + } + + GlobalSimulatedTimeController time_controller_; + webrtc::test::ScopedKeyValueConfig field_trials_; + RtcEventLogNull event_log_; + MockTransport transport_; + RtpTransportControllerSend transport_controller_; + CryptoOptions crypto_options_; +}; + +TEST_F(ChannelSendTest, StopSendShouldResetEncoder) { + std::unique_ptr<ChannelSendInterface> channel = CreateChannelSend(); + rtc::scoped_refptr<AudioEncoderFactory> encoder_factory = + CreateBuiltinAudioEncoderFactory(); + std::unique_ptr<AudioEncoder> encoder = encoder_factory->MakeAudioEncoder( + kPayloadType, SdpAudioFormat("opus", 48000, 2), {}); + channel->SetEncoder(kPayloadType, std::move(encoder)); + channel->RegisterSenderCongestionControlObjects(&transport_controller_, + nullptr); + channel->StartSend(); + + // Insert two frames which should trigger a new packet. + EXPECT_CALL(transport_, SendRtp).Times(1); + channel->ProcessAndEncodeAudio(CreateAudioFrame()); + time_controller_.AdvanceTime(webrtc::TimeDelta::Zero()); + channel->ProcessAndEncodeAudio(CreateAudioFrame()); + time_controller_.AdvanceTime(webrtc::TimeDelta::Zero()); + + EXPECT_CALL(transport_, SendRtp).Times(0); + channel->ProcessAndEncodeAudio(CreateAudioFrame()); + time_controller_.AdvanceTime(webrtc::TimeDelta::Zero()); + // StopSend should clear the previous audio frame stored in the encoder. + channel->StopSend(); + channel->StartSend(); + // The following frame should not trigger a new packet since the encoder + // needs 20 ms audio. + channel->ProcessAndEncodeAudio(CreateAudioFrame()); + time_controller_.AdvanceTime(webrtc::TimeDelta::Zero()); +} + +} // namespace +} // namespace voe +} // namespace webrtc |