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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
47 lines
1.8 KiB
C++
47 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Unit tests for DecisionLogic class and derived classes.
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#include "modules/audio_coding/neteq/decision_logic.h"
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#include "modules/audio_coding/neteq/buffer_level_filter.h"
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#include "modules/audio_coding/neteq/decoder_database.h"
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#include "modules/audio_coding/neteq/delay_manager.h"
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#include "modules/audio_coding/neteq/delay_peak_detector.h"
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#include "modules/audio_coding/neteq/packet_buffer.h"
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#include "modules/audio_coding/neteq/statistics_calculator.h"
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#include "modules/audio_coding/neteq/tick_timer.h"
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#include "test/gtest.h"
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#include "test/mock_audio_decoder_factory.h"
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namespace webrtc {
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TEST(DecisionLogic, CreateAndDestroy) {
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int fs_hz = 8000;
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int output_size_samples = fs_hz / 100; // Samples per 10 ms.
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DecoderDatabase decoder_database(
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new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
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TickTimer tick_timer;
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StatisticsCalculator stats;
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PacketBuffer packet_buffer(10, &tick_timer);
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DelayPeakDetector delay_peak_detector(&tick_timer, false);
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auto delay_manager = DelayManager::Create(240, 0, false, &delay_peak_detector,
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&tick_timer, &stats);
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BufferLevelFilter buffer_level_filter;
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DecisionLogic* logic = DecisionLogic::Create(
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fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
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delay_manager.get(), &buffer_level_filter, &tick_timer);
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delete logic;
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}
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// TODO(hlundin): Write more tests.
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} // namespace webrtc
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