webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
Danil Chapovalov b602123a5a Replace rtc::Optional with absl::optional in modules/audio_coding
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_coding'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 12:46:20 +00:00

55 lines
2.2 KiB
C++

/*
* Copyright (c) 2012 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.
*/
// Unit tests for DecisionLogic class and derived classes.
#include "modules/audio_coding/neteq/decision_logic.h"
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/delay_peak_detector.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
namespace webrtc {
TEST(DecisionLogic, CreateAndDestroy) {
int fs_hz = 8000;
int output_size_samples = fs_hz / 100; // Samples per 10 ms.
DecoderDatabase decoder_database(
new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
TickTimer tick_timer;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer);
DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutOn, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutStreaming, &decoder_database,
packet_buffer, &delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutFax, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutOff, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
}
// TODO(hlundin): Write more tests.
} // namespace webrtc