webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
Jakob Ivarsson‎ 423bae437c Revert "Remove field trial include from decision logic."
This reverts commit d9f798a6b3.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Remove field trial include from decision logic.
> 
> Bug: webrtc:9289
> Change-Id: I2e465bf9eddda8bde50daeb14cfd51405e536ff4
> Reviewed-on: https://webrtc-review.googlesource.com/c/125097
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26925}

TBR=minyue@webrtc.org,jakobi@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9289
Change-Id: I439a7477c9b0d94abe815b375b05b7545e3617f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125683
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26967}
2019-03-05 10:08:49 +00:00

45 lines
1.7 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/field_trial.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, false);
auto delay_manager =
DelayManager::Create(240, 0, false, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(
fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
delay_manager.get(), &buffer_level_filter, &tick_timer);
delete logic;
}
// TODO(hlundin): Write more tests.
} // namespace webrtc