mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 22:30:40 +01:00

This is a reland of d9f798a6b3
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}
Bug: webrtc:9289
Change-Id: I40fbd999fc8495beaeb46799c333f91d72b5be37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125720
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26978}
46 lines
1.8 KiB
C++
46 lines
1.8 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/statistics_calculator.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;
|
|
StatisticsCalculator stats;
|
|
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, &stats);
|
|
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
|