47b17dc59c
The timescale_holdoff_ is a counter in the DecisionLogic class. The purpose is to enforce a minimum number of GetAudio calls between (successfull) time-scaling operations (i.e., Accelerate and Pre-emptive Expand operations). With this change, the counter is replaced with a Countdown timer obtained from a TickTimer object. BUG=webrtc:5608 R=tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1945863002 . Cr-Commit-Position: refs/heads/master@{#12670}
56 lines
2.3 KiB
C++
56 lines
2.3 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 "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
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#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
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#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
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#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
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#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
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#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
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#include "webrtc/modules/audio_coding/neteq/tick_timer.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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std::unique_ptr<MockAudioDecoderFactory>(new MockAudioDecoderFactory));
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TickTimer tick_timer;
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PacketBuffer packet_buffer(10, &tick_timer);
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DelayPeakDetector delay_peak_detector(&tick_timer);
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DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
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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, kPlayoutOn, &decoder_database, packet_buffer,
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&delay_manager, &buffer_level_filter, &tick_timer);
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delete logic;
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logic = DecisionLogic::Create(
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fs_hz, output_size_samples, kPlayoutStreaming, &decoder_database,
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packet_buffer, &delay_manager, &buffer_level_filter, &tick_timer);
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delete logic;
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logic = DecisionLogic::Create(
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fs_hz, output_size_samples, kPlayoutFax, &decoder_database, packet_buffer,
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&delay_manager, &buffer_level_filter, &tick_timer);
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delete logic;
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logic = DecisionLogic::Create(
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fs_hz, output_size_samples, kPlayoutOff, &decoder_database, packet_buffer,
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&delay_manager, &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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