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80c4cca491
The playout modes other than Normal have not been reachable for a long time, other than through tests. It is time to deprecate them. The only meaningful use was that Fax mode was sometimes set from tests, in order to avoid time-stretching operations (accelerate and pre-emptive expand) from messing with the test results. With this CL, a new config is added instead, which lets the user specify exactly this: don't do time-stretching. As a result of Fax and Off modes being removed, the following code clean-up was done: - Fold DecisionLogicNormal into DecisionLogic. - Remove AudioRepetition and AlternativePlc operations, since they can no longer be reached. Bug: webrtc:9421 Change-Id: I651458e9c1931a99f3b07e242817d303bac119df Reviewed-on: https://webrtc-review.googlesource.com/84123 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23704}
43 lines
1.6 KiB
C++
43 lines
1.6 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/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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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, false, &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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