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b1ae5ccd16
This reverts commit f8e62fcb14.
Reason for revert: breaks downstream test.
Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
> packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
> Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}
TBR=ivoc@webrtc.org,jakobi@webrtc.org
Change-Id: I1bdeacce61b902a0003a40c740f6acccf1443e3e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32329}
48 lines
1.7 KiB
C++
48 lines
1.7 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 "api/neteq/neteq_controller.h"
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#include "api/neteq/tick_timer.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/packet_buffer.h"
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#include "modules/audio_coding/neteq/statistics_calculator.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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StatisticsCalculator stats;
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PacketBuffer packet_buffer(10, &tick_timer);
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BufferLevelFilter buffer_level_filter;
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NetEqController::Config config;
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config.tick_timer = &tick_timer;
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config.base_min_delay_ms = 0;
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config.max_packets_in_buffer = 240;
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config.enable_rtx_handling = false;
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config.allow_time_stretching = true;
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auto logic = std::make_unique<DecisionLogic>(std::move(config));
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logic->SetSampleRate(fs_hz, output_size_samples);
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}
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// TODO(hlundin): Write more tests.
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} // namespace webrtc
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