webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
Ruslan Burakov 0ac1d993be Remove streaming_mode as it is always false.
Change-Id: I489b72985f36fd98413ecf729f7d69476c342851

Bug: webrtc:10618
Change-Id: I489b72985f36fd98413ecf729f7d69476c342851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136803
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27948}
2019-05-15 11:12:46 +00:00

65 lines
2.6 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
#include <algorithm>
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "test/gmock.h"
namespace webrtc {
class MockDelayManager : public DelayManager {
public:
MockDelayManager(size_t max_packets_in_buffer,
int base_min_target_delay_ms,
int histogram_quantile,
HistogramMode histogram_mode,
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer,
StatisticsCalculator* stats,
std::unique_ptr<Histogram> histogram)
: DelayManager(max_packets_in_buffer,
base_min_target_delay_ms,
histogram_quantile,
histogram_mode,
enable_rtx_handling,
peak_detector,
tick_timer,
stats,
std::move(histogram)) {}
virtual ~MockDelayManager() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD3(Update,
int(uint16_t sequence_number,
uint32_t timestamp,
int sample_rate_hz));
MOCK_METHOD2(CalculateTargetLevel, int(int iat_packets, bool reordered));
MOCK_METHOD1(SetPacketAudioLength, int(int length_ms));
MOCK_METHOD0(Reset, void());
MOCK_CONST_METHOD0(PeakFound, bool());
MOCK_METHOD0(ResetPacketIatCount, void());
MOCK_CONST_METHOD2(BufferLimits, void(int* lower_limit, int* higher_limit));
MOCK_METHOD1(SetBaseMinimumDelay, bool(int delay_ms));
MOCK_CONST_METHOD0(GetBaseMinimumDelay, int());
MOCK_CONST_METHOD0(TargetLevel, int());
MOCK_METHOD0(RegisterEmptyPacket, void());
MOCK_CONST_METHOD0(base_target_level, int());
MOCK_CONST_METHOD0(last_pack_cng_or_dtmf, int());
MOCK_METHOD1(set_last_pack_cng_or_dtmf, void(int value));
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_