webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
Ivo Creusen 7b463c5f67 Add a "Smart flushing" feature to NetEq.
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.

Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
2020-11-26 11:20:28 +00:00

82 lines
3 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_PACKET_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "test/gmock.h"
namespace webrtc {
class MockPacketBuffer : public PacketBuffer {
public:
MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
: PacketBuffer(max_number_of_packets, tick_timer) {}
~MockPacketBuffer() override { Die(); }
MOCK_METHOD(void, Die, ());
MOCK_METHOD(void, Flush, (StatisticsCalculator * stats), (override));
MOCK_METHOD(void,
PartialFlush,
(int target_level_ms,
size_t sample_rate,
size_t last_decoded_length,
StatisticsCalculator* stats),
(override));
MOCK_METHOD(bool, Empty, (), (const, override));
MOCK_METHOD(int,
InsertPacket,
(Packet && packet,
StatisticsCalculator* stats,
size_t last_decoded_length,
size_t sample_rate,
int target_level_ms,
const DecoderDatabase& decoder_database),
(override));
MOCK_METHOD(int,
InsertPacketList,
(PacketList * packet_list,
const DecoderDatabase& decoder_database,
absl::optional<uint8_t>* current_rtp_payload_type,
absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats,
size_t last_decoded_length,
size_t sample_rate,
int target_level_ms),
(override));
MOCK_METHOD(int,
NextTimestamp,
(uint32_t * next_timestamp),
(const, override));
MOCK_METHOD(int,
NextHigherTimestamp,
(uint32_t timestamp, uint32_t* next_timestamp),
(const, override));
MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
MOCK_METHOD(int,
DiscardNextPacket,
(StatisticsCalculator * stats),
(override));
MOCK_METHOD(void,
DiscardOldPackets,
(uint32_t timestamp_limit,
uint32_t horizon_samples,
StatisticsCalculator* stats),
(override));
MOCK_METHOD(void,
DiscardAllOldPackets,
(uint32_t timestamp_limit, StatisticsCalculator* stats),
(override));
MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override));
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_