webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

51 lines
1.9 KiB
C++

/*
* Copyright (c) 2016 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_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
#include <memory>
#include <set>
#include "modules/audio_coding/neteq/tools/neteq_input.h"
namespace webrtc {
namespace test {
// This class converts the packets from a NetEqInput to fake encodings to be
// decoded by a FakeDecodeFromFile decoder.
class NetEqReplacementInput : public NetEqInput {
public:
NetEqReplacementInput(std::unique_ptr<NetEqInput> source,
uint8_t replacement_payload_type,
const std::set<uint8_t>& comfort_noise_types,
const std::set<uint8_t>& forbidden_types);
rtc::Optional<int64_t> NextPacketTime() const override;
rtc::Optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
rtc::Optional<RTPHeader> NextHeader() const override;
private:
void ReplacePacket();
std::unique_ptr<NetEqInput> source_;
const uint8_t replacement_payload_type_;
const std::set<uint8_t> comfort_noise_types_;
const std::set<uint8_t> forbidden_types_;
std::unique_ptr<PacketData> packet_; // The next packet to deliver.
uint32_t last_frame_size_timestamps_ = 960; // Initial guess: 20 ms @ 48 kHz.
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_