chromium/third_party/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc

/*
 *  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.
 */

#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"

#include <utility>

#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {
namespace test {

EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
                                   std::unique_ptr<AudioEncoder> encoder,
                                   int64_t input_duration_ms)
    :{}

EncodeNetEqInput::~EncodeNetEqInput() = default;

absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {}

absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {}

std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {}

void EncodeNetEqInput::AdvanceOutputEvent() {}

bool EncodeNetEqInput::ended() const {}

absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {}

void EncodeNetEqInput::CreatePacket() {}

}  // namespace test
}  // namespace webrtc