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

/*
 *  Copyright (c) 2017 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/neteq_input.h"

#include "rtc_base/strings/string_builder.h"

namespace webrtc {
namespace test {

NetEqInput::PacketData::PacketData() = default;
NetEqInput::PacketData::~PacketData() = default;

std::string NetEqInput::PacketData::ToString() const {}

TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
                                             int64_t duration_ms)
    :{}

TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;

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

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

absl::optional<NetEqInput::SetMinimumDelayInfo>
TimeLimitedNetEqInput::NextSetMinimumDelayInfo() const {}

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

void TimeLimitedNetEqInput::AdvanceOutputEvent() {}

void TimeLimitedNetEqInput::AdvanceSetMinimumDelay() {}

bool TimeLimitedNetEqInput::ended() const {}

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

void TimeLimitedNetEqInput::MaybeSetEnded() {}

}  // namespace test
}  // namespace webrtc