blob: 9ce9b9dc63559ae8b9aea6e954a671aa71a9a569 [file] [log] [blame]
* 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 <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 {
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);
absl::optional<int64_t> NextPacketTime() const override;
absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
absl::optional<RTPHeader> NextHeader() const override;
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