| /* |
| * 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_ |
| |
| #include <memory> |
| #include <set> |
| |
| #include "webrtc/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 // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_ |