/* * 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 #include #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 source, uint8_t replacement_payload_type, const std::set& comfort_noise_types, const std::set& forbidden_types); absl::optional NextPacketTime() const override; absl::optional NextOutputEventTime() const override; std::unique_ptr PopPacket() override; void AdvanceOutputEvent() override; bool ended() const override; absl::optional NextHeader() const override; private: void ReplacePacket(); std::unique_ptr source_; const uint8_t replacement_payload_type_; const std::set comfort_noise_types_; const std::set forbidden_types_; std::unique_ptr 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_