| /* | |
| * 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_ENCODE_NETEQ_INPUT_H_ | |
| #define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <memory> | |
| #include <optional> | |
| #include <span> | |
| #include "api/audio_codecs/audio_encoder.h" | |
| #include "modules/audio_coding/neteq/tools/neteq_input.h" | |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" | |
| namespace webrtc { | |
| namespace test { | |
| // This class provides a NetEqInput that takes audio from a generator object and | |
| // encodes it using a given audio encoder. | |
| class EncodeNetEqInput : public NetEqInput { | |
| public: | |
| // Generator class, to be provided to the EncodeNetEqInput constructor. | |
| class Generator { | |
| public: | |
| virtual ~Generator() = default; | |
| // Returns the next num_samples values from the signal generator. | |
| virtual std::span<const int16_t> Generate(size_t num_samples) = 0; | |
| }; | |
| // The source will end after the given input duration. | |
| EncodeNetEqInput(std::unique_ptr<Generator> generator, | |
| std::unique_ptr<AudioEncoder> encoder, | |
| int64_t input_duration_ms); | |
| ~EncodeNetEqInput() override; | |
| std::optional<int64_t> NextPacketTime() const override; | |
| std::optional<int64_t> NextOutputEventTime() const override; | |
| std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override { | |
| return std::nullopt; | |
| } | |
| std::unique_ptr<RtpPacketReceived> PopPacket() override; | |
| void AdvanceOutputEvent() override; | |
| void AdvanceSetMinimumDelay() override {} | |
| bool ended() const override; | |
| const RtpPacketReceived* NextPacket() const override; | |
| private: | |
| static constexpr int64_t kOutputPeriodMs = 10; | |
| void CreatePacket(); | |
| std::unique_ptr<Generator> generator_; | |
| std::unique_ptr<AudioEncoder> encoder_; | |
| std::unique_ptr<RtpPacketReceived> packet_data_; | |
| uint32_t rtp_timestamp_ = 0; | |
| int16_t sequence_number_ = 0; | |
| int64_t next_packet_time_ms_ = 0; | |
| int64_t next_output_event_ms_ = 0; | |
| const int64_t input_duration_ms_; | |
| }; | |
| } // namespace test | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ |