| /* |
| * 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 <memory> |
| |
| #include "api/audio_codecs/audio_encoder.h" |
| #include "modules/audio_coding/neteq/tools/neteq_input.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 rtc::ArrayView<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; |
| |
| 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; |
| |
| private: |
| static constexpr int64_t kOutputPeriodMs = 10; |
| |
| void CreatePacket(); |
| |
| std::unique_ptr<Generator> generator_; |
| std::unique_ptr<AudioEncoder> encoder_; |
| std::unique_ptr<PacketData> 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_ |