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/*
* 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 "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/rtp_headers.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 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;
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<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
void AdvanceSetMinimumDelay() override {}
bool ended() const override;
std::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_
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