host/frontend/gcastv2/webrtc/include/webrtc/OpusPacketizer.h

/*
 * Copyright (C) 2019 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#pragma once

#include "Packetizer.h"

#include <https/RunLoop.h>

#include <memory>

#include <source/StreamingSource.h>

struct OpusPacketizer
    : public Packetizer, public std::enable_shared_from_this<OpusPacketizer> {

    using StreamingSource = android::StreamingSource;

    explicit OpusPacketizer(
            std::shared_ptr<RunLoop> runLoop,
            std::shared_ptr<StreamingSource> audioSource);

    void run() override;
    uint32_t rtpNow() const override;
    int32_t requestIDRFrame() override;

private:
    using SBuffer = android::SBuffer;

    std::shared_ptr<RunLoop> mRunLoop;

    std::shared_ptr<StreamingSource> mAudioSource;

    size_t mNumSamplesRead;

    std::chrono::time_point<std::chrono::steady_clock> mStartTimeReal;
    int64_t mStartTimeMedia;
    bool mFirstInTalkspurt;

    void onFrame(const std::shared_ptr<SBuffer> &accessUnit);

    void packetize(const std::shared_ptr<SBuffer> &accessUnit, int64_t timeUs);
};