TeaSpeak-Client/native/serverconnection/src/audio/codec/Converter.h

#pragma once

#include <string>

#if !defined(ssize_t) && defined(WIN32)
    #define ssize_t int64_t
#endif

namespace tc::audio::codec {
    namespace type {
        enum value {
            undefined,

            /* supported */
            opus,
            speex,

            /* unsupported */
            flac,
            celt
        };

        extern bool supported(value);
    }

    class Converter {
        public:
            Converter(size_t /* channels */, size_t /* sample rate */, size_t /* frame size */);
            virtual ~Converter();

            /* initialize parameters depend on the codec */
            virtual bool valid() = 0;
            virtual void finalize() = 0;

            virtual void reset_encoder() = 0;
            virtual void reset_decoder() = 0;

            /**
             * @return number of bytes written on success
             */
            virtual ssize_t encode(std::string& /* error */, const void* /* source */, void* /* destination */, size_t /* destination byte length */, bool /* head package */) = 0;

            /**
             * @return number of samples on success
             */
            virtual ssize_t decode(std::string& /* error */, const void* /* source */, size_t /* source byte length */, void* /* destination */, bool /* fec decoding */) = 0;
            virtual ssize_t decode_lost(std::string& /* error */, size_t /* packets */) = 0;

            virtual size_t expected_encoded_length(size_t /* sample count */) = 0;
            virtual size_t expected_decoded_length(const void* /* source */, size_t /* source byte length */) {
                return this->bytes_per_frame();
            }

            inline size_t channels() { return this->_channels; }
            inline size_t sample_rate() { return this->_sample_rate; }
            inline size_t frame_size() { return this->_frame_size; }

            inline size_t bytes_per_frame() { return this->_channels * this->_frame_size * 4; }
        protected:
            size_t _frame_size;
            size_t _channels;
            size_t _sample_rate;
    };

    struct EncoderBufferInfo {
        size_t sample_count{0};
        bool head_sequence{false};
        bool flush_encoder{false};
    };

    /**
     *  Encoders should only be accessed by one thread at once
     */
    class AudioEncoder {
        public:
            explicit AudioEncoder() = default;
            virtual ~AudioEncoder() = default;

            [[nodiscard]] virtual bool valid() const = 0;
            [[nodiscard]] virtual bool initialize(std::string& /* error */) = 0;
            virtual void reset_sequence() = 0;

            /**
             * Get the codecs sample rate.
             */
            [[nodiscard]] virtual size_t sample_rate() const = 0;

            /**
             * Get the codecs audio channel count.
             */
            [[nodiscard]] virtual size_t channel_count() const = 0;


            /**
             * @returns the expected output length.
             *          If unknown a size near the MTU will be returned.
             */
            [[nodiscard]] virtual size_t expected_encoded_length(const float* /* samples */, size_t /* sample count */) const = 0;

            /**
             * Encode a chunk of audio data.
             * @return `true` on success else `false`
             */
            [[nodiscard]] virtual bool encode(std::string& /* error */, void* /* target buffer */, size_t& /* target length */, const EncoderBufferInfo& /* buffer info */, const float* /* samples */) = 0;
    };

    class AudioDecoder {};
}