// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.

#include <algorithm>
#include <array>
#include "audio_core/codec.h"
#include "audio_core/hle/common.h"
#include "audio_core/hle/source.h"
#include "audio_core/interpolate.h"
#include "common/assert.h"
#include "common/logging/log.h"
#include "core/memory.h"

namespace AudioCore::HLE {

SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config,
                                  const s16_le (&adpcm_coeffs)[16]) {
    ParseConfig(config, adpcm_coeffs);

    if (state.enabled) {
        GenerateFrame();
    }

    return GetCurrentStatus();
}

void Source::MixInto(QuadFrame32& dest, std::size_t intermediate_mix_id) const {
    if (!state.enabled)
        return;

    const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id);
    for (std::size_t samplei = 0; samplei < samples_per_frame; samplei++) {
        // Conversion from stereo (current_frame) to quadraphonic (dest) occurs here.
        dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]);
        dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]);
        dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]);
        dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]);
    }
}

void Source::Reset() {
    current_frame.fill({});
    state = {};
}

void Source::SetMemory(Memory::MemorySystem& memory) {
    memory_system = &memory;
}

void Source::ParseConfig(SourceConfiguration::Configuration& config,
                         const s16_le (&adpcm_coeffs)[16]) {
    if (!config.dirty_raw) {
        return;
    }

    if (config.reset_flag) {
        config.reset_flag.Assign(0);
        Reset();
        LOG_TRACE(Audio_DSP, "source_id={} reset", source_id);
    }

    if (config.partial_reset_flag) {
        config.partial_reset_flag.Assign(0);
        state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{};
        LOG_TRACE(Audio_DSP, "source_id={} partial_reset", source_id);
    }

    if (config.enable_dirty) {
        config.enable_dirty.Assign(0);
        state.enabled = config.enable != 0;
        LOG_TRACE(Audio_DSP, "source_id={} enable={}", source_id, state.enabled);
    }

    if (config.sync_count_dirty) {
        config.sync_count_dirty.Assign(0);
        state.sync_count = config.sync_count;
        LOG_TRACE(Audio_DSP, "source_id={} sync={}", source_id, state.sync_count);
    }

    if (config.rate_multiplier_dirty) {
        config.rate_multiplier_dirty.Assign(0);
        state.rate_multiplier = config.rate_multiplier;
        LOG_TRACE(Audio_DSP, "source_id={} rate={}", source_id, state.rate_multiplier);

        if (state.rate_multiplier <= 0) {
            LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id={} rate={}",
                      source_id, state.rate_multiplier);
            state.rate_multiplier = 1.0f;
            // Note: Actual firmware starts producing garbage if this occurs.
        }
    }

    if (config.adpcm_coefficients_dirty) {
        config.adpcm_coefficients_dirty.Assign(0);
        std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(),
                       state.adpcm_coeffs.begin(),
                       [](const auto& coeff) { return static_cast<s16>(coeff); });
        LOG_TRACE(Audio_DSP, "source_id={} adpcm update", source_id);
    }

    if (config.gain_0_dirty) {
        config.gain_0_dirty.Assign(0);
        std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
                       [](const auto& coeff) { return static_cast<float>(coeff); });
        LOG_TRACE(Audio_DSP, "source_id={} gain 0 update", source_id);
    }

    if (config.gain_1_dirty) {
        config.gain_1_dirty.Assign(0);
        std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
                       [](const auto& coeff) { return static_cast<float>(coeff); });
        LOG_TRACE(Audio_DSP, "source_id={} gain 1 update", source_id);
    }

    if (config.gain_2_dirty) {
        config.gain_2_dirty.Assign(0);
        std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
                       [](const auto& coeff) { return static_cast<float>(coeff); });
        LOG_TRACE(Audio_DSP, "source_id={} gain 2 update", source_id);
    }

    if (config.filters_enabled_dirty) {
        config.filters_enabled_dirty.Assign(0);
        state.filters.Enable(static_cast<bool>(config.simple_filter_enabled),
                             static_cast<bool>(config.biquad_filter_enabled));
        LOG_TRACE(Audio_DSP, "source_id={} enable_simple={} enable_biquad={}", source_id,
                  config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
    }

    if (config.simple_filter_dirty) {
        config.simple_filter_dirty.Assign(0);
        state.filters.Configure(config.simple_filter);
        LOG_TRACE(Audio_DSP, "source_id={} simple filter update", source_id);
    }

    if (config.biquad_filter_dirty) {
        config.biquad_filter_dirty.Assign(0);
        state.filters.Configure(config.biquad_filter);
        LOG_TRACE(Audio_DSP, "source_id={} biquad filter update", source_id);
    }

    if (config.interpolation_dirty) {
        config.interpolation_dirty.Assign(0);
        state.interpolation_mode = config.interpolation_mode;
        LOG_TRACE(Audio_DSP, "source_id={} interpolation_mode={}", source_id,
                  static_cast<std::size_t>(state.interpolation_mode));
    }

    if (config.format_dirty || config.embedded_buffer_dirty) {
        config.format_dirty.Assign(0);
        state.format = config.format;
        LOG_TRACE(Audio_DSP, "source_id={} format={}", source_id,
                  static_cast<std::size_t>(state.format));
    }

    if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
        config.mono_or_stereo_dirty.Assign(0);
        state.mono_or_stereo = config.mono_or_stereo;
        LOG_TRACE(Audio_DSP, "source_id={} mono_or_stereo={}", source_id,
                  static_cast<std::size_t>(state.mono_or_stereo));
    }

    u32_dsp play_position = {};
    if (config.play_position_dirty && config.play_position != 0) {
        config.play_position_dirty.Assign(0);
        play_position = config.play_position;
        // play_position applies only to the embedded buffer, and defaults to 0 w/o a dirty bit
        // This will be the starting sample for the first time the buffer is played.
    }

    // TODO(xperia64): Is this in the correct spot in terms of the bit handling order?
    if (config.partial_embedded_buffer_dirty) {
        config.partial_embedded_buffer_dirty.Assign(0);

        // As this bit is set by the game, three config options are also updated:
        // buffer_id (after a check comparing the buffer_id to something, probably to make sure it's
        // the same buffer?), flags2_raw.is_looping, and length.

        // A quick and dirty way of extending the current buffer is to just read the whole thing
        // again with the new length. Note that this uses the latched physical address instead of
        // whatever is in config, because that may be invalid.
        const u8* const memory =
            memory_system->GetPhysicalPointer(state.current_buffer_physical_address & 0xFFFFFFFC);

        // TODO(xperia64): This could potentially be optimized by only decoding the new data and
        // appending that to the buffer.
        if (memory) {
            const unsigned num_channels = state.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
            bool valid = false;
            switch (state.format) {
            case Format::PCM8:
                // TODO(xperia64): This may just work fine like PCM16, but I haven't tested and
                // couldn't find any test case games
                UNIMPLEMENTED_MSG("{} not handled for partial buffer updates", "PCM8");
                // state.current_buffer = Codec::DecodePCM8(num_channels, memory, config.length);
                break;
            case Format::PCM16:
                state.current_buffer = Codec::DecodePCM16(num_channels, memory, config.length);
                valid = true;
                break;
            case Format::ADPCM:
                // TODO(xperia64): Are partial embedded buffer updates even valid for ADPCM? What
                // about the adpcm state?
                UNIMPLEMENTED_MSG("{} not handled for partial buffer updates", "ADPCM");
                /* state.current_buffer =
                    Codec::DecodeADPCM(memory, config.length, state.adpcm_coeffs,
                   state.adpcm_state); */
                break;
            default:
                UNIMPLEMENTED();
                break;
            }

            // Again, because our interpolation consumes samples instead of using an index, let's
            // just re-consume the samples up to the current sample number. There may be some
            // imprecision here with the current sample number, as Detective Pikachu sounds a little
            // rough at times.
            if (valid) {

                // TODO(xperia64): Tomodachi life apparently can decrease config.length when the
                // user skips dialog. I don't know the correct behavior, but to avoid crashing, just
                // reset the current sample number to 0 and don't try to truncate the buffer
                if (state.current_buffer.size() < state.current_sample_number) {
                    state.current_sample_number = 0;
                } else {
                    state.current_buffer.erase(
                        state.current_buffer.begin(),
                        std::next(state.current_buffer.begin(), state.current_sample_number));
                }
            }
        }
        LOG_TRACE(Audio_DSP, "partially updating embedded buffer addr={:#010x} len={} id={}",
                  state.current_buffer_physical_address, static_cast<u32>(config.length),
                  config.buffer_id);
    }

    if (config.embedded_buffer_dirty) {
        config.embedded_buffer_dirty.Assign(0);
        // HACK
        // Luigi's Mansion Dark Moon configures the embedded buffer with an extremely large value
        // for length, causing the Dequeue method to allocate a buffer of that size, eating up all
        // of the users RAM. It appears that the game is calculating the length of the sample by
        // using some value from the DSP and subtracting another value, which causes it to
        // underflow. We need to investigate further into what value the game is reading from and
        // fix that, but as a stop gap, we can just prevent these underflowed values from playing in
        // the mean time
        if (static_cast<s32>(config.length) < 0) {
            LOG_ERROR(Audio_DSP,
                      "Skipping embedded buffer sample! Game passed in improper value for length. "
                      "addr {:X} length {:X}",
                      static_cast<u32>(config.physical_address), static_cast<u32>(config.length));
        } else {
            state.input_queue.emplace(Buffer{
                config.physical_address,
                config.length,
                static_cast<u8>(config.adpcm_ps),
                {config.adpcm_yn[0], config.adpcm_yn[1]},
                static_cast<bool>(config.adpcm_dirty),
                static_cast<bool>(config.is_looping),
                config.buffer_id,
                state.mono_or_stereo,
                state.format,
                false,
                play_position,
                false,
            });
        }
        LOG_TRACE(Audio_DSP, "enqueuing embedded addr={:#010x} len={} id={} start={}",
                  static_cast<u32>(config.physical_address), static_cast<u32>(config.length),
                  config.buffer_id, static_cast<u32>(config.play_position));
    }

    if (config.loop_related_dirty && config.loop_related != 0) {
        config.loop_related_dirty.Assign(0);
        LOG_WARNING(Audio_DSP, "Unhandled complex loop with loop_related={:#010x}",
                    static_cast<u32>(config.loop_related));
    }

    if (config.buffer_queue_dirty) {
        config.buffer_queue_dirty.Assign(0);
        for (std::size_t i = 0; i < 4; i++) {
            if (config.buffers_dirty & (1 << i)) {
                const auto& b = config.buffers[i];
                if (static_cast<s32>(b.length) < 0) {
                    LOG_ERROR(Audio_DSP,
                              "Skipping buffer queue sample! Game passed in improper value for "
                              "length. addr {:X} length {:X}",
                              static_cast<u32>(b.physical_address), static_cast<u32>(b.length));
                } else {
                    state.input_queue.emplace(Buffer{
                        b.physical_address,
                        b.length,
                        static_cast<u8>(b.adpcm_ps),
                        {b.adpcm_yn[0], b.adpcm_yn[1]},
                        b.adpcm_dirty != 0,
                        b.is_looping != 0,
                        b.buffer_id,
                        state.mono_or_stereo,
                        state.format,
                        true,
                        {}, // 0 in u32_dsp
                        false,
                    });
                }
                LOG_TRACE(Audio_DSP, "enqueuing queued {} addr={:#010x} len={} id={}", i,
                          static_cast<u32>(b.physical_address), static_cast<u32>(b.length),
                          b.buffer_id);
            }
        }
        config.buffers_dirty = 0;
    }

    if (config.dirty_raw) {
        LOG_DEBUG(Audio_DSP, "source_id={} remaining_dirty={:x}", source_id, config.dirty_raw);
    }

    config.dirty_raw = 0;
}

void Source::GenerateFrame() {
    current_frame.fill({});

    if (state.current_buffer.empty() && !DequeueBuffer()) {
        state.enabled = false;
        state.buffer_update = true;
        state.current_buffer_id = 0;
        return;
    }

    std::size_t frame_position = 0;

    state.current_sample_number = state.next_sample_number;
    while (frame_position < current_frame.size()) {
        if (state.current_buffer.empty() && !DequeueBuffer()) {
            break;
        }

        switch (state.interpolation_mode) {
        case InterpolationMode::None:
            AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier,
                              current_frame, frame_position);
            break;
        case InterpolationMode::Linear:
            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
                                current_frame, frame_position);
            break;
        case InterpolationMode::Polyphase:
            // TODO(merry): Implement polyphase interpolation
            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
                                current_frame, frame_position);
            break;
        default:
            UNIMPLEMENTED();
            break;
        }
    }
    // TODO(jroweboy): Keep track of frame_position independently so that it doesn't lose precision
    // over time
    state.next_sample_number += static_cast<u32>(frame_position * state.rate_multiplier);

    state.filters.ProcessFrame(current_frame);
}

bool Source::DequeueBuffer() {
    ASSERT_MSG(state.current_buffer.empty(),
               "Shouldn't dequeue; we still have data in current_buffer");

    if (state.input_queue.empty())
        return false;

    Buffer buf = state.input_queue.top();
    state.input_queue.pop();

    if (buf.adpcm_dirty) {
        state.adpcm_state.yn1 = buf.adpcm_yn[0];
        state.adpcm_state.yn2 = buf.adpcm_yn[1];
    }

    // This physical address masking occurs due to how the DSP DMA hardware is configured by the
    // firmware.
    const u8* const memory = memory_system->GetPhysicalPointer(buf.physical_address & 0xFFFFFFFC);
    if (memory) {
        const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
        switch (buf.format) {
        case Format::PCM8:
            state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
            break;
        case Format::PCM16:
            state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
            break;
        case Format::ADPCM:
            DEBUG_ASSERT(num_channels == 1);
            state.current_buffer =
                Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
            break;
        default:
            UNIMPLEMENTED();
            break;
        }
    } else {
        LOG_WARNING(Audio_DSP,
                    "source_id={} buffer_id={} length={}: Invalid physical address {:#010x}",
                    source_id, buf.buffer_id, buf.length, buf.physical_address);
        state.current_buffer.clear();
        return true;
    }

    // the first playthrough starts at play_position, loops start at the beginning of the buffer
    state.current_sample_number = (!buf.has_played) ? buf.play_position : 0;
    state.next_sample_number = state.current_sample_number;
    state.current_buffer_physical_address = buf.physical_address;
    state.current_buffer_id = buf.buffer_id;
    state.buffer_update = buf.from_queue && !buf.has_played;

    if (buf.is_looping) {
        buf.has_played = true;
        state.input_queue.push(buf);
    }

    LOG_TRACE(Audio_DSP, "source_id={} buffer_id={} from_queue={} current_buffer.size()={}",
              source_id, buf.buffer_id, buf.from_queue, state.current_buffer.size());
    return true;
}

SourceStatus::Status Source::GetCurrentStatus() {
    SourceStatus::Status ret;

    // Applications depend on the correct emulation of
    // current_buffer_id_dirty and current_buffer_id to synchronise
    // audio with video.
    ret.is_enabled = state.enabled;
    ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0;
    state.buffer_update = false;
    ret.current_buffer_id = state.current_buffer_id;
    ret.buffer_position = state.current_sample_number;
    ret.sync_count = state.sync_count;

    return ret;
}

} // namespace AudioCore::HLE
