/*

 * Sierra VMD audio decoder

 * Copyright (c) 2004 The FFmpeg Project

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

/**

 * @file

 * Sierra VMD audio decoder

 * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)

 * for more information on the Sierra VMD format, visit:

 *   http://www.pcisys.net/~melanson/codecs/

 *

 * The audio decoder, expects each encoded data

 * chunk to be prepended with the appropriate 16-byte frame information

 * record from the VMD file. It does not require the 0x330-byte VMD file

 * header, but it does need the audio setup parameters passed in through

 * normal libavcodec API means.

 */

#include <string.h>

#include "libavutil/avassert.h"

#include "libavutil/channel_layout.h"

#include "libavutil/common.h"

#include "libavutil/intreadwrite.h"

#include "avcodec.h"

#include "codec_internal.h"

#include "decode.h"

#define BLOCK_TYPE_AUDIO    1

#define BLOCK_TYPE_INITIAL  2

#define BLOCK_TYPE_SILENCE  3

typedef struct VmdAudioContext {

    int out_bps;

    int chunk_size;

} VmdAudioContext;

static const uint16_t vmdaudio_table[128] = {

    0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,

    0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,

    0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,

    0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,

    0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,

    0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,

    0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,

    0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,

    0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,

    0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,

    0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,

    0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,

    0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000

};

static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)

{

    VmdAudioContext *s = avctx->priv_data;

    int channels = avctx->ch_layout.nb_channels;

    if (channels < 1 || channels > 2) {

        av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");

        return AVERROR(EINVAL);

    }

    if (avctx->block_align < 1 || avctx->block_align % channels ||

        avctx->block_align > INT_MAX - channels) {

        av_log(avctx, AV_LOG_ERROR, "invalid block align\n");

        return AVERROR(EINVAL);

    }

    av_channel_layout_uninit(&avctx->ch_layout);

    av_channel_layout_default(&avctx->ch_layout, channels);

    if (avctx->bits_per_coded_sample == 16)

        avctx->sample_fmt = AV_SAMPLE_FMT_S16;

    else

        avctx->sample_fmt = AV_SAMPLE_FMT_U8;

    s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);

    s->chunk_size = avctx->block_align + channels * (s->out_bps == 2);

    av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "

           "block align = %d, sample rate = %d\n",

           channels, avctx->bits_per_coded_sample, avctx->block_align,

           avctx->sample_rate);

    return 0;

}

static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,

                             int channels)

{

    int ch;

    const uint8_t *buf_end = buf + buf_size;

    int predictor[2];

    int st = channels - 1;

    /* decode initial raw sample */

    for (ch = 0; ch < channels; ch++) {

        predictor[ch] = (int16_t)AV_RL16(buf);

        buf += 2;

        *out++ = predictor[ch];

    }

    /* decode DPCM samples */

    ch = 0;

    while (buf < buf_end) {

        uint8_t b = *buf++;

        if (b & 0x80)

            predictor[ch] -= vmdaudio_table[b & 0x7F];

        else

            predictor[ch] += vmdaudio_table[b];

        predictor[ch] = av_clip_int16(predictor[ch]);

        *out++ = predictor[ch];

        ch ^= st;

    }

}

static int vmdaudio_decode_frame(AVCodecContext *avctx, AVFrame *frame,

                                 int *got_frame_ptr, AVPacket *avpkt)

{

    const uint8_t *buf = avpkt->data;

    const uint8_t *buf_end;

    int buf_size = avpkt->size;

    VmdAudioContext *s = avctx->priv_data;

    int block_type, silent_chunks, audio_chunks;

    int ret;

    uint8_t *output_samples_u8;

    int16_t *output_samples_s16;

    int channels = avctx->ch_layout.nb_channels;

    if (buf_size < 16) {

        av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");

        *got_frame_ptr = 0;

        return buf_size;

    }

    block_type = buf[6];

    if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {

        av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);

        return AVERROR(EINVAL);

    }

    buf      += 16;

    buf_size -= 16;

    /* get number of silent chunks */

    silent_chunks = 0;

    if (block_type == BLOCK_TYPE_INITIAL) {

        uint32_t flags;

        if (buf_size < 4) {

            av_log(avctx, AV_LOG_ERROR, "packet is too small\n");

            return AVERROR(EINVAL);

        }

        flags         = AV_RB32(buf);

        silent_chunks = av_popcount(flags);

        buf      += 4;

        buf_size -= 4;

    } else if (block_type == BLOCK_TYPE_SILENCE) {

        silent_chunks = 1;

        buf_size = 0; // should already be zero but set it just to be sure

    }

    /* ensure output buffer is large enough */

    audio_chunks = buf_size / s->chunk_size;

    /* drop incomplete chunks */

    buf_size     = audio_chunks * s->chunk_size;

    if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)

        return AVERROR_INVALIDDATA;

    /* get output buffer */

    frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /

                        avctx->ch_layout.nb_channels;

    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

        return ret;

    output_samples_u8  =            frame->data[0];

    output_samples_s16 = (int16_t *)frame->data[0];

    /* decode silent chunks */

    if (silent_chunks > 0) {

        int silent_size = avctx->block_align * silent_chunks;

        av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels);

        if (s->out_bps == 2) {

            memset(output_samples_s16, 0x00, silent_size * 2);

            output_samples_s16 += silent_size;

        } else {

            memset(output_samples_u8,  0x80, silent_size);

            output_samples_u8 += silent_size;

        }

    }

    /* decode audio chunks */

    if (audio_chunks > 0) {

        buf_end = buf + buf_size;

        av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0);

        while (buf_end - buf >= s->chunk_size) {

            if (s->out_bps == 2) {

                decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels);

                output_samples_s16 += avctx->block_align;

            } else {

                memcpy(output_samples_u8, buf, s->chunk_size);

                output_samples_u8  += avctx->block_align;

            }

            buf += s->chunk_size;

        }

    }

    *got_frame_ptr = 1;

    return avpkt->size;

}

const FFCodec ff_vmdaudio_decoder = {

    .p.name         = "vmdaudio",

    CODEC_LONG_NAME("Sierra VMD audio"),

    .p.type         = AVMEDIA_TYPE_AUDIO,

    .p.id           = AV_CODEC_ID_VMDAUDIO,

    .priv_data_size = sizeof(VmdAudioContext),

    .init           = vmdaudio_decode_init,

    FF_CODEC_DECODE_CB(vmdaudio_decode_frame),

    .p.capabilities = AV_CODEC_CAP_DR1,

};