/***************************************************************************
 *             __________               __   ___.
 *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
 *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
 *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
 *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
 *                     \/            \/     \/    \/            \/
 * $Id$
 *
 * Copyright (C) 2005 Dave Chapman
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
 * as published by the Free Software Foundation; either version 2
 * of the License, or (at your option) any later version.
 *
 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
 * KIND, either express or implied.
 *
 ****************************************************************************/

#include "codeclib.h"
#include "libm4a/m4a.h"
#include "libfaad/common.h"
#include "libfaad/structs.h"
#include "libfaad/decoder.h"

CODEC_HEADER

/* Global buffers to be used in the mdct synthesis. This way the arrays can
 * be moved to IRAM for some targets */
#define GB_BUF_SIZE 1024
static ALIGN real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
static ALIGN real_t gb_fb_intermed[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;

/* this is the codec entry point */
enum codec_status codec_main(void)
{
    /* Note that when dealing with QuickTime/MPEG4 files, terminology is
     * a bit confusing. Files with sound are split up in chunks, where
     * each chunk contains one or more samples. Each sample in turn
     * contains a number of "sound samples" (the kind you refer to with
     * the sampling frequency).
     */
    size_t n;
    demux_res_t demux_res;
    stream_t input_stream;
    uint32_t sound_samples_done;
    uint32_t elapsed_time;
    uint32_t sample_duration;
    uint32_t sample_byte_size;
    int file_offset;
    int framelength;
    int lead_trim = 0;
    int needed_bufsize;
    unsigned int i;
    unsigned char* buffer;
    NeAACDecFrameInfo frame_info;
    NeAACDecHandle decoder;
    int err;
    uint32_t s = 0;
    unsigned char c = 0;
    void *ret;

    /* Generic codec initialisation */
    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);

next_track:
    err = CODEC_OK;

    /* Clean and initialize decoder structures */
    memset(&demux_res , 0, sizeof(demux_res));
    if (codec_init()) {
        LOGF("FAAD: Codec init error\n");
        err = CODEC_ERROR;
        goto exit;
    }

    while (!*ci->taginfo_ready && !ci->stop_codec)
        ci->sleep(1);
  
    file_offset = ci->id3->offset;

    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);

    stream_create(&input_stream,ci);

    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(&input_stream, &demux_res)) {
        LOGF("FAAD: File init error\n");
        err = CODEC_ERROR;
        goto done;
    }

    /* initialise the sound converter */
    decoder = NeAACDecOpen();

    if (!decoder) {
        LOGF("FAAD: Decode open error\n");
        err = CODEC_ERROR;
        goto done;
    }

    NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
    conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
    NeAACDecSetConfiguration(decoder, conf);

    err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
    if (err) {
        LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
        err = CODEC_ERROR;
        goto done;
    }
    
    /* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
     * be called after NeAACDecOpen(). */
    /* A buffer of framelength or 2*frameLenght size must be allocated for
     * time_out. If frameLength is too big or SBR/forceUpSampling is active, 
     * we do not use the IRAM buffer and keep faad's internal allocation (see 
     * specrec.c). */
    needed_bufsize = decoder->frameLength;
#ifdef SBR_DEC
    if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
    {
        needed_bufsize *= 2;
    }
#endif
    if (needed_bufsize <= GB_BUF_SIZE)
    {
        decoder->time_out[0]    = &gb_time_buffer[0][0];
        decoder->time_out[1]    = &gb_time_buffer[1][0];
    }
    /* A buffer of with frameLength elements must be allocated for fb_intermed. 
     * If frameLength is too big, we do not use the IRAM buffer and keep faad's 
     * internal allocation (see specrec.c). */
    needed_bufsize = decoder->frameLength;
    if (needed_bufsize <= GB_BUF_SIZE)
    {
        decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
        decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
    }

    ci->id3->frequency = s;

    i = 0;
    
    if (file_offset > 0) {
        if (alac_seek_raw(&demux_res, &input_stream, file_offset,
                          &sound_samples_done, (int*) &i)) {
            elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
            ci->set_elapsed(elapsed_time);
        } else {
            sound_samples_done = 0;
        }
    } else {
        sound_samples_done = 0;
    }
    
    if (i == 0) 
    {
        lead_trim = ci->id3->lead_trim;
    }

    /* The main decoding loop */
    while (i < demux_res.num_sample_byte_sizes) {
        ci->yield();

        if (ci->stop_codec || ci->new_track) {
            break;
        }

        /* Deal with any pending seek requests */
        if (ci->seek_time) {
            if (alac_seek(&demux_res, &input_stream,
                          ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
                          &sound_samples_done, (int*) &i)) {
                elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
                ci->set_elapsed(elapsed_time);
            
                if (i == 0) 
                {
                    lead_trim = ci->id3->lead_trim;
                }
            }
            ci->seek_complete();
        }

        /* Lookup the length (in samples and bytes) of block i */
        if (!get_sample_info(&demux_res, i, &sample_duration, 
                             &sample_byte_size)) {
            LOGF("AAC: get_sample_info error\n");
            err = CODEC_ERROR;
            goto done;
        }

        /* There can be gaps between chunks, so skip ahead if needed. It
         * doesn't seem to happen much, but it probably means that a 
         * "proper" file can have chunks out of order. Why one would want
         * that an good question (but files with gaps do exist, so who 
         * knows?), so we don't support that - for now, at least.
         */        
        file_offset = get_sample_offset(&demux_res, i);
        
        if (file_offset > ci->curpos)
        {
            ci->advance_buffer(file_offset - ci->curpos);
        }
        else if (file_offset == 0)
        {
            LOGF("AAC: get_sample_offset error\n");
            err = CODEC_ERROR;
            goto done;
        }
        
        /* Request the required number of bytes from the input buffer */
        buffer=ci->request_buffer(&n,sample_byte_size);

        /* Decode one block - returned samples will be host-endian */
        ret = NeAACDecDecode(decoder, &frame_info, buffer, n);

        /* NeAACDecDecode may sometimes return NULL without setting error. */
        if (ret == NULL || frame_info.error > 0) {
            LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
            err = CODEC_ERROR;
            goto done;
        }

        /* Advance codec buffer (no need to call set_offset because of this) */
        ci->advance_buffer(n);

        /* Output the audio */
        ci->yield();
        
        framelength = (frame_info.samples >> 1) - lead_trim;
        
        if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
        {
            /* Currently limited to at most one frame of tail_trim.
             * Seems to be enough.
             */
            if (ci->id3->tail_trim == 0 
                && sample_duration < (frame_info.samples >> 1))
            {
                /* Subtract lead_trim just in case we decode a file with
                 * only one audio frame with actual data.
                 */
                framelength = sample_duration - lead_trim;
            }
            else
            {
                framelength -= ci->id3->tail_trim;
            }
        }

        if (framelength > 0)
        {
            ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
                              &decoder->time_out[1][lead_trim],
                              framelength);
        } 
        
        if (lead_trim > 0)
        {
            /* frame_info.samples can be 0 for the first frame */
            lead_trim -= (i > 0 || frame_info.samples)
                ? (frame_info.samples >> 1) : sample_duration;

            if (lead_trim < 0 || ci->id3->lead_trim == 0)
            {
                lead_trim = 0;
            }
        }

        /* Update the elapsed-time indicator */
        sound_samples_done += sample_duration;
        elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
        ci->set_elapsed(elapsed_time);
        i++;
    }

    err = CODEC_OK;

done:
    LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);

    if (ci->request_next_track())
        goto next_track;

exit:
    return err;
}