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path: root/lib/rbcodec/codecs/aac.c
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/***************************************************************************
 *             __________               __   ___.
 *   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

/* The maximum buffer size handled by faad. 12 bytes are required by libfaad
 * as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
 * for each frame. */
#define FAAD_BYTE_BUFFER_SIZE (2048-12)

/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
    if (reason == CODEC_LOAD) {
        /* Generic codec initialisation */
        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
    }

    return CODEC_OK;
}

/* this is called for each file to process */
enum codec_status codec_run(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;
    uint64_t sound_samples_done;
    uint32_t elapsed_time;
    int file_offset;
    int framelength;
    int lead_trim = 0;
    unsigned int frame_samples;
    uint32_t i;
    unsigned char* buffer;
    NeAACDecFrameInfo frame_info;
    NeAACDecHandle decoder;
    int err;
    uint32_t seek_idx = 0;
    uint32_t s = 0;
    uint32_t sbr_fac = 1;
    unsigned char c = 0;
    void *ret;
    long action;
    intptr_t param;
    bool empty_first_frame = false;

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

    action = CODEC_ACTION_NULL;
    param = ci->id3->elapsed;
    file_offset = ci->id3->offset;

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

    stream_create(&input_stream,ci);

    ci->seek_buffer(0);

    /* 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");
        return CODEC_ERROR;
    }

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

    if (!decoder) {
        LOGF("FAAD: Decode open error\n");
        return CODEC_ERROR;
    }

    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);
        return CODEC_ERROR;
    }

#ifdef SBR_DEC
    /* Check for need of special handling for seek/resume and elapsed time. */
    if (ci->id3->needs_upsampling_correction) {
        sbr_fac = 2;
    } else {
        sbr_fac = 1;
    }
#endif

    i = 0;

    if (param) {
        elapsed_time = param;
        action = CODEC_ACTION_SEEK_TIME;
    } else if (file_offset > 0) {
        /* Resume the desired (byte) position. Important: When resuming SBR
         * upsampling files the resulting sound_samples_done must be expanded
         * by a factor of 2. This is done via using sbr_fac. */
        if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
                         &sound_samples_done, &i, &seek_idx)) {
            sound_samples_done *= sbr_fac;
        } else {
            sound_samples_done = 0;
        }
        NeAACDecPostSeekReset(decoder, i);
        elapsed_time = sound_samples_done * 1000LL / ci->id3->frequency;
    } else {
        elapsed_time = 0;
        sound_samples_done = 0;
    }

    ci->set_elapsed(elapsed_time);

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

    /* The main decoding loop */
    while (i < demux_res.num_sample_byte_sizes) {
        if (action == CODEC_ACTION_NULL)
            action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        /* Deal with any pending seek requests */
        if (action == CODEC_ACTION_SEEK_TIME) {
            /* Seek to the desired time position. Important: When seeking in SBR
             * upsampling files the seek_time must be divided by 2 when calling
             * m4a_seek and the resulting sound_samples_done must be expanded
             * by a factor 2. This is done via using sbr_fac. */
            if (m4a_seek(&demux_res, &input_stream,
                         (uint64_t) param * ci->id3->frequency / sbr_fac / 1000ULL,
                         &sound_samples_done, &i, &seek_idx)) {
                sound_samples_done *= sbr_fac;
                elapsed_time = sound_samples_done * 1000LL / ci->id3->frequency;
                ci->set_elapsed(elapsed_time);

                if (i == 0)
                {
                    lead_trim = ci->id3->lead_trim;
                }
            }
            NeAACDecPostSeekReset(decoder, i);
            ci->seek_complete();
            if (i >= demux_res.num_sample_byte_sizes)
                break;
        }

        action = CODEC_ACTION_NULL;

        /* 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?), and we might not properly support it.
         * Metadata can also be placed after audio data so skip back if needed.
         */
        file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);

        if (file_offset > 0 && file_offset != ci->curpos)
        {
            ci->seek_buffer(file_offset);
        }

        /* Request the required number of bytes from the input buffer */
        buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_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));

            // In files with gaps between chunks and reduced lookup_table we can't properly detect all gaps
            // in m4a_check_sample_offset.  So just ignore decode errors till next chunk present in lookup_table 
            if (file_offset > 0)
                return CODEC_ERROR;
        }

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

        /* Output the audio */
        ci->yield();

        frame_samples = frame_info.samples >> 1;

        if (empty_first_frame)
        {
            /* Remove the first frame from lead_trim, under the assumption
             * that it had the same size as this frame
             */
            empty_first_frame = false;
            lead_trim -= frame_samples;

            if (lead_trim < 0)
            {
                lead_trim = 0;
            }
        }

        /* Gather number of samples for the decoded frame. */
        framelength = frame_samples - lead_trim;

        if (i == demux_res.num_sample_byte_sizes - 1)
        {
            // Size of the last frame
            const uint32_t sample_duration = (demux_res.num_time_to_samples > 0) ?
                demux_res.time_to_sample[demux_res.num_time_to_samples - 1].sample_duration :
                frame_samples;

            /* Currently limited to at most one frame of tail_trim.
             * Seems to be enough.
             */
            if (ci->id3->tail_trim == 0 && sample_duration < frame_samples)
            {
                /* Subtract lead_trim just in case we decode a file with only
                 * one audio frame with actual data (lead_trim is usually zero
                 * here).
                 */
                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);
            sound_samples_done += framelength;
            /* Update the elapsed-time indicator */
            elapsed_time = ((uint64_t) sound_samples_done * 1000) /
                ci->id3->frequency;
            ci->set_elapsed(elapsed_time);
        }

        if (lead_trim > 0)
        {
            /* frame_info.samples can be 0 for frame 0. We still want to
             * remove it from lead_trim, so do that during frame 1.
             */
            if (0 == i && 0 == frame_info.samples)
            {
                empty_first_frame = true;
            }

            lead_trim -= frame_samples;

            if (lead_trim < 0)
            {
                lead_trim = 0;
            }
        }

        ++i;
    }

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