rockbox/apps/codecs/libm4a/m4a.c
Daniel Stenberg 2acc0ac542 Updated our source code header to explicitly mention that we are GPL v2 or
later. We still need to hunt down snippets used that are not. 1324 modified
files...
http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
2008-06-28 18:10:04 +00:00

432 lines
12 KiB
C

/***************************************************************************
* __________ __ ___.
* 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 <codecs.h>
#include <inttypes.h>
#include "m4a.h"
#if defined(DEBUG) || defined(SIMULATOR)
extern struct codec_api* rb;
#define DEBUGF rb->debugf
#else
#define DEBUGF(...)
#endif
/* Implementation of the stream.h functions used by libalac */
#define _Swap32(v) do { \
v = (((v) & 0x000000FF) << 0x18) | \
(((v) & 0x0000FF00) << 0x08) | \
(((v) & 0x00FF0000) >> 0x08) | \
(((v) & 0xFF000000) >> 0x18); } while(0)
#define _Swap16(v) do { \
v = (((v) & 0x00FF) << 0x08) | \
(((v) & 0xFF00) >> 0x08); } while (0)
/* A normal read without any byte-swapping */
void stream_read(stream_t *stream, size_t size, void *buf)
{
stream->ci->read_filebuf(buf,size);
if (stream->ci->curpos >= stream->ci->filesize) { stream->eof=1; }
}
int32_t stream_read_int32(stream_t *stream)
{
int32_t v;
stream_read(stream, 4, &v);
#ifdef ROCKBOX_LITTLE_ENDIAN
_Swap32(v);
#endif
return v;
}
int32_t stream_tell(stream_t *stream)
{
return stream->ci->curpos;
}
uint32_t stream_read_uint32(stream_t *stream)
{
uint32_t v;
stream_read(stream, 4, &v);
#ifdef ROCKBOX_LITTLE_ENDIAN
_Swap32(v);
#endif
return v;
}
int16_t stream_read_int16(stream_t *stream)
{
int16_t v;
stream_read(stream, 2, &v);
#ifdef ROCKBOX_LITTLE_ENDIAN
_Swap16(v);
#endif
return v;
}
uint16_t stream_read_uint16(stream_t *stream)
{
uint16_t v;
stream_read(stream, 2, &v);
#ifdef ROCKBOX_LITTLE_ENDIAN
_Swap16(v);
#endif
return v;
}
int8_t stream_read_int8(stream_t *stream)
{
int8_t v;
stream_read(stream, 1, &v);
return v;
}
uint8_t stream_read_uint8(stream_t *stream)
{
uint8_t v;
stream_read(stream, 1, &v);
return v;
}
void stream_skip(stream_t *stream, size_t skip)
{
stream->ci->advance_buffer(skip);
}
int stream_eof(stream_t *stream)
{
return stream->eof;
}
void stream_create(stream_t *stream,struct codec_api* ci)
{
stream->ci=ci;
stream->eof=0;
}
/* This function was part of the original alac decoder implementation */
int get_sample_info(demux_res_t *demux_res, uint32_t samplenum,
uint32_t *sample_duration,
uint32_t *sample_byte_size)
{
unsigned int duration_index_accum = 0;
unsigned int duration_cur_index = 0;
if (samplenum >= demux_res->num_sample_byte_sizes) {
return 0;
}
if (!demux_res->num_time_to_samples) {
return 0;
}
while ((demux_res->time_to_sample[duration_cur_index].sample_count
+ duration_index_accum) <= samplenum) {
duration_index_accum +=
demux_res->time_to_sample[duration_cur_index].sample_count;
duration_cur_index++;
if (duration_cur_index >= demux_res->num_time_to_samples) {
return 0;
}
}
*sample_duration =
demux_res->time_to_sample[duration_cur_index].sample_duration;
*sample_byte_size = demux_res->sample_byte_size[samplenum];
return 1;
}
unsigned int get_sample_offset(demux_res_t *demux_res, uint32_t sample)
{
uint32_t chunk = 1;
uint32_t range_samples = 0;
uint32_t total_samples = 0;
uint32_t chunk_sample;
uint32_t prev_chunk;
uint32_t prev_chunk_samples;
uint32_t file_offset;
uint32_t i;
/* First check we have the appropriate metadata - we should always
* have it.
*/
if (sample >= demux_res->num_sample_byte_sizes ||
!demux_res->num_sample_to_chunks ||
!demux_res->num_chunk_offsets)
{
return 0;
}
/* Locate the chunk containing the sample */
prev_chunk = demux_res->sample_to_chunk[0].first_chunk;
prev_chunk_samples = demux_res->sample_to_chunk[0].num_samples;
for (i = 1; i < demux_res->num_sample_to_chunks; i++)
{
chunk = demux_res->sample_to_chunk[i].first_chunk;
range_samples = (chunk - prev_chunk) * prev_chunk_samples;
if (sample < total_samples + range_samples)
{
break;
}
total_samples += range_samples;
prev_chunk = demux_res->sample_to_chunk[i].first_chunk;
prev_chunk_samples = demux_res->sample_to_chunk[i].num_samples;
}
if (sample >= demux_res->sample_to_chunk[0].num_samples)
{
chunk = prev_chunk + (sample - total_samples) / prev_chunk_samples;
}
else
{
chunk = 1;
}
/* Get sample of the first sample in the chunk */
chunk_sample = total_samples + (chunk - prev_chunk) * prev_chunk_samples;
/* Get offset in file */
if (chunk > demux_res->num_chunk_offsets)
{
file_offset = demux_res->chunk_offset[demux_res->num_chunk_offsets - 1];
}
else
{
file_offset = demux_res->chunk_offset[chunk - 1];
}
if (chunk_sample > sample)
{
return 0;
}
for (i = chunk_sample; i < sample; i++)
{
file_offset += demux_res->sample_byte_size[i];
}
if (file_offset > demux_res->mdat_offset + demux_res->mdat_len)
{
return 0;
}
return file_offset;
}
/* Seek to the sample containing sound_sample_loc. Return 1 on success
* (and modify sound_samples_done and current_sample), 0 if failed.
*
* Seeking uses the following arrays:
*
* 1) the time_to_sample array contains the duration (in sound samples)
* of each sample of data.
*
* 2) the sample_byte_size array contains the length in bytes of each
* sample.
*
* 3) the sample_to_chunk array contains information about which chunk
* of samples each sample belongs to.
*
* 4) the chunk_offset array contains the file offset of each chunk.
*
* So find the sample number we are going to seek to (using time_to_sample)
* and then find the offset in the file (using sample_to_chunk,
* chunk_offset sample_byte_size, in that order.).
*
*/
unsigned int alac_seek(demux_res_t* demux_res, stream_t* stream,
uint32_t sound_sample_loc, uint32_t* sound_samples_done,
int* current_sample)
{
uint32_t i;
uint32_t j;
uint32_t new_sample;
uint32_t new_sound_sample;
uint32_t new_pos;
/* First check we have the appropriate metadata - we should always
* have it.
*/
if ((demux_res->num_time_to_samples==0) ||
(demux_res->num_sample_byte_sizes==0))
{
return 0;
}
/* Find the destination block from time_to_sample array */
i = 0;
new_sample = 0;
new_sound_sample = 0;
while ((i < demux_res->num_time_to_samples) &&
(new_sound_sample < sound_sample_loc))
{
j = (sound_sample_loc - new_sound_sample) /
demux_res->time_to_sample[i].sample_duration;
if (j <= demux_res->time_to_sample[i].sample_count)
{
new_sample += j;
new_sound_sample += j *
demux_res->time_to_sample[i].sample_duration;
break;
}
else
{
new_sound_sample += (demux_res->time_to_sample[i].sample_duration
* demux_res->time_to_sample[i].sample_count);
new_sample += demux_res->time_to_sample[i].sample_count;
i++;
}
}
/* We know the new block, now calculate the file position. */
new_pos = get_sample_offset(demux_res, new_sample);
/* We know the new file position, so let's try to seek to it */
if (stream->ci->seek_buffer(new_pos))
{
*sound_samples_done = new_sound_sample;
*current_sample = new_sample;
return 1;
}
return 0;
}
/* Seek to the sample containing file_loc. Return 1 on success (and modify
* sound_samples_done and current_sample), 0 if failed.
*
* Seeking uses the following arrays:
*
* 1) the chunk_offset array contains the file offset of each chunk.
*
* 2) the sample_to_chunk array contains information about which chunk
* of samples each sample belongs to.
*
* 3) the sample_byte_size array contains the length in bytes of each
* sample.
*
* 4) the time_to_sample array contains the duration (in sound samples)
* of each sample of data.
*
* Locate the chunk containing location (using chunk_offset), find the
* sample of that chunk (using sample_to_chunk) and finally the location
* of that sample (using sample_byte_size). Then use time_to_sample to
* calculate the sound_samples_done value.
*/
unsigned int alac_seek_raw(demux_res_t* demux_res, stream_t* stream,
uint32_t file_loc, uint32_t* sound_samples_done,
int* current_sample)
{
uint32_t chunk_sample = 0;
uint32_t total_samples = 0;
uint32_t new_sound_sample = 0;
uint32_t new_pos;
uint32_t chunk;
uint32_t i;
if (!demux_res->num_chunk_offsets ||
!demux_res->num_sample_to_chunks)
{
return 0;
}
/* Locate the chunk containing file_loc. */
for (i = 0; i < demux_res->num_chunk_offsets &&
file_loc < demux_res->chunk_offset[i]; i++)
{
}
chunk = i + 1;
new_pos = demux_res->chunk_offset[chunk - 1];
/* Get the first sample of the chunk. */
for (i = 1; i < demux_res->num_sample_to_chunks &&
chunk < demux_res->sample_to_chunk[i - 1].first_chunk; i++)
{
chunk_sample += demux_res->sample_to_chunk[i - 1].num_samples *
(demux_res->sample_to_chunk[i].first_chunk -
demux_res->sample_to_chunk[i - 1].first_chunk);
}
chunk_sample += (chunk - demux_res->sample_to_chunk[i - 1].first_chunk) *
demux_res->sample_to_chunk[i - 1].num_samples;
/* Get the position within the chunk. */
for (; chunk_sample < demux_res->num_sample_byte_sizes; chunk_sample++)
{
if (file_loc < new_pos + demux_res->sample_byte_size[chunk_sample])
{
break;
}
new_pos += demux_res->sample_byte_size[chunk_sample];
}
/* Get sound sample offset. */
for (i = 0; i < demux_res->num_time_to_samples; i++)
{
if (chunk_sample <
total_samples + demux_res->time_to_sample[i].sample_count)
{
break;
}
total_samples += demux_res->time_to_sample[i].sample_count;
new_sound_sample += demux_res->time_to_sample[i].sample_count
* demux_res->time_to_sample[i].sample_duration;
}
new_sound_sample += (chunk_sample - total_samples)
* demux_res->time_to_sample[i].sample_duration;
/* Go to the new file position. */
if (stream->ci->seek_buffer(new_pos))
{
*sound_samples_done = new_sound_sample;
*current_sample = chunk_sample;
return 1;
}
return 0;
}