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Major rework of the m4a parser for aac/alac playback, seek and resume support. As a result the memory consumption was drastically reduced. This allows to play several files with long duration -- especially on low memory targets. The change builds a lookup table from m4a's sample_to_chunk[] and chunk_offset[] and completely removes the allocation of the large tables chunk_offset[] and sample_byte_size[]. To be able to remove reading and allocating sample_byte_offset[] the aac and alac decoder now buffer a fixed amount of bytes for each frame. The generated lookup table is used for seek/resume and skipping bytes in empty chunks (aac decoder only). The precision for seek/resume is somewhat lower but still equals 0.5 sec for the worst case. Fixes FS#8923.

Browse source 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29745 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Andree Buschmann 2011-04-18 19:12:51 +00:00
parent b6a6809879
commit 68199cc195
5 changed files with 137 additions and 245 deletions
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15
apps/codecs/aac.c
View file

@ -27,6 +27,11 @@
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)
/* 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
@ -159,7 +164,7 @@ next_track:
/* 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 (alac_seek_raw(&demux_res, &input_stream, file_offset,
if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
&sound_samples_done, (int*) &i)) {
sound_samples_done *= sbr_fac;
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
@ -188,9 +193,9 @@ next_track:
if (ci->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
* alac_seek and the resulting sound_samples_done must be expanded
* m4a_seek and the resulting sound_samples_done must be expanded
* by a factor 2. This is done via using sbr_fac. */
if (alac_seek(&demux_res, &input_stream,
if (m4a_seek(&demux_res, &input_stream,
((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
&sound_samples_done, (int*) &i)) {
sound_samples_done *= sbr_fac;
@ -211,7 +216,7 @@ next_track:
* 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);
file_offset = m4a_check_sample_offset(&demux_res, i);
if (file_offset > ci->curpos)
{
@ -225,7 +230,7 @@ next_track:
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, demux_res.sample_byte_size[i]);
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);

14
apps/codecs/alac.c
View file

@ -25,6 +25,10 @@
CODEC_HEADER
/* The maximum buffer size handled. This amount of bytes is buffered for each
* frame. */
#define ALAC_BYTE_BUFFER_SIZE 32768
static int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE] IBSS_ATTR;
/* this is the codec entry point */
@ -83,7 +87,7 @@ enum codec_status codec_main(void)
/* Set i for first frame, seek to desired sample position for resuming. */
i=0;
if (samplesdone > 0) {
if (alac_seek(&demux_res, &input_stream, samplesdone,
if (m4a_seek(&demux_res, &input_stream, samplesdone,
&samplesdone, (int*) &i)) {
elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsedtime);
@ -101,7 +105,7 @@ enum codec_status codec_main(void)
/* Deal with any pending seek requests */
if (ci->seek_time) {
if (alac_seek(&demux_res, &input_stream,
if (m4a_seek(&demux_res, &input_stream,
((ci->seek_time-1)/10) * (ci->id3->frequency/100),
&samplesdone, (int *)&i)) {
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
@ -111,11 +115,7 @@ enum codec_status codec_main(void)
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, demux_res.sample_byte_size[i]);
if (n!=demux_res.sample_byte_size[i]) {
retval = CODEC_ERROR;
goto done;
}
buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
/* Decode one block - returned samples will be host-endian */
ci->yield();

92
apps/codecs/libm4a/demux.c
View file

@ -354,8 +354,6 @@ static bool read_chunk_stts(qtmovie_t *qtmovie, size_t chunk_len)
static bool read_chunk_stsz(qtmovie_t *qtmovie, size_t chunk_len)
{
unsigned int i;
uint32_t numentries;
size_t size_remaining = chunk_len - 8;
/* version */
@ -377,35 +375,11 @@ static bool read_chunk_stsz(qtmovie_t *qtmovie, size_t chunk_len)
}
size_remaining -= 4;
numentries = stream_read_uint32(qtmovie->stream);
qtmovie->res->num_sample_byte_sizes = stream_read_uint32(qtmovie->stream);
size_remaining -= 4;
qtmovie->res->num_sample_byte_sizes = numentries;
qtmovie->res->sample_byte_size = malloc(numentries * sizeof(*qtmovie->res->sample_byte_size));
if (!qtmovie->res->sample_byte_size)
{
DEBUGF("stsz too large\n");
return false;
}
for (i = 0; i < numentries; i++)
{
uint32_t v = stream_read_uint32(qtmovie->stream);
if (v > 0x0000ffff)
{
DEBUGF("stsz[%d] > 65 kB (%ld)\n", i, (long)v);
return false;
}
qtmovie->res->sample_byte_size[i] = v;
size_remaining -= 4;
}
if (size_remaining)
{
DEBUGF("ehm, size remianing?\n");
stream_skip(qtmovie->stream, size_remaining);
}
@ -426,8 +400,7 @@ static bool read_chunk_stsc(qtmovie_t *qtmovie, size_t chunk_len)
size_remaining -= 4;
qtmovie->res->num_sample_to_chunks = numentries;
qtmovie->res->sample_to_chunk = malloc(numentries *
sizeof(*qtmovie->res->sample_to_chunk));
qtmovie->res->sample_to_chunk = malloc(numentries * sizeof(sample_to_chunk_t));
if (!qtmovie->res->sample_to_chunk)
{
@ -456,8 +429,14 @@ static bool read_chunk_stsc(qtmovie_t *qtmovie, size_t chunk_len)
static bool read_chunk_stco(qtmovie_t *qtmovie, size_t chunk_len)
{
unsigned int i;
uint32_t i, k;
uint32_t numentries;
int32_t idx = 0;
int32_t frame;
int32_t offset;
int32_t old_first;
int32_t new_first;
int32_t old_frame;
size_t size_remaining = chunk_len - 8;
/* version + flags */
@ -466,20 +445,57 @@ static bool read_chunk_stco(qtmovie_t *qtmovie, size_t chunk_len)
numentries = stream_read_uint32(qtmovie->stream);
size_remaining -= 4;
qtmovie->res->num_lookup_table = numentries;
qtmovie->res->lookup_table = malloc(numentries * sizeof(*qtmovie->res->lookup_table));
qtmovie->res->num_chunk_offsets = numentries;
qtmovie->res->chunk_offset = malloc(numentries *
sizeof(*qtmovie->res->chunk_offset));
if (!qtmovie->res->chunk_offset)
if (!qtmovie->res->lookup_table)
{
DEBUGF("stco too large\n");
DEBUGF("stco too large to allocate lookup_table[]\n");
return false;
}
for (i = 0; i < numentries; i++)
/* read first offset */
offset = stream_read_uint32(qtmovie->stream);
size_remaining -= 4;
/* Build up lookup table. The lookup table contains the sample index and
* byte position in the file for each chunk. This table is used to seek
* and resume (see m4a_seek() and m4a_seek_raw() in libm4a/m4a.c) and
* to skip empty chunks (see m4a_check_sample_offset() in codecs/aac.c and
* libm4a/m4a.c).
* The seek/resume precision is lower than using sample_byte_size[] and
* depends on numentries. Typically the resolution is ~1/10 of all frames
* which equals about 1/4-1/2 seconds. The loss of seek precision is
* accepted to be able to avoid allocation of the large sample_byte_size[]
* table. This reduces the memory consumption by a factor of 2 or even
* more. */
i = 1;
frame = 0;
old_frame = qtmovie->res->sample_to_chunk[0].num_samples;
old_first = qtmovie->res->sample_to_chunk[0].first_chunk;
for (k = 1; k < numentries; ++k)
{
qtmovie->res->chunk_offset[i] = stream_read_uint32(qtmovie->stream);
for (; i < qtmovie->res->num_sample_to_chunks; ++i)
{
old_frame = qtmovie->res->sample_to_chunk[i-1].num_samples;
old_first = qtmovie->res->sample_to_chunk[i-1].first_chunk;
new_first = qtmovie->res->sample_to_chunk[i ].first_chunk;
if (qtmovie->res->sample_to_chunk[i].first_chunk > k)
break;
frame += (new_first - old_first) * old_frame;
}
frame += (k - old_first) * old_frame;
qtmovie->res->lookup_table[idx].sample = frame;
qtmovie->res->lookup_table[idx].offset = offset;
idx++;
frame -= (k - old_first) * old_frame;
offset = stream_read_uint32(qtmovie->stream);
size_remaining -= 4;
}

245
apps/codecs/libm4a/m4a.c
View file

@ -7,7 +7,7 @@
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
* Copyright (C) 2005 Dave Chapman, 2011 Andree Buschmann
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -122,148 +122,46 @@ void stream_create(stream_t *stream,struct codec_api* 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)
/* Check if there is a dedicated byte position contained for the given frame.
* Return this byte position in case of success or return -1. This allows to
* skip empty samples. */
int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame)
{
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)
uint32_t i = 0;
for (i=0; i<demux_res->num_lookup_table; ++i)
{
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)
{
if (demux_res->lookup_table[i].sample > frame ||
demux_res->lookup_table[i].offset == 0)
return -1;
if (demux_res->lookup_table[i].sample == frame)
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 (prev_chunk_samples > 0 &&
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;
return demux_res->lookup_table[i].offset;
}
/* Seek to the sample containing sound_sample_loc. Return 1 on success
* (and modify sound_samples_done and current_sample), 0 if failed.
/* Find the exact or preceding frame in lookup_table[]. Return both frame
* and byte position of this match. */
static void gather_offset(demux_res_t *demux_res, uint32_t *frame, uint32_t *offset)
{
uint32_t i = 0;
for (i=0; i<demux_res->num_lookup_table; ++i)
{
if (demux_res->lookup_table[i].offset == 0)
break;
if (demux_res->lookup_table[i].sample > *frame)
break;
}
i = (i>0) ? i-1 : 0; /* We want the last chunk _before_ *frame. */
*frame = demux_res->lookup_table[i].sample;
*offset = demux_res->lookup_table[i].offset;
}
/* Seek to desired sound sample location. 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,
* Find the sample (=frame) that contains the given sound sample, find a best
* fit for this sample in the lookup_table[], seek to the byte position. */
unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
uint32_t sound_sample_loc, uint32_t* sound_samples_done,
int* current_sample)
{
@ -300,8 +198,8 @@ unsigned int alac_seek(demux_res_t* demux_res, stream_t* stream,
++i;
}
/* We know the new block, now calculate the file position. */
new_pos = get_sample_offset(demux_res, new_sample);
/* We know the new sample (=frame), now calculate the file position. */
gather_offset(demux_res, &new_sample, &new_pos);
/* We know the new file position, so let's try to seek to it */
if (stream->ci->seek_buffer(new_pos))
@ -319,71 +217,38 @@ unsigned int alac_seek(demux_res_t* demux_res, stream_t* stream,
*
* Seeking uses the following arrays:
*
* 1) the chunk_offset array contains the file offset of each chunk.
* 1) the lookup_table array contains the file offset for the first sample
* 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)
* 2) 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
* Locate the chunk containing location (using lookup_table), find the first
* sample of that chunk (using lookup_table). 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,
unsigned int m4a_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; /* Holds the chunk/frame index. */
uint32_t total_samples = 0; /* Sums up total amount of chunks/frames. */
uint32_t new_sound_sample = 0; /* Sums up total amount of samples. */
uint32_t i;
uint32_t chunk_sample = 0;
uint32_t total_samples = 0;
uint32_t new_sound_sample = 0;
uint32_t tmp_dur;
uint32_t tmp_cnt;
uint32_t new_pos;
uint32_t chunk;
uint32_t i, tmp_dur, tmp_cnt;
/* There is no metadata available to perform raw seek. */
if (!demux_res->num_chunk_offsets || !demux_res->num_sample_to_chunks)
/* We know the desired byte offset, search for the chunk right before.
* Return the associated sample to this chunk as chunk_sample. */
for (i=0; i < demux_res->num_lookup_table; ++i)
{
return 0;
}
/* Locate the chunk containing file_loc. */
chunk = 0;
while (chunk < demux_res->num_chunk_offsets)
{
if (file_loc < demux_res->chunk_offset[chunk])
{
if (demux_res->lookup_table[i].offset > file_loc)
break;
}
++chunk;
}
new_pos = demux_res->chunk_offset[chunk > 0 ? chunk - 1 : 0];
/* Get the first sample of the chunk. */
i = 1;
sample_to_chunk_t *tab1 = demux_res->sample_to_chunk;
while (i < demux_res->num_sample_to_chunks)
{
if (chunk <= tab1[i].first_chunk)
{
break;
}
chunk_sample += tab1[i-1].num_samples * (tab1[i].first_chunk - tab1[i-1].first_chunk);
++i;
}
chunk_sample += (chunk - tab1[i-1].first_chunk) * tab1[i-1].num_samples;
/* Get the position within the chunk. */
while (chunk_sample < demux_res->num_sample_byte_sizes &&
file_loc >= new_pos + demux_res->sample_byte_size[chunk_sample])
{
new_pos += demux_res->sample_byte_size[chunk_sample++];
}
i = (i>0) ? i-1 : 0; /* We want the last chunk _before_ file_loc. */
chunk_sample = demux_res->lookup_table[i].sample;
new_pos = demux_res->lookup_table[i].offset;
/* Get sound sample offset. */
i = 0;

16
apps/codecs/libm4a/m4a.h
View file

@ -57,6 +57,12 @@ typedef struct
uint32_t sample_duration;
} time_to_sample_t;
typedef struct
{
uint32_t sample;
uint32_t offset;
} sample_offset_t;
typedef struct
{
uint16_t num_channels;
@ -68,13 +74,12 @@ typedef struct
sample_to_chunk_t *sample_to_chunk;
uint32_t num_sample_to_chunks;
uint32_t *chunk_offset;
uint32_t num_chunk_offsets;
sample_offset_t *lookup_table;
uint32_t num_lookup_table;
time_to_sample_t *time_to_sample;
uint32_t num_time_to_samples;
uint16_t *sample_byte_size;
uint32_t num_sample_byte_sizes;
uint32_t codecdata_len;
@ -127,10 +132,11 @@ void stream_create(stream_t *stream,struct codec_api* ci);
int get_sample_info(demux_res_t *demux_res, uint32_t sample,
uint32_t *sample_duration, uint32_t *sample_byte_size);
unsigned int get_sample_offset(demux_res_t *demux_res, uint32_t sample);
unsigned int alac_seek (demux_res_t* demux_res, stream_t* stream,
unsigned int m4a_seek (demux_res_t* demux_res, stream_t* stream,
uint32_t sound_sample_loc, uint32_t* sound_samples_done,
int* current_sample);
unsigned int alac_seek_raw (demux_res_t* demux_res, stream_t* stream,
unsigned int m4a_seek_raw (demux_res_t* demux_res, stream_t* stream,
uint32_t file_loc, uint32_t* sound_samples_done, int* current_sample);
int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame);
#endif /* STREAM_H */