/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* 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 program 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 General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../decoder_api.h"
#include "decoder_buffer.h"
#include "config.h"
#define AAC_MAX_CHANNELS 6
#include <assert.h>
#include <unistd.h>
#include <faad.h>
#include <glib.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "faad"
static const unsigned adts_sample_rates[] =
{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
/**
* Check whether the buffer head is an AAC frame, and return the frame
* length. Returns 0 if it is not a frame.
*/
static size_t
adts_check_frame(const unsigned char *data)
{
/* check syncword */
if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0)))
return 0;
return (((unsigned int)data[3] & 0x3) << 11) |
(((unsigned int)data[4]) << 3) |
(data[5] >> 5);
}
/**
* Find the next AAC frame in the buffer. Returns 0 if no frame is
* found or if not enough data is available.
*/
static size_t
adts_find_frame(struct decoder_buffer *buffer)
{
const unsigned char *data, *p;
size_t length, frame_length;
bool ret;
while (true) {
data = decoder_buffer_read(buffer, &length);
if (data == NULL || length < 8) {
/* not enough data yet */
ret = decoder_buffer_fill(buffer);
if (!ret)
/* failed */
return 0;
continue;
}
/* find the 0xff marker */
p = memchr(data, 0xff, length);
if (p == NULL) {
/* no marker - discard the buffer */
decoder_buffer_consume(buffer, length);
continue;
}
if (p > data) {
/* discard data before 0xff */
decoder_buffer_consume(buffer, p - data);
continue;
}
/* is it a frame? */
frame_length = adts_check_frame(data);
if (frame_length == 0) {
/* it's just some random 0xff byte; discard it
and continue searching */
decoder_buffer_consume(buffer, 1);
continue;
}
if (length < frame_length) {
/* available buffer size is smaller than the
frame will be - attempt to read more
data */
ret = decoder_buffer_fill(buffer);
if (!ret) {
/* not enough data; discard this frame
to prevent a possible buffer
overflow */
data = decoder_buffer_read(buffer, &length);
if (data != NULL)
decoder_buffer_consume(buffer, length);
}
continue;
}
/* found a full frame! */
return frame_length;
}
}
static float
adts_song_duration(struct decoder_buffer *buffer)
{
unsigned int frames, frame_length;
unsigned sample_rate = 0;
float frames_per_second;
/* Read all frames to ensure correct time and bitrate */
for (frames = 0;; frames++) {
frame_length = adts_find_frame(buffer);
if (frame_length == 0)
break;
if (frames == 0) {
const unsigned char *data;
size_t buffer_length;
data = decoder_buffer_read(buffer, &buffer_length);
assert(data != NULL);
assert(frame_length <= buffer_length);
sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
}
decoder_buffer_consume(buffer, frame_length);
}
frames_per_second = (float)sample_rate / 1024.0;
if (frames_per_second <= 0)
return -1;
return (float)frames / frames_per_second;
}
static float
faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
{
size_t fileread;
size_t tagsize;
const unsigned char *data;
size_t length;
fileread = is->size >= 0 ? is->size : 0;
decoder_buffer_fill(buffer);
data = decoder_buffer_read(buffer, &length);
if (data == NULL)
return -1;
tagsize = 0;
if (length >= 10 && !memcmp(data, "ID3", 3)) {
/* skip the ID3 tag */
tagsize = (data[6] << 21) | (data[7] << 14) |
(data[8] << 7) | (data[9] << 0);
tagsize += 10;
decoder_buffer_consume(buffer, tagsize);
decoder_buffer_fill(buffer);
data = decoder_buffer_read(buffer, &length);
if (data == NULL)
return -1;
}
if (is->seekable && length >= 2 &&
data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) {
/* obtain the duration from the ADTS header */
float song_length = adts_song_duration(buffer);
input_stream_seek(is, tagsize, SEEK_SET);
data = decoder_buffer_read(buffer, &length);
if (data != NULL)
decoder_buffer_consume(buffer, length);
decoder_buffer_fill(buffer);
return song_length;
} else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) {
/* obtain the duration from the ADIF header */
unsigned bit_rate;
size_t skip_size = (data[4] & 0x80) ? 9 : 0;
if (8 + skip_size > length)
/* not enough data yet; skip parsing this
header */
return -1;
bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
(data[5 + skip_size] << 11) |
(data[6 + skip_size] << 3) |
(data[7 + skip_size] & 0xE0);
if (fileread != 0 && bit_rate != 0)
return fileread * 8.0 / bit_rate;
else
return fileread;
} else
return -1;
}
/**
* Wrapper for faacDecInit() which works around some API
* inconsistencies in libfaad.
*/
static bool
faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
struct audio_format *audio_format)
{
union {
/* deconst hack for libfaad */
const void *in;
void *out;
} u;
size_t length;
int32_t nbytes;
uint32_t sample_rate;
uint8_t channels;
#ifdef HAVE_FAAD_LONG
/* neaacdec.h declares all arguments as "unsigned long", but
internally expects uint32_t pointers. To avoid gcc
warnings, use this workaround. */
unsigned long *sample_rate_r = (unsigned long *)(void *)&sample_rate;
#else
uint32_t *sample_rate_r = &sample_rate;
#endif
u.in = decoder_buffer_read(buffer, &length);
if (u.in == NULL)
return false;
nbytes = faacDecInit(decoder, u.out,
#ifdef HAVE_FAAD_BUFLEN_FUNCS
length,
#endif
sample_rate_r, &channels);
if (nbytes < 0)
return false;
decoder_buffer_consume(buffer, nbytes);
*audio_format = (struct audio_format){
.bits = 16,
.channels = channels,
.sample_rate = sample_rate,
};
return true;
}
/**
* Wrapper for faacDecDecode() which works around some API
* inconsistencies in libfaad.
*/
static const void *
faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer,
NeAACDecFrameInfo *frame_info)
{
union {
/* deconst hack for libfaad */
const void *in;
void *out;
} u;
size_t length;
void *result;
u.in = decoder_buffer_read(buffer, &length);
if (u.in == NULL)
return false;
result = faacDecDecode(decoder, frame_info,
u.out
#ifdef HAVE_FAAD_BUFLEN_FUNCS
, length
#endif
);
return result;
}
/**
* Get a song file's total playing time in seconds, as a float.
* Returns 0 if the duration is unknown, and a negative value if the
* file is invalid.
*/
static float
faad_get_file_time_float(const char *file)
{
struct decoder_buffer *buffer;
float length;
faacDecHandle decoder;
faacDecConfigurationPtr config;
struct input_stream is;
if (!input_stream_open(&is, file))
return -1;
buffer = decoder_buffer_new(NULL, &is,
FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
length = faad_song_duration(buffer, &is);
if (length < 0) {
bool ret;
struct audio_format audio_format;
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
faacDecSetConfiguration(decoder, config);
decoder_buffer_fill(buffer);
ret = faad_decoder_init(decoder, buffer, &audio_format);
if (ret && audio_format_valid(&audio_format))
length = 0;
faacDecClose(decoder);
}
decoder_buffer_free(buffer);
input_stream_close(&is);
return length;
}
/**
* Get a song file's total playing time in seconds, as an int.
* Returns 0 if the duration is unknown, and a negative value if the
* file is invalid.
*/
static int
faad_get_file_time(const char *file)
{
int file_time = -1;
float length;
if ((length = faad_get_file_time_float(file)) >= 0)
file_time = length + 0.5;
return file_time;
}
static void
faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
{
float file_time;
float total_time = 0;
faacDecHandle decoder;
struct audio_format audio_format;
faacDecFrameInfo frame_info;
faacDecConfigurationPtr config;
unsigned long sample_count;
bool ret;
const void *decoded;
size_t decoded_length;
uint16_t bit_rate = 0;
struct decoder_buffer *buffer;
enum decoder_command cmd;
buffer = decoder_buffer_new(mpd_decoder, is,
FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
total_time = faad_song_duration(buffer, is);
/* create the libfaad decoder */
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
config->dontUpSampleImplicitSBR = 0;
#endif
faacDecSetConfiguration(decoder, config);
while (!decoder_buffer_is_full(buffer) &&
!input_stream_eof(is) &&
decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
adts_find_frame(buffer);
decoder_buffer_fill(buffer);
}
/* initialize it */
ret = faad_decoder_init(decoder, buffer, &audio_format);
if (!ret) {
g_warning("Error not a AAC stream.\n");
faacDecClose(decoder);
return;
}
if (!audio_format_valid(&audio_format)) {
g_warning("invalid audio format\n");
faacDecClose(decoder);
return;
}
/* initialize the MPD core */
decoder_initialized(mpd_decoder, &audio_format, false, total_time);
/* the decoder loop */
file_time = 0.0;
do {
size_t frame_size;
/* find the next frame */
frame_size = adts_find_frame(buffer);
if (frame_size == 0)
/* end of file */
break;
/* decode it */
decoded = faad_decoder_decode(decoder, buffer, &frame_info);
if (frame_info.error > 0) {
g_warning("error decoding AAC stream: %s\n",
faacDecGetErrorMessage(frame_info.error));
break;
}
if (frame_info.channels != audio_format.channels) {
g_warning("channel count changed from %u to %u",
audio_format.channels, frame_info.channels);
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
if (frame_info.samplerate != audio_format.sample_rate) {
g_warning("sample rate changed from %u to %lu",
audio_format.sample_rate,
(unsigned long)frame_info.samplerate);
break;
}
#endif
decoder_buffer_consume(buffer, frame_info.bytesconsumed);
/* update bit rate and position */
sample_count = (unsigned long)frame_info.samples;
if (sample_count > 0) {
bit_rate = frame_info.bytesconsumed * 8.0 *
frame_info.channels * audio_format.sample_rate /
frame_info.samples / 1000 + 0.5;
file_time +=
(float)(frame_info.samples) / frame_info.channels /
audio_format.sample_rate;
}
/* send PCM samples to MPD */
decoded_length = sample_count * 2;
cmd = decoder_data(mpd_decoder, is, decoded,
decoded_length, file_time,
bit_rate, NULL);
} while (cmd != DECODE_COMMAND_STOP);
/* cleanup */
faacDecClose(decoder);
}
static struct tag *
faad_tag_dup(const char *file)
{
int file_time = faad_get_file_time(file);
struct tag *tag;
if (file_time < 0) {
g_debug("Failed to get total song time from: %s", file);
return NULL;
}
tag = tag_new();
tag->time = file_time;
return tag;
}
static const char *const faad_suffixes[] = { "aac", NULL };
static const char *const faad_mime_types[] = {
"audio/aac", "audio/aacp", NULL
};
const struct decoder_plugin faad_decoder_plugin = {
.name = "faad",
.stream_decode = faad_stream_decode,
.tag_dup = faad_tag_dup,
.suffixes = faad_suffixes,
.mime_types = faad_mime_types,
};