mpd/src/decoder/FaadDecoderPlugin.cxx
2013-04-17 22:45:10 +02:00

503 lines
12 KiB
C++

/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "FaadDecoderPlugin.hxx"
#include "decoder_api.h"
#include "DecoderBuffer.hxx"
#include "audio_check.h"
#include "tag_handler.h"
#include <neaacdec.h>
#include <glib.h>
#include <assert.h>
#include <unistd.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "faad"
#define AAC_MAX_CHANNELS 6
static const unsigned adts_sample_rates[] =
{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
/**
* The GLib quark used for errors reported by this plugin.
*/
static inline GQuark
faad_decoder_quark(void)
{
return g_quark_from_static_string("faad");
}
/**
* 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(DecoderBuffer *buffer)
{
size_t length, frame_length;
bool ret;
while (true) {
const uint8_t *data = (const uint8_t *)
decoder_buffer_read(buffer, &length);
if (data == nullptr || length < 8) {
/* not enough data yet */
ret = decoder_buffer_fill(buffer);
if (!ret)
/* failed */
return 0;
continue;
}
/* find the 0xff marker */
const uint8_t *p = (const uint8_t *)memchr(data, 0xff, length);
if (p == nullptr) {
/* 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 = (const uint8_t *)
decoder_buffer_read(buffer, &length);
if (data != nullptr)
decoder_buffer_consume(buffer, length);
}
continue;
}
/* found a full frame! */
return frame_length;
}
}
static float
adts_song_duration(DecoderBuffer *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) {
size_t buffer_length;
const uint8_t *data = (const uint8_t *)
decoder_buffer_read(buffer, &buffer_length);
assert(data != nullptr);
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(DecoderBuffer *buffer, struct input_stream *is)
{
size_t fileread;
size_t tagsize;
size_t length;
bool success;
const goffset size = input_stream_get_size(is);
fileread = size >= 0 ? size : 0;
decoder_buffer_fill(buffer);
const uint8_t *data = (const uint8_t *)
decoder_buffer_read(buffer, &length);
if (data == nullptr)
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;
success = decoder_buffer_skip(buffer, tagsize) &&
decoder_buffer_fill(buffer);
if (!success)
return -1;
data = (const uint8_t *)decoder_buffer_read(buffer, &length);
if (data == nullptr)
return -1;
}
if (input_stream_is_seekable(is) && 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_lock_seek(is, tagsize, SEEK_SET, nullptr);
data = (const uint8_t *)decoder_buffer_read(buffer, &length);
if (data != nullptr)
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 NeAACDecInit() which works around some API
* inconsistencies in libfaad.
*/
static bool
faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
struct audio_format *audio_format, GError **error_r)
{
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_p = (unsigned long *)(void *)&sample_rate;
#else
uint32_t *sample_rate_p = &sample_rate;
#endif
size_t length;
const unsigned char *data = (const unsigned char *)
decoder_buffer_read(buffer, &length);
if (data == nullptr) {
g_set_error(error_r, faad_decoder_quark(), 0,
"Empty file");
return false;
}
nbytes = NeAACDecInit(decoder,
/* deconst hack, libfaad requires this */
const_cast<unsigned char *>(data),
length,
sample_rate_p, &channels);
if (nbytes < 0) {
g_set_error(error_r, faad_decoder_quark(), 0,
"Not an AAC stream");
return false;
}
decoder_buffer_consume(buffer, nbytes);
return audio_format_init_checked(audio_format, sample_rate,
SAMPLE_FORMAT_S16, channels, error_r);
}
/**
* Wrapper for NeAACDecDecode() which works around some API
* inconsistencies in libfaad.
*/
static const void *
faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer,
NeAACDecFrameInfo *frame_info)
{
size_t length;
const unsigned char *data = (const unsigned char *)
decoder_buffer_read(buffer, &length);
if (data == nullptr)
return nullptr;
return NeAACDecDecode(decoder, frame_info,
/* deconst hack, libfaad requires this */
const_cast<unsigned char *>(data),
length);
}
/**
* 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(struct input_stream *is)
{
DecoderBuffer *buffer;
float length;
buffer = decoder_buffer_new(nullptr, is,
FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
length = faad_song_duration(buffer, is);
if (length < 0) {
bool ret;
struct audio_format audio_format;
NeAACDecHandle decoder = NeAACDecOpen();
NeAACDecConfigurationPtr config =
NeAACDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
NeAACDecSetConfiguration(decoder, config);
decoder_buffer_fill(buffer);
ret = faad_decoder_init(decoder, buffer, &audio_format, nullptr);
if (ret)
length = 0;
NeAACDecClose(decoder);
}
decoder_buffer_free(buffer);
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(struct input_stream *is)
{
int file_time = -1;
float length;
if ((length = faad_get_file_time_float(is)) >= 0)
file_time = length + 0.5;
return file_time;
}
static void
faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
{
GError *error = nullptr;
float total_time = 0;
struct audio_format audio_format;
bool ret;
uint16_t bit_rate = 0;
DecoderBuffer *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 */
NeAACDecHandle decoder = NeAACDecOpen();
NeAACDecConfigurationPtr config =
NeAACDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 1;
config->dontUpSampleImplicitSBR = 0;
NeAACDecSetConfiguration(decoder, config);
while (!decoder_buffer_is_full(buffer) &&
!input_stream_lock_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, &error);
if (!ret) {
g_warning("%s", error->message);
g_error_free(error);
NeAACDecClose(decoder);
return;
}
/* initialize the MPD core */
decoder_initialized(mpd_decoder, &audio_format, false, total_time);
/* the decoder loop */
do {
size_t frame_size;
const void *decoded;
NeAACDecFrameInfo frame_info;
/* 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",
NeAACDecGetErrorMessage(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;
}
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;
}
decoder_buffer_consume(buffer, frame_info.bytesconsumed);
/* update bit rate and position */
if (frame_info.samples > 0) {
bit_rate = frame_info.bytesconsumed * 8.0 *
frame_info.channels * audio_format.sample_rate /
frame_info.samples / 1000 + 0.5;
}
/* send PCM samples to MPD */
cmd = decoder_data(mpd_decoder, is, decoded,
(size_t)frame_info.samples * 2,
bit_rate);
} while (cmd != DECODE_COMMAND_STOP);
/* cleanup */
NeAACDecClose(decoder);
}
static bool
faad_scan_stream(struct input_stream *is,
const struct tag_handler *handler, void *handler_ctx)
{
int file_time = faad_get_file_time(is);
if (file_time < 0)
return false;
tag_handler_invoke_duration(handler, handler_ctx, file_time);
return true;
}
static const char *const faad_suffixes[] = { "aac", nullptr };
static const char *const faad_mime_types[] = {
"audio/aac", "audio/aacp", nullptr
};
const struct decoder_plugin faad_decoder_plugin = {
"faad",
nullptr,
nullptr,
faad_stream_decode,
nullptr,
nullptr,
faad_scan_stream,
nullptr,
faad_suffixes,
faad_mime_types,
};