mpd/src/decoder/plugins/FaadDecoderPlugin.cxx

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/*
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* Copyright 2003-2016 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"
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#include "FaadDecoderPlugin.hxx"
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#include "../DecoderAPI.hxx"
#include "../DecoderBuffer.hxx"
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#include "input/InputStream.hxx"
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#include "CheckAudioFormat.hxx"
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#include "tag/TagHandler.hxx"
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#include "util/ScopeExit.hxx"
#include "util/ConstBuffer.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "Log.hxx"
#include <neaacdec.h>
#include <stdexcept>
#include <assert.h>
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#include <string.h>
static const unsigned adts_sample_rates[] =
{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
static constexpr Domain faad_decoder_domain("faad_decoder");
/**
* 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)
{
while (true) {
auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(8));
if (data.IsNull())
/* failed */
return 0;
/* find the 0xff marker */
const uint8_t *p = (const uint8_t *)
memchr(data.data, 0xff, data.size);
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if (p == nullptr) {
/* no marker - discard the buffer */
buffer.Clear();
continue;
}
if (p > data.data) {
/* discard data before 0xff */
buffer.Consume(p - data.data);
continue;
}
/* is it a frame? */
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const size_t frame_length = adts_check_frame(data.data);
if (frame_length == 0) {
/* it's just some random 0xff byte; discard it
and continue searching */
buffer.Consume(1);
continue;
}
if (buffer.Need(frame_length).IsNull()) {
/* not enough data; discard this frame to
prevent a possible buffer overflow */
buffer.Clear();
continue;
}
/* found a full frame! */
return frame_length;
}
}
static SignedSongTime
adts_song_duration(DecoderBuffer &buffer)
{
const InputStream &is = buffer.GetStream();
const bool estimate = !is.CheapSeeking();
if (estimate && !is.KnownSize())
return SignedSongTime::Negative();
unsigned sample_rate = 0;
/* Read all frames to ensure correct time and bitrate */
unsigned frames = 0;
for (;; frames++) {
const unsigned frame_length = adts_find_frame(buffer);
if (frame_length == 0)
break;
if (frames == 0) {
auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
assert(!data.IsEmpty());
assert(frame_length <= data.size);
sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
if (sample_rate == 0)
break;
}
buffer.Consume(frame_length);
if (estimate && frames == 128) {
/* if this is a remote file, don't slurp the
whole file just for checking the song
duration; instead, stop after some time and
extrapolate the song duration from what we
have until now */
const auto offset = is.GetOffset()
- buffer.GetAvailable();
if (offset <= 0)
return SignedSongTime::Negative();
const auto file_size = is.GetSize();
frames = (frames * file_size) / offset;
break;
}
}
if (sample_rate == 0)
return SignedSongTime::Negative();
return SignedSongTime::FromScale<uint64_t>(frames * uint64_t(1024),
sample_rate);
}
static SignedSongTime
faad_song_duration(DecoderBuffer &buffer, InputStream &is)
{
auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(5));
if (data.IsNull())
return SignedSongTime::Negative();
size_t tagsize = 0;
if (data.size >= 10 && !memcmp(data.data, "ID3", 3)) {
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/* skip the ID3 tag */
tagsize = (data.data[6] << 21) | (data.data[7] << 14) |
(data.data[8] << 7) | (data.data[9] << 0);
tagsize += 10;
if (!buffer.Skip(tagsize))
return SignedSongTime::Negative();
data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(5));
if (data.IsNull())
return SignedSongTime::Negative();
}
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if (data.size >= 8 && adts_check_frame(data.data) > 0) {
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/* obtain the duration from the ADTS header */
if (!is.IsSeekable())
return SignedSongTime::Negative();
auto song_length = adts_song_duration(buffer);
try {
is.LockSeek(tagsize);
} catch (const std::runtime_error &) {
}
buffer.Clear();
return song_length;
} else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
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/* obtain the duration from the ADIF header */
if (!is.KnownSize())
return SignedSongTime::Negative();
size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
if (8 + skip_size > data.size)
/* not enough data yet; skip parsing this
header */
return SignedSongTime::Negative();
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unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
(data.data[5 + skip_size] << 11) |
(data.data[6 + skip_size] << 3) |
(data.data[7 + skip_size] & 0xE0);
const auto size = is.GetSize();
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if (bit_rate == 0)
return SignedSongTime::Negative();
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return SongTime::FromScale(size, bit_rate / 8);
} else
return SignedSongTime::Negative();
}
static NeAACDecHandle
faad_decoder_new()
{
const NeAACDecHandle decoder = NeAACDecOpen();
NeAACDecConfigurationPtr config =
NeAACDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 1;
config->dontUpSampleImplicitSBR = 0;
NeAACDecSetConfiguration(decoder, config);
return decoder;
}
/**
* Wrapper for NeAACDecInit() which works around some API
* inconsistencies in libfaad.
*/
static bool
faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer &buffer,
AudioFormat &audio_format, Error &error)
{
auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
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if (data.IsEmpty()) {
error.Set(faad_decoder_domain, "Empty file");
return false;
}
uint8_t channels;
unsigned long sample_rate;
long nbytes = NeAACDecInit(decoder,
/* deconst hack, libfaad requires this */
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const_cast<unsigned char *>(data.data),
data.size,
&sample_rate, &channels);
if (nbytes < 0) {
error.Set(faad_decoder_domain, "Not an AAC stream");
return false;
}
buffer.Consume(nbytes);
return audio_format_init_checked(audio_format, sample_rate,
SampleFormat::S16, channels, error);
}
/**
* Wrapper for NeAACDecDecode() which works around some API
* inconsistencies in libfaad.
*/
static const void *
faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer &buffer,
NeAACDecFrameInfo *frame_info)
{
auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
if (data.IsEmpty())
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return nullptr;
return NeAACDecDecode(decoder, frame_info,
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/* deconst hack, libfaad requires this */
const_cast<uint8_t *>(data.data),
data.size);
}
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/**
* Determine a song file's total playing time.
*
* The first return value specifies whether the file was recognized.
* The second return value is the duration.
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*/
static std::pair<bool, SignedSongTime>
faad_get_file_time(InputStream &is)
{
DecoderBuffer buffer(nullptr, is,
FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
auto duration = faad_song_duration(buffer, is);
bool recognized = !duration.IsNegative();
if (!recognized) {
NeAACDecHandle decoder = faad_decoder_new();
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AtScopeExit(decoder) { NeAACDecClose(decoder); };
buffer.Fill();
AudioFormat audio_format;
if (faad_decoder_init(decoder, buffer, audio_format,
IgnoreError()))
recognized = true;
}
return std::make_pair(recognized, duration);
}
static void
faad_stream_decode(Decoder &mpd_decoder, InputStream &is,
DecoderBuffer &buffer, const NeAACDecHandle decoder)
{
const auto total_time = faad_song_duration(buffer, is);
if (adts_find_frame(buffer) == 0)
return;
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/* initialize it */
Error error;
AudioFormat audio_format;
if (!faad_decoder_init(decoder, buffer, audio_format, error)) {
LogError(error);
return;
}
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/* initialize the MPD core */
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decoder_initialized(mpd_decoder, audio_format, false, total_time);
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/* the decoder loop */
DecoderCommand cmd;
unsigned bit_rate = 0;
do {
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/* find the next frame */
const size_t frame_size = adts_find_frame(buffer);
if (frame_size == 0)
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/* end of file */
break;
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/* decode it */
NeAACDecFrameInfo frame_info;
const void *const decoded =
faad_decoder_decode(decoder, buffer, &frame_info);
if (frame_info.error > 0) {
FormatWarning(faad_decoder_domain,
"error decoding AAC stream: %s",
NeAACDecGetErrorMessage(frame_info.error));
break;
}
if (frame_info.channels != audio_format.channels) {
FormatDefault(faad_decoder_domain,
"channel count changed from %u to %u",
audio_format.channels, frame_info.channels);
break;
}
if (frame_info.samplerate != audio_format.sample_rate) {
FormatDefault(faad_decoder_domain,
"sample rate changed from %u to %lu",
audio_format.sample_rate,
(unsigned long)frame_info.samplerate);
break;
}
buffer.Consume(frame_info.bytesconsumed);
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/* 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;
}
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/* send PCM samples to MPD */
cmd = decoder_data(mpd_decoder, is, decoded,
(size_t)frame_info.samples * 2,
bit_rate);
} while (cmd != DecoderCommand::STOP);
}
static void
faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
{
DecoderBuffer buffer(&mpd_decoder, is,
FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
/* create the libfaad decoder */
const NeAACDecHandle decoder = faad_decoder_new();
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AtScopeExit(decoder) { NeAACDecClose(decoder); };
faad_stream_decode(mpd_decoder, is, buffer, decoder);
}
static bool
faad_scan_stream(InputStream &is,
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const TagHandler &handler, void *handler_ctx)
{
auto result = faad_get_file_time(is);
if (!result.first)
return false;
if (!result.second.IsNegative())
tag_handler_invoke_duration(handler, handler_ctx,
SongTime(result.second));
return true;
}
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static const char *const faad_suffixes[] = { "aac", nullptr };
static const char *const faad_mime_types[] = {
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"audio/aac", "audio/aacp", nullptr
};
const DecoderPlugin faad_decoder_plugin = {
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"faad",
nullptr,
nullptr,
faad_stream_decode,
nullptr,
nullptr,
faad_scan_stream,
nullptr,
faad_suffixes,
faad_mime_types,
};