/* * Copyright (C) 2003-2014 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 "../DecoderAPI.hxx" #include "../DecoderBuffer.hxx" #include "input/InputStream.hxx" #include "CheckAudioFormat.hxx" #include "tag/TagHandler.hxx" #include "util/ConstBuffer.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" #include "Log.hxx" #include #include #include #include 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::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); 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? */ 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::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(frames * uint64_t(1024), sample_rate); } static SignedSongTime faad_song_duration(DecoderBuffer &buffer, InputStream &is) { auto data = ConstBuffer::FromVoid(buffer.Need(5)); if (data.IsNull()) return SignedSongTime::Negative(); size_t tagsize = 0; if (data.size >= 10 && !memcmp(data.data, "ID3", 3)) { /* 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::FromVoid(buffer.Need(5)); if (data.IsNull()) return SignedSongTime::Negative(); } if (data.size >= 8 && adts_check_frame(data.data) > 0) { /* obtain the duration from the ADTS header */ if (!is.IsSeekable()) return SignedSongTime::Negative(); auto song_length = adts_song_duration(buffer); is.LockSeek(tagsize, IgnoreError()); buffer.Clear(); return song_length; } else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) { /* 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(); 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(); if (bit_rate == 0) return SignedSongTime::Negative(); 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::FromVoid(buffer.Read()); 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 */ const_cast(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::FromVoid(buffer.Read()); if (data.IsEmpty()) return nullptr; return NeAACDecDecode(decoder, frame_info, /* deconst hack, libfaad requires this */ const_cast(data.data), data.size); } /** * 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. */ static std::pair 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(); buffer.Fill(); AudioFormat audio_format; if (faad_decoder_init(decoder, buffer, audio_format, IgnoreError())) recognized = true; NeAACDecClose(decoder); } 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; /* initialize it */ Error error; AudioFormat audio_format; if (!faad_decoder_init(decoder, buffer, audio_format, error)) { LogError(error); return; } /* initialize the MPD core */ decoder_initialized(mpd_decoder, audio_format, false, total_time); /* the decoder loop */ DecoderCommand cmd; unsigned bit_rate = 0; do { /* find the next frame */ const size_t frame_size = adts_find_frame(buffer); if (frame_size == 0) /* end of file */ break; /* 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); /* 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 != 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(); faad_stream_decode(mpd_decoder, is, buffer, decoder); /* cleanup */ NeAACDecClose(decoder); } static bool faad_scan_stream(InputStream &is, const struct tag_handler *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; } static const char *const faad_suffixes[] = { "aac", nullptr }; static const char *const faad_mime_types[] = { "audio/aac", "audio/aacp", nullptr }; const DecoderPlugin faad_decoder_plugin = { "faad", nullptr, nullptr, faad_stream_decode, nullptr, nullptr, faad_scan_stream, nullptr, faad_suffixes, faad_mime_types, };