/* * 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 "DecoderAPI.hxx" #include "DecoderBuffer.hxx" #include "InputStream.hxx" #include "CheckAudioFormat.hxx" #include "tag/TagHandler.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) { size_t length; const uint8_t *data = (const uint8_t *) decoder_buffer_read(buffer, &length); if (data == nullptr || length < 8) { /* not enough data yet */ if (!decoder_buffer_fill(buffer)) /* 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_clear(buffer); continue; } if (p > data) { /* discard data before 0xff */ decoder_buffer_consume(buffer, p - data); continue; } /* is it a frame? */ const size_t 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 */ if (!decoder_buffer_fill(buffer)) { /* not enough data; discard this frame to prevent a possible buffer overflow */ decoder_buffer_clear(buffer); } continue; } /* found a full frame! */ return frame_length; } } static float adts_song_duration(DecoderBuffer *buffer) { unsigned sample_rate = 0; /* Read all frames to ensure correct time and bitrate */ unsigned frames; for (frames = 0;; frames++) { const unsigned 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]; if (sample_rate == 0) break; } decoder_buffer_consume(buffer, frame_length); } if (sample_rate == 0) return -1; float frames_per_second = (float)sample_rate / 1024.0; assert(frames_per_second > 0); return (float)frames / frames_per_second; } static float faad_song_duration(DecoderBuffer *buffer, InputStream &is) { const auto size = is.GetSize(); const size_t fileread = size >= 0 ? size : 0; decoder_buffer_fill(buffer); size_t length; const uint8_t *data = (const uint8_t *) decoder_buffer_read(buffer, &length); if (data == nullptr) return -1; size_t 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; const bool 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 (length >= 8 && adts_check_frame(data) > 0) { /* obtain the duration from the ADTS header */ if (!is.IsSeekable()) return -1; float song_length = adts_song_duration(buffer); is.LockSeek(tagsize, SEEK_SET, IgnoreError()); decoder_buffer_clear(buffer); return song_length; } else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) { /* obtain the duration from the ADIF header */ size_t skip_size = (data[4] & 0x80) ? 9 : 0; if (8 + skip_size > length) /* not enough data yet; skip parsing this header */ return -1; unsigned 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; } 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) { size_t length; const unsigned char *data = (const unsigned char *) decoder_buffer_read(buffer, &length); if (data == nullptr) { error.Set(faad_decoder_domain, "Empty file"); return false; } uint8_t channels; uint32_t sample_rate; #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 long nbytes = NeAACDecInit(decoder, /* deconst hack, libfaad requires this */ const_cast(data), length, sample_rate_p, &channels); if (nbytes < 0) { error.Set(faad_decoder_domain, "Not an AAC stream"); return false; } decoder_buffer_consume(buffer, 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) { 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(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(InputStream &is) { DecoderBuffer *const buffer = decoder_buffer_new(nullptr, is, FAAD_MIN_STREAMSIZE * MAX_CHANNELS); float length = faad_song_duration(buffer, is); if (length < 0) { AudioFormat audio_format; NeAACDecHandle decoder = faad_decoder_new(); decoder_buffer_fill(buffer); if (faad_decoder_init(decoder, buffer, audio_format, IgnoreError())) 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(InputStream &is) { float length = faad_get_file_time_float(is); if (length < 0) return -1; return int(length + 0.5); } static void faad_stream_decode(Decoder &mpd_decoder, InputStream &is) { DecoderBuffer *const buffer = decoder_buffer_new(&mpd_decoder, is, FAAD_MIN_STREAMSIZE * MAX_CHANNELS); const float total_time = faad_song_duration(buffer, is); /* create the libfaad decoder */ const NeAACDecHandle decoder = faad_decoder_new(); while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() && decoder_get_command(mpd_decoder) == DecoderCommand::NONE) { adts_find_frame(buffer); decoder_buffer_fill(buffer); } /* initialize it */ Error error; AudioFormat audio_format; if (!faad_decoder_init(decoder, buffer, audio_format, error)) { LogError(error); NeAACDecClose(decoder); decoder_buffer_free(buffer); return; } /* initialize the MPD core */ decoder_initialized(mpd_decoder, audio_format, false, total_time); /* the decoder loop */ DecoderCommand cmd; uint16_t 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; } 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 != DecoderCommand::STOP); /* cleanup */ NeAACDecClose(decoder); decoder_buffer_free(buffer); } static bool faad_scan_stream(InputStream &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 DecoderPlugin faad_decoder_plugin = { "faad", nullptr, nullptr, faad_stream_decode, nullptr, nullptr, faad_scan_stream, nullptr, faad_suffixes, faad_mime_types, };