From cafaf33aa89f7002522712e12e26a2565e6d832d Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Mon, 16 Feb 2009 19:30:54 +0100 Subject: renamed decoder plugin "aac" to "faad" A decoder plugin should be named after the library which is used. --- src/decoder/aac_plugin.c | 469 ---------------------------------------------- src/decoder/faad_plugin.c | 469 ++++++++++++++++++++++++++++++++++++++++++++++ src/decoder_list.c | 4 +- 3 files changed, 471 insertions(+), 471 deletions(-) delete mode 100644 src/decoder/aac_plugin.c create mode 100644 src/decoder/faad_plugin.c (limited to 'src') diff --git a/src/decoder/aac_plugin.c b/src/decoder/aac_plugin.c deleted file mode 100644 index 69407541..00000000 --- a/src/decoder/aac_plugin.c +++ /dev/null @@ -1,469 +0,0 @@ -/* 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 "config.h" - -#define AAC_MAX_CHANNELS 6 - -#include -#include -#include -#include - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "aac" - -/* all code here is either based on or copied from FAAD2's frontend code */ -typedef struct { - struct decoder *decoder; - struct input_stream *inStream; - size_t bytesIntoBuffer; - size_t bytesConsumed; - off_t fileOffset; - unsigned char buffer[FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS]; -} AacBuffer; - -static void aac_buffer_shift(AacBuffer * b, size_t length) -{ - assert(length >= b->bytesConsumed); - assert(length <= b->bytesConsumed + b->bytesIntoBuffer); - - memmove(b->buffer, b->buffer + length, - b->bytesConsumed + b->bytesIntoBuffer - length); - - length -= b->bytesConsumed; - b->bytesConsumed = 0; - b->bytesIntoBuffer -= length; -} - -static void fillAacBuffer(AacBuffer * b) -{ - size_t rest, bread; - - if (b->bytesConsumed > 0) - aac_buffer_shift(b, b->bytesConsumed); - - rest = sizeof(b->buffer) - b->bytesIntoBuffer; - if (rest == 0) - /* buffer already full */ - return; - - bread = decoder_read(b->decoder, b->inStream, - (void *)(b->buffer + b->bytesIntoBuffer), - rest); - b->bytesIntoBuffer += bread; - - if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) || - (b->bytesIntoBuffer > 11 && - memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) || - (b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0)) - b->bytesIntoBuffer = 0; -} - -static void advanceAacBuffer(AacBuffer * b, size_t bytes) -{ - b->fileOffset += bytes; - b->bytesConsumed = bytes; - b->bytesIntoBuffer -= bytes; -} - -static const unsigned adtsSampleRates[] = - { 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(AacBuffer * b) -{ - if (b->bytesIntoBuffer <= 7) - return 0; - - /* check syncword */ - if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0))) - return 0; - - return (((unsigned int)b->buffer[3] & 0x3) << 11) | - (((unsigned int)b->buffer[4]) << 3) | - (b->buffer[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(AacBuffer * b) -{ - const unsigned char *p; - size_t frame_length; - - while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) { - /* discard data before 0xff */ - if (p > b->buffer) - aac_buffer_shift(b, p - b->buffer); - - if (b->bytesIntoBuffer <= 7) - /* not enough data yet */ - return 0; - - /* is it a frame? */ - frame_length = adts_check_frame(b); - if (frame_length > 0) - /* yes, it is */ - return frame_length; - - /* it's just some random 0xff byte; discard and and - continue searching */ - aac_buffer_shift(b, 1); - } - - /* nothing at all; discard the whole buffer */ - aac_buffer_shift(b, b->bytesIntoBuffer); - return 0; -} - -static void adtsParse(AacBuffer * b, float *length) -{ - unsigned int frames, frameLength; - unsigned sample_rate = 0; - float framesPerSec; - - /* Read all frames to ensure correct time and bitrate */ - for (frames = 0;; frames++) { - fillAacBuffer(b); - - frameLength = adts_find_frame(b); - if (frameLength > 0) { - if (frames == 0) { - sample_rate = adtsSampleRates[(b-> - buffer[2] & 0x3c) - >> 2]; - } - - if (frameLength > b->bytesIntoBuffer) - break; - - advanceAacBuffer(b, frameLength); - } else - break; - } - - framesPerSec = (float)sample_rate / 1024.0; - if (framesPerSec > 0) - *length = (float)frames / framesPerSec; -} - -static void -initAacBuffer(AacBuffer * b, struct decoder *decoder, - struct input_stream *inStream) -{ - memset(b, 0, sizeof(AacBuffer)); - - b->decoder = decoder; - b->inStream = inStream; -} - -static void aac_parse_header(AacBuffer * b, float *length) -{ - size_t fileread; - size_t tagsize; - - if (length) - *length = -1; - - fileread = b->inStream->size >= 0 ? b->inStream->size : 0; - - fillAacBuffer(b); - - tagsize = 0; - if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) { - tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) | - (b->buffer[8] << 7) | (b->buffer[9] << 0); - - tagsize += 10; - advanceAacBuffer(b, tagsize); - fillAacBuffer(b); - } - - if (length == NULL) - return; - - if (b->inStream->seekable && - b->bytesIntoBuffer >= 2 && - (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) { - adtsParse(b, length); - input_stream_seek(b->inStream, tagsize, SEEK_SET); - - b->bytesIntoBuffer = 0; - b->bytesConsumed = 0; - b->fileOffset = tagsize; - - fillAacBuffer(b); - } else if (memcmp(b->buffer, "ADIF", 4) == 0) { - unsigned bitRate; - size_t skipSize = (b->buffer[4] & 0x80) ? 9 : 0; - - - if (8 + skipSize > b->bytesIntoBuffer) - /* not enough data yet; skip parsing this - header */ - return; - - bitRate = - ((unsigned int)(b-> - buffer[4 + - skipSize] & 0x0F) << 19) | ((unsigned - int)b-> - buffer[5 - + - skipSize] - << 11) | - ((unsigned int)b-> - buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 + - skipSize] - & 0xE0); - - if (fileread != 0 && bitRate != 0) - *length = fileread * 8.0 / bitRate; - else - *length = fileread; - } -} - -static float getAacFloatTotalTime(const char *file) -{ - AacBuffer b; - float length; - faacDecHandle decoder; - faacDecConfigurationPtr config; - 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_r = (unsigned long *)(void *)&sample_rate; -#else - uint32_t *sample_rate_r = &sample_rate; -#endif - unsigned char channels; - struct input_stream inStream; - long bread; - - if (!input_stream_open(&inStream, file)) - return -1; - - initAacBuffer(&b, NULL, &inStream); - aac_parse_header(&b, &length); - - if (length < 0) { - decoder = faacDecOpen(); - - config = faacDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; - faacDecSetConfiguration(decoder, config); - - fillAacBuffer(&b); -#ifdef HAVE_FAAD_BUFLEN_FUNCS - bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, - sample_rate_r, &channels); -#else - bread = faacDecInit(decoder, b.buffer, sample_rate_r, &channels); -#endif - if (bread >= 0 && sample_rate > 0 && channels > 0) - length = 0; - - faacDecClose(decoder); - } - - input_stream_close(&inStream); - - return length; -} - -static int getAacTotalTime(const char *file) -{ - int file_time = -1; - float length; - - if ((length = getAacFloatTotalTime(file)) >= 0) - file_time = length + 0.5; - - return file_time; -} - -static void -aac_stream_decode(struct decoder *mpd_decoder, struct input_stream *inStream) -{ - float file_time; - float totalTime = 0; - faacDecHandle decoder; - faacDecFrameInfo frameInfo; - faacDecConfigurationPtr config; - long bread; - 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_r = (unsigned long *)(void *)&sample_rate; -#else - uint32_t *sample_rate_r = &sample_rate; -#endif - unsigned char channels; - unsigned int sampleCount; - char *sampleBuffer; - size_t sampleBufferLen; - uint16_t bitRate = 0; - AacBuffer b; - bool initialized = false; - enum decoder_command cmd; - - initAacBuffer(&b, mpd_decoder, inStream); - aac_parse_header(&b, &totalTime); - - 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 (b.bytesIntoBuffer < sizeof(b.buffer) && - !input_stream_eof(b.inStream) && - decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) { - fillAacBuffer(&b); - adts_find_frame(&b); - fillAacBuffer(&b); - } - -#ifdef HAVE_FAAD_BUFLEN_FUNCS - bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, - sample_rate_r, &channels); -#else - bread = faacDecInit(decoder, b.buffer, sample_rate_r, &channels); -#endif - if (bread < 0) { - g_warning("Error not a AAC stream.\n"); - faacDecClose(decoder); - return; - } - - file_time = 0.0; - - advanceAacBuffer(&b, bread); - - do { - fillAacBuffer(&b); - adts_find_frame(&b); - fillAacBuffer(&b); - - if (b.bytesIntoBuffer == 0) - break; - -#ifdef HAVE_FAAD_BUFLEN_FUNCS - sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer, - b.bytesIntoBuffer); -#else - sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer); -#endif - - if (frameInfo.error > 0) { - g_warning("error decoding AAC stream: %s\n", - faacDecGetErrorMessage(frameInfo.error)); - break; - } -#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE - sample_rate = frameInfo.samplerate; -#endif - - if (!initialized) { - const struct audio_format audio_format = { - .bits = 16, - .channels = frameInfo.channels, - .sample_rate = sample_rate, - }; - - if (!audio_format_valid(&audio_format)) { - g_warning("aac: invalid audio format\n"); - break; - } - - decoder_initialized(mpd_decoder, &audio_format, - false, totalTime); - initialized = true; - } - - advanceAacBuffer(&b, frameInfo.bytesconsumed); - - sampleCount = (unsigned long)(frameInfo.samples); - - if (sampleCount > 0) { - bitRate = frameInfo.bytesconsumed * 8.0 * - frameInfo.channels * sample_rate / - frameInfo.samples / 1000 + 0.5; - file_time += - (float)(frameInfo.samples) / frameInfo.channels / - sample_rate; - } - - sampleBufferLen = sampleCount * 2; - - cmd = decoder_data(mpd_decoder, inStream, sampleBuffer, - sampleBufferLen, file_time, - bitRate, NULL); - if (cmd == DECODE_COMMAND_SEEK) - decoder_seek_error(mpd_decoder); - } while (cmd != DECODE_COMMAND_STOP); - - faacDecClose(decoder); -} - -static struct tag *aacTagDup(const char *file) -{ - int file_time = getAacTotalTime(file); - struct tag *tag; - - if (file_time < 0) { - g_debug("aacTagDup: Failed to get total song time from: %s\n", - file); - return NULL; - } - - tag = tag_new(); - tag->time = file_time; - return tag; -} - -static const char *const aac_suffixes[] = { "aac", NULL }; -static const char *const aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL }; - -const struct decoder_plugin aacPlugin = { - .name = "aac", - .stream_decode = aac_stream_decode, - .tag_dup = aacTagDup, - .suffixes = aac_suffixes, - .mime_types = aac_mimeTypes -}; diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c new file mode 100644 index 00000000..15fbf030 --- /dev/null +++ b/src/decoder/faad_plugin.c @@ -0,0 +1,469 @@ +/* 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 "config.h" + +#define AAC_MAX_CHANNELS 6 + +#include +#include +#include +#include + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "aac" + +/* all code here is either based on or copied from FAAD2's frontend code */ +typedef struct { + struct decoder *decoder; + struct input_stream *inStream; + size_t bytesIntoBuffer; + size_t bytesConsumed; + off_t fileOffset; + unsigned char buffer[FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS]; +} AacBuffer; + +static void aac_buffer_shift(AacBuffer * b, size_t length) +{ + assert(length >= b->bytesConsumed); + assert(length <= b->bytesConsumed + b->bytesIntoBuffer); + + memmove(b->buffer, b->buffer + length, + b->bytesConsumed + b->bytesIntoBuffer - length); + + length -= b->bytesConsumed; + b->bytesConsumed = 0; + b->bytesIntoBuffer -= length; +} + +static void fillAacBuffer(AacBuffer * b) +{ + size_t rest, bread; + + if (b->bytesConsumed > 0) + aac_buffer_shift(b, b->bytesConsumed); + + rest = sizeof(b->buffer) - b->bytesIntoBuffer; + if (rest == 0) + /* buffer already full */ + return; + + bread = decoder_read(b->decoder, b->inStream, + (void *)(b->buffer + b->bytesIntoBuffer), + rest); + b->bytesIntoBuffer += bread; + + if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) || + (b->bytesIntoBuffer > 11 && + memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) || + (b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0)) + b->bytesIntoBuffer = 0; +} + +static void advanceAacBuffer(AacBuffer * b, size_t bytes) +{ + b->fileOffset += bytes; + b->bytesConsumed = bytes; + b->bytesIntoBuffer -= bytes; +} + +static const unsigned adtsSampleRates[] = + { 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(AacBuffer * b) +{ + if (b->bytesIntoBuffer <= 7) + return 0; + + /* check syncword */ + if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0))) + return 0; + + return (((unsigned int)b->buffer[3] & 0x3) << 11) | + (((unsigned int)b->buffer[4]) << 3) | + (b->buffer[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(AacBuffer * b) +{ + const unsigned char *p; + size_t frame_length; + + while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) { + /* discard data before 0xff */ + if (p > b->buffer) + aac_buffer_shift(b, p - b->buffer); + + if (b->bytesIntoBuffer <= 7) + /* not enough data yet */ + return 0; + + /* is it a frame? */ + frame_length = adts_check_frame(b); + if (frame_length > 0) + /* yes, it is */ + return frame_length; + + /* it's just some random 0xff byte; discard and and + continue searching */ + aac_buffer_shift(b, 1); + } + + /* nothing at all; discard the whole buffer */ + aac_buffer_shift(b, b->bytesIntoBuffer); + return 0; +} + +static void adtsParse(AacBuffer * b, float *length) +{ + unsigned int frames, frameLength; + unsigned sample_rate = 0; + float framesPerSec; + + /* Read all frames to ensure correct time and bitrate */ + for (frames = 0;; frames++) { + fillAacBuffer(b); + + frameLength = adts_find_frame(b); + if (frameLength > 0) { + if (frames == 0) { + sample_rate = adtsSampleRates[(b-> + buffer[2] & 0x3c) + >> 2]; + } + + if (frameLength > b->bytesIntoBuffer) + break; + + advanceAacBuffer(b, frameLength); + } else + break; + } + + framesPerSec = (float)sample_rate / 1024.0; + if (framesPerSec > 0) + *length = (float)frames / framesPerSec; +} + +static void +initAacBuffer(AacBuffer * b, struct decoder *decoder, + struct input_stream *inStream) +{ + memset(b, 0, sizeof(AacBuffer)); + + b->decoder = decoder; + b->inStream = inStream; +} + +static void aac_parse_header(AacBuffer * b, float *length) +{ + size_t fileread; + size_t tagsize; + + if (length) + *length = -1; + + fileread = b->inStream->size >= 0 ? b->inStream->size : 0; + + fillAacBuffer(b); + + tagsize = 0; + if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) { + tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) | + (b->buffer[8] << 7) | (b->buffer[9] << 0); + + tagsize += 10; + advanceAacBuffer(b, tagsize); + fillAacBuffer(b); + } + + if (length == NULL) + return; + + if (b->inStream->seekable && + b->bytesIntoBuffer >= 2 && + (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) { + adtsParse(b, length); + input_stream_seek(b->inStream, tagsize, SEEK_SET); + + b->bytesIntoBuffer = 0; + b->bytesConsumed = 0; + b->fileOffset = tagsize; + + fillAacBuffer(b); + } else if (memcmp(b->buffer, "ADIF", 4) == 0) { + unsigned bitRate; + size_t skipSize = (b->buffer[4] & 0x80) ? 9 : 0; + + + if (8 + skipSize > b->bytesIntoBuffer) + /* not enough data yet; skip parsing this + header */ + return; + + bitRate = + ((unsigned int)(b-> + buffer[4 + + skipSize] & 0x0F) << 19) | ((unsigned + int)b-> + buffer[5 + + + skipSize] + << 11) | + ((unsigned int)b-> + buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 + + skipSize] + & 0xE0); + + if (fileread != 0 && bitRate != 0) + *length = fileread * 8.0 / bitRate; + else + *length = fileread; + } +} + +static float getAacFloatTotalTime(const char *file) +{ + AacBuffer b; + float length; + faacDecHandle decoder; + faacDecConfigurationPtr config; + 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_r = (unsigned long *)(void *)&sample_rate; +#else + uint32_t *sample_rate_r = &sample_rate; +#endif + unsigned char channels; + struct input_stream inStream; + long bread; + + if (!input_stream_open(&inStream, file)) + return -1; + + initAacBuffer(&b, NULL, &inStream); + aac_parse_header(&b, &length); + + if (length < 0) { + decoder = faacDecOpen(); + + config = faacDecGetCurrentConfiguration(decoder); + config->outputFormat = FAAD_FMT_16BIT; + faacDecSetConfiguration(decoder, config); + + fillAacBuffer(&b); +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, + sample_rate_r, &channels); +#else + bread = faacDecInit(decoder, b.buffer, sample_rate_r, &channels); +#endif + if (bread >= 0 && sample_rate > 0 && channels > 0) + length = 0; + + faacDecClose(decoder); + } + + input_stream_close(&inStream); + + return length; +} + +static int getAacTotalTime(const char *file) +{ + int file_time = -1; + float length; + + if ((length = getAacFloatTotalTime(file)) >= 0) + file_time = length + 0.5; + + return file_time; +} + +static void +aac_stream_decode(struct decoder *mpd_decoder, struct input_stream *inStream) +{ + float file_time; + float totalTime = 0; + faacDecHandle decoder; + faacDecFrameInfo frameInfo; + faacDecConfigurationPtr config; + long bread; + 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_r = (unsigned long *)(void *)&sample_rate; +#else + uint32_t *sample_rate_r = &sample_rate; +#endif + unsigned char channels; + unsigned int sampleCount; + char *sampleBuffer; + size_t sampleBufferLen; + uint16_t bitRate = 0; + AacBuffer b; + bool initialized = false; + enum decoder_command cmd; + + initAacBuffer(&b, mpd_decoder, inStream); + aac_parse_header(&b, &totalTime); + + 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 (b.bytesIntoBuffer < sizeof(b.buffer) && + !input_stream_eof(b.inStream) && + decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) { + fillAacBuffer(&b); + adts_find_frame(&b); + fillAacBuffer(&b); + } + +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, + sample_rate_r, &channels); +#else + bread = faacDecInit(decoder, b.buffer, sample_rate_r, &channels); +#endif + if (bread < 0) { + g_warning("Error not a AAC stream.\n"); + faacDecClose(decoder); + return; + } + + file_time = 0.0; + + advanceAacBuffer(&b, bread); + + do { + fillAacBuffer(&b); + adts_find_frame(&b); + fillAacBuffer(&b); + + if (b.bytesIntoBuffer == 0) + break; + +#ifdef HAVE_FAAD_BUFLEN_FUNCS + sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer, + b.bytesIntoBuffer); +#else + sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer); +#endif + + if (frameInfo.error > 0) { + g_warning("error decoding AAC stream: %s\n", + faacDecGetErrorMessage(frameInfo.error)); + break; + } +#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE + sample_rate = frameInfo.samplerate; +#endif + + if (!initialized) { + const struct audio_format audio_format = { + .bits = 16, + .channels = frameInfo.channels, + .sample_rate = sample_rate, + }; + + if (!audio_format_valid(&audio_format)) { + g_warning("aac: invalid audio format\n"); + break; + } + + decoder_initialized(mpd_decoder, &audio_format, + false, totalTime); + initialized = true; + } + + advanceAacBuffer(&b, frameInfo.bytesconsumed); + + sampleCount = (unsigned long)(frameInfo.samples); + + if (sampleCount > 0) { + bitRate = frameInfo.bytesconsumed * 8.0 * + frameInfo.channels * sample_rate / + frameInfo.samples / 1000 + 0.5; + file_time += + (float)(frameInfo.samples) / frameInfo.channels / + sample_rate; + } + + sampleBufferLen = sampleCount * 2; + + cmd = decoder_data(mpd_decoder, inStream, sampleBuffer, + sampleBufferLen, file_time, + bitRate, NULL); + if (cmd == DECODE_COMMAND_SEEK) + decoder_seek_error(mpd_decoder); + } while (cmd != DECODE_COMMAND_STOP); + + faacDecClose(decoder); +} + +static struct tag *aacTagDup(const char *file) +{ + int file_time = getAacTotalTime(file); + struct tag *tag; + + if (file_time < 0) { + g_debug("aacTagDup: Failed to get total song time from: %s\n", + file); + return NULL; + } + + tag = tag_new(); + tag->time = file_time; + return tag; +} + +static const char *const aac_suffixes[] = { "aac", NULL }; +static const char *const aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL }; + +const struct decoder_plugin faad_decoder_plugin = { + .name = "faad", + .stream_decode = aac_stream_decode, + .tag_dup = aacTagDup, + .suffixes = aac_suffixes, + .mime_types = aac_mimeTypes +}; diff --git a/src/decoder_list.c b/src/decoder_list.c index b6093a9b..d7360ce7 100644 --- a/src/decoder_list.c +++ b/src/decoder_list.c @@ -32,7 +32,7 @@ extern const struct decoder_plugin flac_decoder_plugin; extern const struct decoder_plugin oggflac_decoder_plugin; extern const struct decoder_plugin audiofilePlugin; extern const struct decoder_plugin mp4_plugin; -extern const struct decoder_plugin aacPlugin; +extern const struct decoder_plugin faad_decoder_plugin; extern const struct decoder_plugin mpcPlugin; extern const struct decoder_plugin wavpack_plugin; extern const struct decoder_plugin modplug_plugin; @@ -59,7 +59,7 @@ static const struct decoder_plugin *const decoder_plugins[] = { &audiofilePlugin, #endif #ifdef HAVE_FAAD - &aacPlugin, + &faad_decoder_plugin, #endif #ifdef HAVE_MP4 &mp4_plugin, -- cgit v1.2.3