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author | Max Kellermann <max@duempel.org> | 2009-02-16 19:30:54 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2009-02-16 19:30:54 +0100 |
commit | cafaf33aa89f7002522712e12e26a2565e6d832d (patch) | |
tree | dc63655d6a3a3dd2ff9d1b549921af918a6b1e30 /src/decoder/faad_plugin.c | |
parent | c5edb53797fb15b4753688b1a653bdfda53f25aa (diff) | |
download | mpd-cafaf33aa89f7002522712e12e26a2565e6d832d.tar.gz mpd-cafaf33aa89f7002522712e12e26a2565e6d832d.tar.xz mpd-cafaf33aa89f7002522712e12e26a2565e6d832d.zip |
renamed decoder plugin "aac" to "faad"
A decoder plugin should be named after the library which is used.
Diffstat (limited to 'src/decoder/faad_plugin.c')
-rw-r--r-- | src/decoder/faad_plugin.c | 469 |
1 files changed, 469 insertions, 0 deletions
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c new file mode 100644 index 000000000..15fbf030e --- /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 <assert.h> +#include <unistd.h> +#include <faad.h> +#include <glib.h> + +#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 +}; |