From e11355f47d545fe523b019481415b1347aecd4bd Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Sun, 26 Oct 2008 11:29:25 +0100 Subject: renamed src/inputPlugins/ to src/decoder/ These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name. --- src/decoder/aac_plugin.c | 602 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 602 insertions(+) create mode 100644 src/decoder/aac_plugin.c (limited to 'src/decoder/aac_plugin.c') diff --git a/src/decoder/aac_plugin.c b/src/decoder/aac_plugin.c new file mode 100644 index 000000000..7842bcc22 --- /dev/null +++ b/src/decoder/aac_plugin.c @@ -0,0 +1,602 @@ +/* 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" + +#define AAC_MAX_CHANNELS 6 + +#include "../utils.h" +#include "../log.h" + +#include +#include + +/* all code here is either based on or copied from FAAD2's frontend code */ +typedef struct { + struct decoder *decoder; + InputStream *inStream; + size_t bytesIntoBuffer; + size_t bytesConsumed; + off_t fileOffset; + unsigned char *buffer; + int atEof; +} 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 bread; + + if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS) + /* buffer already full */ + return; + + aac_buffer_shift(b, b->bytesConsumed); + + if (!b->atEof) { + size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS - + b->bytesIntoBuffer; + + bread = decoder_read(b->decoder, b->inStream, + (void *)(b->buffer + b->bytesIntoBuffer), + rest); + if (bread == 0 && inputStreamAtEOF(b->inStream)) + b->atEof = 1; + 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 int 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; + int 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, InputStream * inStream) +{ + memset(b, 0, sizeof(AacBuffer)); + + b->decoder = decoder; + b->inStream = inStream; + + b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); + memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); +} + +static void aac_parse_header(AacBuffer * b, float *length) +{ + size_t fileread; + size_t tagsize; + + if (length) + *length = -1; + + fileread = b->inStream->size; + + 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->bytesIntoBuffer >= 2 && + (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) { + adtsParse(b, length); + seekInputStream(b->inStream, tagsize, SEEK_SET); + + b->bytesIntoBuffer = 0; + b->bytesConsumed = 0; + b->fileOffset = tagsize; + + fillAacBuffer(b); + } else if (memcmp(b->buffer, "ADIF", 4) == 0) { + int bitRate; + int skipSize = (b->buffer[4] & 0x80) ? 9 : 0; + 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(char *file) +{ + AacBuffer b; + float length; + faacDecHandle decoder; + faacDecConfigurationPtr config; + uint32_t sample_rate; + unsigned char channels; + InputStream inStream; + long bread; + + if (openInputStream(&inStream, file) < 0) + 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, &channels); +#else + bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels); +#endif + if (bread >= 0 && sample_rate > 0 && channels > 0) + length = 0; + + faacDecClose(decoder); + } + + if (b.buffer) + free(b.buffer); + closeInputStream(&inStream); + + return length; +} + +static int getAacTotalTime(char *file) +{ + int file_time = -1; + float length; + + if ((length = getAacFloatTotalTime(file)) >= 0) + file_time = length + 0.5; + + return file_time; +} + +static int aac_stream_decode(struct decoder * mpd_decoder, + InputStream *inStream) +{ + float file_time; + float totalTime = 0; + faacDecHandle decoder; + faacDecFrameInfo frameInfo; + faacDecConfigurationPtr config; + long bread; + struct audio_format audio_format; + uint32_t sample_rate; + unsigned char channels; + unsigned int sampleCount; + char *sampleBuffer; + size_t sampleBufferLen; + uint16_t bitRate = 0; + AacBuffer b; + int initialized = 0; + + initAacBuffer(&b, mpd_decoder, inStream); + + 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 < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS && + !b.atEof && + decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) { + fillAacBuffer(&b); + adts_find_frame(&b); + fillAacBuffer(&b); + my_usleep(10000); + } + +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, + &sample_rate, &channels); +#else + bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels); +#endif + if (bread < 0) { + ERROR("Error not a AAC stream.\n"); + faacDecClose(decoder); + if (b.buffer) + free(b.buffer); + return -1; + } + + audio_format.bits = 16; + + file_time = 0.0; + + advanceAacBuffer(&b, bread); + + while (1) { + 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) { + ERROR("error decoding AAC stream\n"); + ERROR("faad2 error: %s\n", + faacDecGetErrorMessage(frameInfo.error)); + break; + } +#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE + sample_rate = frameInfo.samplerate; +#endif + + if (!initialized) { + audio_format.channels = frameInfo.channels; + audio_format.sample_rate = sample_rate; + decoder_initialized(mpd_decoder, &audio_format, totalTime); + initialized = 1; + } + + 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; + + decoder_data(mpd_decoder, NULL, 0, sampleBuffer, + sampleBufferLen, file_time, + bitRate, NULL); + if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) { + decoder_seek_error(mpd_decoder); + } else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) + break; + } + + decoder_flush(mpd_decoder); + + faacDecClose(decoder); + if (b.buffer) + free(b.buffer); + + if (!initialized) + return -1; + + if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) { + decoder_seek_error(mpd_decoder); + } + + return 0; +} + + +static int aac_decode(struct decoder * mpd_decoder, char *path) +{ + float file_time; + float totalTime; + faacDecHandle decoder; + faacDecFrameInfo frameInfo; + faacDecConfigurationPtr config; + long bread; + struct audio_format audio_format; + uint32_t sample_rate; + unsigned char channels; + unsigned int sampleCount; + char *sampleBuffer; + size_t sampleBufferLen; + /*float * seekTable; + long seekTableEnd = -1; + int seekPositionFound = 0; */ + uint16_t bitRate = 0; + AacBuffer b; + InputStream inStream; + int initialized = 0; + + if ((totalTime = getAacFloatTotalTime(path)) < 0) + return -1; + + if (openInputStream(&inStream, path) < 0) + return -1; + + initAacBuffer(&b, mpd_decoder, &inStream); + aac_parse_header(&b, NULL); + + 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); + + fillAacBuffer(&b); + +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer, + &sample_rate, &channels); +#else + bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels); +#endif + if (bread < 0) { + ERROR("Error not a AAC stream.\n"); + faacDecClose(decoder); + if (b.buffer) + free(b.buffer); + return -1; + } + + audio_format.bits = 16; + + file_time = 0.0; + + advanceAacBuffer(&b, bread); + + while (1) { + 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) { + ERROR("error decoding AAC file: %s\n", path); + ERROR("faad2 error: %s\n", + faacDecGetErrorMessage(frameInfo.error)); + break; + } +#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE + sample_rate = frameInfo.samplerate; +#endif + + if (!initialized) { + audio_format.channels = frameInfo.channels; + audio_format.sample_rate = sample_rate; + decoder_initialized(mpd_decoder, &audio_format, + totalTime); + initialized = 1; + } + + 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; + + decoder_data(mpd_decoder, NULL, 0, sampleBuffer, + sampleBufferLen, file_time, + bitRate, NULL); + if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) { + decoder_seek_error(mpd_decoder); + } else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) + break; + } + + decoder_flush(mpd_decoder); + + faacDecClose(decoder); + if (b.buffer) + free(b.buffer); + + if (!initialized) + return -1; + + if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) { + decoder_seek_error(mpd_decoder); + } + + return 0; +} + +static struct tag *aacTagDup(char *file) +{ + struct tag *ret = NULL; + int file_time = getAacTotalTime(file); + + if (file_time >= 0) { + if ((ret = tag_id3_load(file)) == NULL) + ret = tag_new(); + ret->time = file_time; + } else { + DEBUG("aacTagDup: Failed to get total song time from: %s\n", + file); + } + + return ret; +} + +static const char *aac_suffixes[] = { "aac", NULL }; +static const char *aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL }; + +struct decoder_plugin aacPlugin = { + .name = "aac", + .stream_decode = aac_stream_decode, + .file_decode = aac_decode, + .tag_dup = aacTagDup, + .stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL, + .suffixes = aac_suffixes, + .mime_types = aac_mimeTypes +}; -- cgit v1.2.3