renamed src/inputPlugins/ to src/decoder/
These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name.
This commit is contained in:
Max Kellermann
602
src/decoder/aac_plugin.c
Normal file
602
src/decoder/aac_plugin.c
Normal file
@@ -0,0 +1,602 @@
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/* the Music Player Daemon (MPD)
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* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "../decoder_api.h"
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#define AAC_MAX_CHANNELS 6
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#include "../utils.h"
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#include "../log.h"
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#include <assert.h>
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#include <faad.h>
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/* all code here is either based on or copied from FAAD2's frontend code */
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typedef struct {
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struct decoder *decoder;
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InputStream *inStream;
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size_t bytesIntoBuffer;
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size_t bytesConsumed;
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off_t fileOffset;
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unsigned char *buffer;
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int atEof;
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} AacBuffer;
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static void aac_buffer_shift(AacBuffer * b, size_t length)
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{
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assert(length >= b->bytesConsumed);
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assert(length <= b->bytesConsumed + b->bytesIntoBuffer);
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memmove(b->buffer, b->buffer + length,
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b->bytesConsumed + b->bytesIntoBuffer - length);
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length -= b->bytesConsumed;
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b->bytesConsumed = 0;
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b->bytesIntoBuffer -= length;
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}
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static void fillAacBuffer(AacBuffer * b)
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{
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size_t bread;
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if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS)
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/* buffer already full */
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return;
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aac_buffer_shift(b, b->bytesConsumed);
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if (!b->atEof) {
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size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS -
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b->bytesIntoBuffer;
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bread = decoder_read(b->decoder, b->inStream,
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(void *)(b->buffer + b->bytesIntoBuffer),
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rest);
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if (bread == 0 && inputStreamAtEOF(b->inStream))
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b->atEof = 1;
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b->bytesIntoBuffer += bread;
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}
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if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) ||
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(b->bytesIntoBuffer > 11 &&
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memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) ||
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(b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0))
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b->bytesIntoBuffer = 0;
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}
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static void advanceAacBuffer(AacBuffer * b, size_t bytes)
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{
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b->fileOffset += bytes;
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b->bytesConsumed = bytes;
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b->bytesIntoBuffer -= bytes;
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}
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static int adtsSampleRates[] =
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{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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/**
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* Check whether the buffer head is an AAC frame, and return the frame
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* length. Returns 0 if it is not a frame.
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*/
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static size_t adts_check_frame(AacBuffer * b)
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{
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if (b->bytesIntoBuffer <= 7)
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return 0;
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/* check syncword */
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if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)))
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return 0;
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return (((unsigned int)b->buffer[3] & 0x3) << 11) |
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(((unsigned int)b->buffer[4]) << 3) |
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(b->buffer[5] >> 5);
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}
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/**
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* Find the next AAC frame in the buffer. Returns 0 if no frame is
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* found or if not enough data is available.
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*/
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static size_t adts_find_frame(AacBuffer * b)
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{
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const unsigned char *p;
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size_t frame_length;
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while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) {
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/* discard data before 0xff */
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if (p > b->buffer)
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aac_buffer_shift(b, p - b->buffer);
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if (b->bytesIntoBuffer <= 7)
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/* not enough data yet */
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return 0;
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/* is it a frame? */
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frame_length = adts_check_frame(b);
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if (frame_length > 0)
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/* yes, it is */
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return frame_length;
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/* it's just some random 0xff byte; discard and and
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continue searching */
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aac_buffer_shift(b, 1);
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}
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/* nothing at all; discard the whole buffer */
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aac_buffer_shift(b, b->bytesIntoBuffer);
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return 0;
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}
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static void adtsParse(AacBuffer * b, float *length)
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{
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unsigned int frames, frameLength;
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int sample_rate = 0;
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float framesPerSec;
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/* Read all frames to ensure correct time and bitrate */
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for (frames = 0;; frames++) {
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fillAacBuffer(b);
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frameLength = adts_find_frame(b);
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if (frameLength > 0) {
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if (frames == 0) {
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sample_rate = adtsSampleRates[(b->
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buffer[2] & 0x3c)
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>> 2];
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}
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if (frameLength > b->bytesIntoBuffer)
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break;
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advanceAacBuffer(b, frameLength);
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} else
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break;
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}
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framesPerSec = (float)sample_rate / 1024.0;
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if (framesPerSec != 0)
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*length = (float)frames / framesPerSec;
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}
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static void initAacBuffer(AacBuffer * b,
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struct decoder *decoder, InputStream * inStream)
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{
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memset(b, 0, sizeof(AacBuffer));
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b->decoder = decoder;
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b->inStream = inStream;
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b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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}
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static void aac_parse_header(AacBuffer * b, float *length)
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{
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size_t fileread;
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size_t tagsize;
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if (length)
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*length = -1;
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fileread = b->inStream->size;
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fillAacBuffer(b);
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tagsize = 0;
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if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) {
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tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
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(b->buffer[8] << 7) | (b->buffer[9] << 0);
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tagsize += 10;
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advanceAacBuffer(b, tagsize);
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fillAacBuffer(b);
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}
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if (length == NULL)
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return;
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if (b->bytesIntoBuffer >= 2 &&
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(b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
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adtsParse(b, length);
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seekInputStream(b->inStream, tagsize, SEEK_SET);
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b->bytesIntoBuffer = 0;
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b->bytesConsumed = 0;
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b->fileOffset = tagsize;
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fillAacBuffer(b);
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} else if (memcmp(b->buffer, "ADIF", 4) == 0) {
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int bitRate;
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int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
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bitRate =
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((unsigned int)(b->
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buffer[4 +
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skipSize] & 0x0F) << 19) | ((unsigned
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int)b->
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buffer[5
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+
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skipSize]
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<< 11) |
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((unsigned int)b->
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buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 +
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skipSize]
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& 0xE0);
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if (fileread != 0 && bitRate != 0)
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*length = fileread * 8.0 / bitRate;
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else
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*length = fileread;
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}
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}
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static float getAacFloatTotalTime(char *file)
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{
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AacBuffer b;
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float length;
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faacDecHandle decoder;
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faacDecConfigurationPtr config;
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uint32_t sample_rate;
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unsigned char channels;
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InputStream inStream;
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long bread;
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if (openInputStream(&inStream, file) < 0)
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return -1;
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initAacBuffer(&b, NULL, &inStream);
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aac_parse_header(&b, &length);
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if (length < 0) {
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decoder = faacDecOpen();
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config = faacDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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faacDecSetConfiguration(decoder, config);
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fillAacBuffer(&b);
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
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&sample_rate, &channels);
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#else
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bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
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#endif
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if (bread >= 0 && sample_rate > 0 && channels > 0)
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length = 0;
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faacDecClose(decoder);
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}
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if (b.buffer)
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free(b.buffer);
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closeInputStream(&inStream);
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return length;
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}
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static int getAacTotalTime(char *file)
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{
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int file_time = -1;
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float length;
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if ((length = getAacFloatTotalTime(file)) >= 0)
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file_time = length + 0.5;
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return file_time;
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}
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static int aac_stream_decode(struct decoder * mpd_decoder,
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InputStream *inStream)
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{
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float file_time;
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float totalTime = 0;
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faacDecHandle decoder;
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faacDecFrameInfo frameInfo;
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faacDecConfigurationPtr config;
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long bread;
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struct audio_format audio_format;
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uint32_t sample_rate;
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unsigned char channels;
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unsigned int sampleCount;
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char *sampleBuffer;
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size_t sampleBufferLen;
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uint16_t bitRate = 0;
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AacBuffer b;
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int initialized = 0;
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initAacBuffer(&b, mpd_decoder, inStream);
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decoder = faacDecOpen();
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config = faacDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
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config->downMatrix = 1;
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#endif
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#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
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config->dontUpSampleImplicitSBR = 0;
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#endif
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faacDecSetConfiguration(decoder, config);
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while (b.bytesIntoBuffer < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS &&
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!b.atEof &&
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decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
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fillAacBuffer(&b);
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adts_find_frame(&b);
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fillAacBuffer(&b);
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my_usleep(10000);
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}
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
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&sample_rate, &channels);
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#else
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bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
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#endif
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if (bread < 0) {
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ERROR("Error not a AAC stream.\n");
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faacDecClose(decoder);
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if (b.buffer)
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free(b.buffer);
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return -1;
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}
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audio_format.bits = 16;
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file_time = 0.0;
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advanceAacBuffer(&b, bread);
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while (1) {
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fillAacBuffer(&b);
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adts_find_frame(&b);
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fillAacBuffer(&b);
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if (b.bytesIntoBuffer == 0)
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break;
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
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b.bytesIntoBuffer);
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#else
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sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
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#endif
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if (frameInfo.error > 0) {
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ERROR("error decoding AAC stream\n");
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ERROR("faad2 error: %s\n",
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faacDecGetErrorMessage(frameInfo.error));
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break;
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}
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#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
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sample_rate = frameInfo.samplerate;
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#endif
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if (!initialized) {
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audio_format.channels = frameInfo.channels;
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audio_format.sample_rate = sample_rate;
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decoder_initialized(mpd_decoder, &audio_format, totalTime);
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initialized = 1;
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}
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advanceAacBuffer(&b, frameInfo.bytesconsumed);
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sampleCount = (unsigned long)(frameInfo.samples);
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|
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if (sampleCount > 0) {
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bitRate = frameInfo.bytesconsumed * 8.0 *
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frameInfo.channels * sample_rate /
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frameInfo.samples / 1000 + 0.5;
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file_time +=
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(float)(frameInfo.samples) / frameInfo.channels /
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sample_rate;
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}
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|
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sampleBufferLen = sampleCount * 2;
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|
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decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
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sampleBufferLen, file_time,
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bitRate, NULL);
|
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if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
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decoder_seek_error(mpd_decoder);
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} else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
|
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break;
|
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}
|
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|
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decoder_flush(mpd_decoder);
|
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|
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faacDecClose(decoder);
|
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if (b.buffer)
|
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free(b.buffer);
|
||||
|
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if (!initialized)
|
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return -1;
|
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|
||||
if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
|
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decoder_seek_error(mpd_decoder);
|
||||
}
|
||||
|
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return 0;
|
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}
|
||||
|
||||
|
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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);
|
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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
|
||||
};
|
||||
Reference in New Issue
Block a user