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-rw-r--r--src/decoder/faad_plugin.c469
1 files changed, 469 insertions, 0 deletions
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
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+++ b/src/decoder/faad_plugin.c
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+/* 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
+};