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-rw-r--r--src/decoder/faad_decoder_plugin.c515
1 files changed, 515 insertions, 0 deletions
diff --git a/src/decoder/faad_decoder_plugin.c b/src/decoder/faad_decoder_plugin.c
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+++ b/src/decoder/faad_decoder_plugin.c
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+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "decoder_api.h"
+#include "decoder_buffer.h"
+#include "audio_check.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 "faad"
+
+static const unsigned adts_sample_rates[] =
+ { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+/**
+ * The GLib quark used for errors reported by this plugin.
+ */
+static inline GQuark
+faad_decoder_quark(void)
+{
+ return g_quark_from_static_string("faad");
+}
+
+/**
+ * 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(const unsigned char *data)
+{
+ /* check syncword */
+ if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0)))
+ return 0;
+
+ return (((unsigned int)data[3] & 0x3) << 11) |
+ (((unsigned int)data[4]) << 3) |
+ (data[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(struct decoder_buffer *buffer)
+{
+ const unsigned char *data, *p;
+ size_t length, frame_length;
+ bool ret;
+
+ while (true) {
+ data = decoder_buffer_read(buffer, &length);
+ if (data == NULL || length < 8) {
+ /* not enough data yet */
+ ret = decoder_buffer_fill(buffer);
+ if (!ret)
+ /* failed */
+ return 0;
+
+ continue;
+ }
+
+ /* find the 0xff marker */
+ p = memchr(data, 0xff, length);
+ if (p == NULL) {
+ /* no marker - discard the buffer */
+ decoder_buffer_consume(buffer, length);
+ continue;
+ }
+
+ if (p > data) {
+ /* discard data before 0xff */
+ decoder_buffer_consume(buffer, p - data);
+ continue;
+ }
+
+ /* is it a frame? */
+ frame_length = adts_check_frame(data);
+ if (frame_length == 0) {
+ /* it's just some random 0xff byte; discard it
+ and continue searching */
+ decoder_buffer_consume(buffer, 1);
+ continue;
+ }
+
+ if (length < frame_length) {
+ /* available buffer size is smaller than the
+ frame will be - attempt to read more
+ data */
+ ret = decoder_buffer_fill(buffer);
+ if (!ret) {
+ /* not enough data; discard this frame
+ to prevent a possible buffer
+ overflow */
+ data = decoder_buffer_read(buffer, &length);
+ if (data != NULL)
+ decoder_buffer_consume(buffer, length);
+ }
+
+ continue;
+ }
+
+ /* found a full frame! */
+ return frame_length;
+ }
+}
+
+static float
+adts_song_duration(struct decoder_buffer *buffer)
+{
+ unsigned int frames, frame_length;
+ unsigned sample_rate = 0;
+ float frames_per_second;
+
+ /* Read all frames to ensure correct time and bitrate */
+ for (frames = 0;; frames++) {
+ frame_length = adts_find_frame(buffer);
+ if (frame_length == 0)
+ break;
+
+
+ if (frames == 0) {
+ const unsigned char *data;
+ size_t buffer_length;
+
+ data = decoder_buffer_read(buffer, &buffer_length);
+ assert(data != NULL);
+ assert(frame_length <= buffer_length);
+
+ sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
+ }
+
+ decoder_buffer_consume(buffer, frame_length);
+ }
+
+ frames_per_second = (float)sample_rate / 1024.0;
+ if (frames_per_second <= 0)
+ return -1;
+
+ return (float)frames / frames_per_second;
+}
+
+static float
+faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
+{
+ size_t fileread;
+ size_t tagsize;
+ const unsigned char *data;
+ size_t length;
+ bool success;
+
+ fileread = is->size >= 0 ? is->size : 0;
+
+ decoder_buffer_fill(buffer);
+ data = decoder_buffer_read(buffer, &length);
+ if (data == NULL)
+ return -1;
+
+ tagsize = 0;
+ if (length >= 10 && !memcmp(data, "ID3", 3)) {
+ /* skip the ID3 tag */
+
+ tagsize = (data[6] << 21) | (data[7] << 14) |
+ (data[8] << 7) | (data[9] << 0);
+
+ tagsize += 10;
+
+ success = decoder_buffer_skip(buffer, tagsize) &&
+ decoder_buffer_fill(buffer);
+ if (!success)
+ return -1;
+
+ data = decoder_buffer_read(buffer, &length);
+ if (data == NULL)
+ return -1;
+ }
+
+ if (is->seekable && length >= 2 &&
+ data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) {
+ /* obtain the duration from the ADTS header */
+ float song_length = adts_song_duration(buffer);
+
+ input_stream_seek(is, tagsize, SEEK_SET, NULL);
+
+ data = decoder_buffer_read(buffer, &length);
+ if (data != NULL)
+ decoder_buffer_consume(buffer, length);
+ decoder_buffer_fill(buffer);
+
+ return song_length;
+ } else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) {
+ /* obtain the duration from the ADIF header */
+ unsigned bit_rate;
+ size_t skip_size = (data[4] & 0x80) ? 9 : 0;
+
+ if (8 + skip_size > length)
+ /* not enough data yet; skip parsing this
+ header */
+ return -1;
+
+ bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
+ (data[5 + skip_size] << 11) |
+ (data[6 + skip_size] << 3) |
+ (data[7 + skip_size] & 0xE0);
+
+ if (fileread != 0 && bit_rate != 0)
+ return fileread * 8.0 / bit_rate;
+ else
+ return fileread;
+ } else
+ return -1;
+}
+
+/**
+ * Wrapper for faacDecInit() which works around some API
+ * inconsistencies in libfaad.
+ */
+static bool
+faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
+ struct audio_format *audio_format, GError **error_r)
+{
+ union {
+ /* deconst hack for libfaad */
+ const void *in;
+ void *out;
+ } u;
+ size_t length;
+ int32_t nbytes;
+ uint32_t sample_rate;
+ uint8_t channels;
+#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_p = (unsigned long *)(void *)&sample_rate;
+#else
+ uint32_t *sample_rate_p = &sample_rate;
+#endif
+
+ u.in = decoder_buffer_read(buffer, &length);
+ if (u.in == NULL) {
+ g_set_error(error_r, faad_decoder_quark(), 0,
+ "Empty file");
+ return false;
+ }
+
+ nbytes = faacDecInit(decoder, u.out,
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ length,
+#endif
+ sample_rate_p, &channels);
+ if (nbytes < 0) {
+ g_set_error(error_r, faad_decoder_quark(), 0,
+ "Not an AAC stream");
+ return false;
+ }
+
+ decoder_buffer_consume(buffer, nbytes);
+
+ return audio_format_init_checked(audio_format, sample_rate,
+ SAMPLE_FORMAT_S16, channels, error_r);
+}
+
+/**
+ * Wrapper for faacDecDecode() which works around some API
+ * inconsistencies in libfaad.
+ */
+static const void *
+faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer,
+ faacDecFrameInfo *frame_info)
+{
+ union {
+ /* deconst hack for libfaad */
+ const void *in;
+ void *out;
+ } u;
+ size_t length;
+ void *result;
+
+ u.in = decoder_buffer_read(buffer, &length);
+ if (u.in == NULL)
+ return NULL;
+
+ result = faacDecDecode(decoder, frame_info,
+ u.out
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ , length
+#endif
+ );
+
+ return result;
+}
+
+/**
+ * Get a song file's total playing time in seconds, as a float.
+ * Returns 0 if the duration is unknown, and a negative value if the
+ * file is invalid.
+ */
+static float
+faad_get_file_time_float(struct input_stream *is)
+{
+ struct decoder_buffer *buffer;
+ float length;
+ faacDecHandle decoder;
+ faacDecConfigurationPtr config;
+
+ buffer = decoder_buffer_new(NULL, is,
+ FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+ length = faad_song_duration(buffer, is);
+
+ if (length < 0) {
+ bool ret;
+ struct audio_format audio_format;
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+ faacDecSetConfiguration(decoder, config);
+
+ decoder_buffer_fill(buffer);
+
+ ret = faad_decoder_init(decoder, buffer, &audio_format, NULL);
+ if (ret)
+ length = 0;
+
+ faacDecClose(decoder);
+ }
+
+ decoder_buffer_free(buffer);
+
+ return length;
+}
+
+/**
+ * Get a song file's total playing time in seconds, as an int.
+ * Returns 0 if the duration is unknown, and a negative value if the
+ * file is invalid.
+ */
+static int
+faad_get_file_time(struct input_stream *is)
+{
+ int file_time = -1;
+ float length;
+
+ if ((length = faad_get_file_time_float(is)) >= 0)
+ file_time = length + 0.5;
+
+ return file_time;
+}
+
+static void
+faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
+{
+ GError *error = NULL;
+ float total_time = 0;
+ faacDecHandle decoder;
+ struct audio_format audio_format;
+ faacDecConfigurationPtr config;
+ bool ret;
+ uint16_t bit_rate = 0;
+ struct decoder_buffer *buffer;
+ enum decoder_command cmd;
+
+ buffer = decoder_buffer_new(mpd_decoder, is,
+ FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+ total_time = faad_song_duration(buffer, is);
+
+ /* create the libfaad decoder */
+
+ 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 (!decoder_buffer_is_full(buffer) &&
+ !input_stream_eof(is) &&
+ decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+ adts_find_frame(buffer);
+ decoder_buffer_fill(buffer);
+ }
+
+ /* initialize it */
+
+ ret = faad_decoder_init(decoder, buffer, &audio_format, &error);
+ if (!ret) {
+ g_warning("%s", error->message);
+ g_error_free(error);
+ faacDecClose(decoder);
+ return;
+ }
+
+ /* initialize the MPD core */
+
+ decoder_initialized(mpd_decoder, &audio_format, false, total_time);
+
+ /* the decoder loop */
+
+ do {
+ size_t frame_size;
+ const void *decoded;
+ faacDecFrameInfo frame_info;
+
+ /* find the next frame */
+
+ frame_size = adts_find_frame(buffer);
+ if (frame_size == 0)
+ /* end of file */
+ break;
+
+ /* decode it */
+
+ decoded = faad_decoder_decode(decoder, buffer, &frame_info);
+
+ if (frame_info.error > 0) {
+ g_warning("error decoding AAC stream: %s\n",
+ faacDecGetErrorMessage(frame_info.error));
+ break;
+ }
+
+ if (frame_info.channels != audio_format.channels) {
+ g_warning("channel count changed from %u to %u",
+ audio_format.channels, frame_info.channels);
+ break;
+ }
+
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ if (frame_info.samplerate != audio_format.sample_rate) {
+ g_warning("sample rate changed from %u to %lu",
+ audio_format.sample_rate,
+ (unsigned long)frame_info.samplerate);
+ break;
+ }
+#endif
+
+ decoder_buffer_consume(buffer, frame_info.bytesconsumed);
+
+ /* update bit rate and position */
+
+ if (frame_info.samples > 0) {
+ bit_rate = frame_info.bytesconsumed * 8.0 *
+ frame_info.channels * audio_format.sample_rate /
+ frame_info.samples / 1000 + 0.5;
+ }
+
+ /* send PCM samples to MPD */
+
+ cmd = decoder_data(mpd_decoder, is, decoded,
+ (size_t)frame_info.samples * 2,
+ bit_rate);
+ } while (cmd != DECODE_COMMAND_STOP);
+
+ /* cleanup */
+
+ faacDecClose(decoder);
+}
+
+static struct tag *
+faad_stream_tag(struct input_stream *is)
+{
+ int file_time = faad_get_file_time(is);
+ struct tag *tag;
+
+ if (file_time < 0)
+ return NULL;
+
+ tag = tag_new();
+ tag->time = file_time;
+ return tag;
+}
+
+static const char *const faad_suffixes[] = { "aac", NULL };
+static const char *const faad_mime_types[] = {
+ "audio/aac", "audio/aacp", NULL
+};
+
+const struct decoder_plugin faad_decoder_plugin = {
+ .name = "faad",
+ .stream_decode = faad_stream_decode,
+ .stream_tag = faad_stream_tag,
+ .suffixes = faad_suffixes,
+ .mime_types = faad_mime_types,
+};