1 files changed, 515 insertions, 0 deletions
diff --git a/src/decoder/faad_decoder_plugin.c b/src/decoder/faad_decoder_plugin.c
new file mode 100644
index 000000000..8f932ad58
--- /dev/null
+++ b/src/decoder/faad_decoder_plugin.c
@@ -0,0 +1,515 @@
+/*
+ * 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,
+};