1 files changed, 476 insertions, 0 deletions
diff --git a/src/decoder/plugins/FaadDecoderPlugin.cxx b/src/decoder/plugins/FaadDecoderPlugin.cxx
new file mode 100644
index 000000000..90e5dc40f
--- /dev/null
+++ b/src/decoder/plugins/FaadDecoderPlugin.cxx
@@ -0,0 +1,476 @@
+/*
+ * Copyright (C) 2003-2014 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 "FaadDecoderPlugin.hxx"
+#include "../DecoderAPI.hxx"
+#include "../DecoderBuffer.hxx"
+#include "input/InputStream.hxx"
+#include "CheckAudioFormat.hxx"
+#include "tag/TagHandler.hxx"
+#include "util/ConstBuffer.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+#include <neaacdec.h>
+
+#include <assert.h>
+#include <string.h>
+#include <unistd.h>
+
+static const unsigned adts_sample_rates[] =
+    { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+	16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+static constexpr Domain faad_decoder_domain("faad_decoder");
+
+/**
+ * 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(DecoderBuffer *buffer)
+{
+	while (true) {
+		auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 8));
+		if (data.IsNull())
+			/* failed */
+			return 0;
+
+		/* find the 0xff marker */
+		const uint8_t *p = (const uint8_t *)
+			memchr(data.data, 0xff, data.size);
+		if (p == nullptr) {
+			/* no marker - discard the buffer */
+			decoder_buffer_clear(buffer);
+			continue;
+		}
+
+		if (p > data.data) {
+			/* discard data before 0xff */
+			decoder_buffer_consume(buffer, p - data.data);
+			continue;
+		}
+
+		/* is it a frame? */
+		const size_t frame_length = adts_check_frame(data.data);
+		if (frame_length == 0) {
+			/* it's just some random 0xff byte; discard it
+			   and continue searching */
+			decoder_buffer_consume(buffer, 1);
+			continue;
+		}
+
+		if (decoder_buffer_need(buffer, frame_length).IsNull()) {
+			/* not enough data; discard this frame to
+			   prevent a possible buffer overflow */
+			decoder_buffer_clear(buffer);
+			continue;
+		}
+
+		/* found a full frame! */
+		return frame_length;
+	}
+}
+
+static float
+adts_song_duration(DecoderBuffer *buffer)
+{
+	const InputStream &is = decoder_buffer_get_stream(buffer);
+	const bool estimate = !is.CheapSeeking();
+	if (estimate && !is.KnownSize())
+		return -1;
+
+	unsigned sample_rate = 0;
+
+	/* Read all frames to ensure correct time and bitrate */
+	unsigned frames = 0;
+	for (;; frames++) {
+		const unsigned frame_length = adts_find_frame(buffer);
+		if (frame_length == 0)
+			break;
+
+		if (frames == 0) {
+			auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+			assert(!data.IsEmpty());
+			assert(frame_length <= data.size);
+
+			sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
+			if (sample_rate == 0)
+				break;
+		}
+
+		decoder_buffer_consume(buffer, frame_length);
+
+		if (estimate && frames == 128) {
+			/* if this is a remote file, don't slurp the
+			   whole file just for checking the song
+			   duration; instead, stop after some time and
+			   extrapolate the song duration from what we
+			   have until now */
+
+			const auto offset = is.GetOffset()
+				- decoder_buffer_available(buffer);
+			if (offset <= 0)
+				return -1;
+
+			const auto file_size = is.GetSize();
+			frames = (frames * file_size) / offset;
+			break;
+		}
+	}
+
+	if (sample_rate == 0)
+		return -1;
+
+	float frames_per_second = (float)sample_rate / 1024.0;
+	assert(frames_per_second > 0);
+
+	return (float)frames / frames_per_second;
+}
+
+static float
+faad_song_duration(DecoderBuffer *buffer, InputStream &is)
+{
+	auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 5));
+	if (data.IsNull())
+		return -1;
+
+	size_t tagsize = 0;
+	if (data.size >= 10 && !memcmp(data.data, "ID3", 3)) {
+		/* skip the ID3 tag */
+
+		tagsize = (data.data[6] << 21) | (data.data[7] << 14) |
+		    (data.data[8] << 7) | (data.data[9] << 0);
+
+		tagsize += 10;
+
+		if (!decoder_buffer_skip(buffer, tagsize))
+			return -1;
+
+		data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 5));
+		if (data.IsNull())
+			return -1;
+	}
+
+	if (data.size >= 8 && adts_check_frame(data.data) > 0) {
+		/* obtain the duration from the ADTS header */
+
+		if (!is.IsSeekable())
+			return -1;
+
+		float song_length = adts_song_duration(buffer);
+
+		is.LockSeek(tagsize, IgnoreError());
+
+		decoder_buffer_clear(buffer);
+
+		return song_length;
+	} else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
+		/* obtain the duration from the ADIF header */
+
+		if (!is.KnownSize())
+			return -1;
+
+		size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
+
+		if (8 + skip_size > data.size)
+			/* not enough data yet; skip parsing this
+			   header */
+			return -1;
+
+		unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
+			(data.data[5 + skip_size] << 11) |
+			(data.data[6 + skip_size] << 3) |
+			(data.data[7 + skip_size] & 0xE0);
+
+		const auto size = is.GetSize();
+		if (bit_rate != 0)
+			return size * 8.0 / bit_rate;
+		else
+			return size;
+	} else
+		return -1;
+}
+
+static NeAACDecHandle
+faad_decoder_new()
+{
+	const NeAACDecHandle decoder = NeAACDecOpen();
+
+	NeAACDecConfigurationPtr config =
+		NeAACDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	config->downMatrix = 1;
+	config->dontUpSampleImplicitSBR = 0;
+	NeAACDecSetConfiguration(decoder, config);
+
+	return decoder;
+}
+
+/**
+ * Wrapper for NeAACDecInit() which works around some API
+ * inconsistencies in libfaad.
+ */
+static bool
+faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
+		  AudioFormat &audio_format, Error &error)
+{
+	auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+	if (data.IsEmpty()) {
+		error.Set(faad_decoder_domain, "Empty file");
+		return false;
+	}
+
+	uint8_t channels;
+	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_p = (unsigned long *)(void *)&sample_rate;
+#else
+	uint32_t *sample_rate_p = &sample_rate;
+#endif
+	long nbytes = NeAACDecInit(decoder,
+				   /* deconst hack, libfaad requires this */
+				   const_cast<unsigned char *>(data.data),
+				   data.size,
+				   sample_rate_p, &channels);
+	if (nbytes < 0) {
+		error.Set(faad_decoder_domain, "Not an AAC stream");
+		return false;
+	}
+
+	decoder_buffer_consume(buffer, nbytes);
+
+	return audio_format_init_checked(audio_format, sample_rate,
+					 SampleFormat::S16, channels, error);
+}
+
+/**
+ * Wrapper for NeAACDecDecode() which works around some API
+ * inconsistencies in libfaad.
+ */
+static const void *
+faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer,
+		    NeAACDecFrameInfo *frame_info)
+{
+	auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+	if (data.IsEmpty())
+		return nullptr;
+
+	return NeAACDecDecode(decoder, frame_info,
+			      /* deconst hack, libfaad requires this */
+			      const_cast<uint8_t *>(data.data),
+			      data.size);
+}
+
+/**
+ * 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(InputStream &is)
+{
+	DecoderBuffer *buffer =
+		decoder_buffer_new(nullptr, is,
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
+	float length = faad_song_duration(buffer, is);
+
+	if (length < 0) {
+		NeAACDecHandle decoder = faad_decoder_new();
+
+		decoder_buffer_fill(buffer);
+
+		AudioFormat audio_format;
+		if (faad_decoder_init(decoder, buffer, audio_format,
+				      IgnoreError()))
+			length = 0;
+
+		NeAACDecClose(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(InputStream &is)
+{
+	float length = faad_get_file_time_float(is);
+	if (length < 0)
+		return -1;
+
+	return int(length + 0.5);
+}
+
+static void
+faad_stream_decode(Decoder &mpd_decoder, InputStream &is,
+		   DecoderBuffer *buffer, const NeAACDecHandle decoder)
+{
+	const float total_time = faad_song_duration(buffer, is);
+
+	if (adts_find_frame(buffer) == 0)
+		return;
+
+	/* initialize it */
+
+	Error error;
+	AudioFormat audio_format;
+	if (!faad_decoder_init(decoder, buffer, audio_format, error)) {
+		LogError(error);
+		return;
+	}
+
+	/* initialize the MPD core */
+
+	decoder_initialized(mpd_decoder, audio_format, false, total_time);
+
+	/* the decoder loop */
+
+	DecoderCommand cmd;
+	unsigned bit_rate = 0;
+	do {
+		/* find the next frame */
+
+		const size_t frame_size = adts_find_frame(buffer);
+		if (frame_size == 0)
+			/* end of file */
+			break;
+
+		/* decode it */
+
+		NeAACDecFrameInfo frame_info;
+		const void *const decoded =
+			faad_decoder_decode(decoder, buffer, &frame_info);
+
+		if (frame_info.error > 0) {
+			FormatWarning(faad_decoder_domain,
+				      "error decoding AAC stream: %s",
+				      NeAACDecGetErrorMessage(frame_info.error));
+			break;
+		}
+
+		if (frame_info.channels != audio_format.channels) {
+			FormatDefault(faad_decoder_domain,
+				      "channel count changed from %u to %u",
+				      audio_format.channels, frame_info.channels);
+			break;
+		}
+
+		if (frame_info.samplerate != audio_format.sample_rate) {
+			FormatDefault(faad_decoder_domain,
+				      "sample rate changed from %u to %lu",
+				      audio_format.sample_rate,
+				      (unsigned long)frame_info.samplerate);
+			break;
+		}
+
+		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 != DecoderCommand::STOP);
+}
+
+static void
+faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
+{
+	DecoderBuffer *buffer =
+		decoder_buffer_new(&mpd_decoder, is,
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
+
+	/* create the libfaad decoder */
+
+	const NeAACDecHandle decoder = faad_decoder_new();
+
+	faad_stream_decode(mpd_decoder, is, buffer, decoder);
+
+	/* cleanup */
+
+	NeAACDecClose(decoder);
+	decoder_buffer_free(buffer);
+}
+
+static bool
+faad_scan_stream(InputStream &is,
+		 const struct tag_handler *handler, void *handler_ctx)
+{
+	int file_time = faad_get_file_time(is);
+	if (file_time < 0)
+		return false;
+
+	tag_handler_invoke_duration(handler, handler_ctx, file_time);
+	return true;
+}
+
+static const char *const faad_suffixes[] = { "aac", nullptr };
+static const char *const faad_mime_types[] = {
+	"audio/aac", "audio/aacp", nullptr
+};
+
+const DecoderPlugin faad_decoder_plugin = {
+	"faad",
+	nullptr,
+	nullptr,
+	faad_stream_decode,
+	nullptr,
+	nullptr,
+	faad_scan_stream,
+	nullptr,
+	faad_suffixes,
+	faad_mime_types,
+};