renamed src/inputPlugins/ to src/decoder/

These plugins are not input plugins, they are decoder plugins.  No
CamelCase in the directory name.

1 files changed, 602 insertions, 0 deletions

diff --git a/src/decoder/aac_plugin.c b/src/decoder/aac_plugin.c
new file mode 100644
index 000000000..7842bcc22
--- /dev/null
+++ b/src/decoder/aac_plugin.c
@@ -0,0 +1,602 @@
+/* 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"
+
+#define AAC_MAX_CHANNELS	6
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+#include <faad.h>
+
+/* all code here is either based on or copied from FAAD2's frontend code */
+typedef struct {
+	struct decoder *decoder;
+	InputStream *inStream;
+	size_t bytesIntoBuffer;
+	size_t bytesConsumed;
+	off_t fileOffset;
+	unsigned char *buffer;
+	int atEof;
+} 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 bread;
+
+	if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS)
+		/* buffer already full */
+		return;
+
+	aac_buffer_shift(b, b->bytesConsumed);
+
+	if (!b->atEof) {
+		size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS -
+			b->bytesIntoBuffer;
+
+		bread = decoder_read(b->decoder, b->inStream,
+				     (void *)(b->buffer + b->bytesIntoBuffer),
+				     rest);
+		if (bread == 0 && inputStreamAtEOF(b->inStream))
+			b->atEof = 1;
+		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 int 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;
+	int 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, InputStream * inStream)
+{
+	memset(b, 0, sizeof(AacBuffer));
+
+	b->decoder = decoder;
+	b->inStream = inStream;
+
+	b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+	memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+}
+
+static void aac_parse_header(AacBuffer * b, float *length)
+{
+	size_t fileread;
+	size_t tagsize;
+
+	if (length)
+		*length = -1;
+
+	fileread = b->inStream->size;
+
+	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->bytesIntoBuffer >= 2 &&
+	    (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
+		adtsParse(b, length);
+		seekInputStream(b->inStream, tagsize, SEEK_SET);
+
+		b->bytesIntoBuffer = 0;
+		b->bytesConsumed = 0;
+		b->fileOffset = tagsize;
+
+		fillAacBuffer(b);
+	} else if (memcmp(b->buffer, "ADIF", 4) == 0) {
+		int bitRate;
+		int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
+		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(char *file)
+{
+	AacBuffer b;
+	float length;
+	faacDecHandle decoder;
+	faacDecConfigurationPtr config;
+	uint32_t sample_rate;
+	unsigned char channels;
+	InputStream inStream;
+	long bread;
+
+	if (openInputStream(&inStream, file) < 0)
+		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, &channels);
+#else
+		bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+		if (bread >= 0 && sample_rate > 0 && channels > 0)
+			length = 0;
+
+		faacDecClose(decoder);
+	}
+
+	if (b.buffer)
+		free(b.buffer);
+	closeInputStream(&inStream);
+
+	return length;
+}
+
+static int getAacTotalTime(char *file)
+{
+	int file_time = -1;
+	float length;
+
+	if ((length = getAacFloatTotalTime(file)) >= 0)
+		file_time = length + 0.5;
+
+	return file_time;
+}
+
+static int aac_stream_decode(struct decoder * mpd_decoder,
+			     InputStream *inStream)
+{
+	float file_time;
+	float totalTime = 0;
+	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;
+	uint16_t bitRate = 0;
+	AacBuffer b;
+	int initialized = 0;
+
+	initAacBuffer(&b, mpd_decoder, inStream);
+
+	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 < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS &&
+	       !b.atEof &&
+	       decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+	       	fillAacBuffer(&b);
+		adts_find_frame(&b);
+		fillAacBuffer(&b);
+		my_usleep(10000);
+	}
+
+#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);
+		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) {
+			ERROR("error decoding AAC stream\n");
+			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 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);
+	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
+};