1 files changed, 552 insertions, 0 deletions

diff --git a/trunk/formats/format_ogg_vorbis.c b/trunk/formats/format_ogg_vorbis.c
new file mode 100644
index 000000000..669e96a7d
--- /dev/null
+++ b/trunk/formats/format_ogg_vorbis.c
@@ -0,0 +1,552 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Jeff Ollie
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief OGG/Vorbis streams.
+ * \arg File name extension: ogg
+ * \ingroup formats
+ */
+
+/* the order of these dependencies is important... it also specifies
+   the link order of the libraries during linking
+*/
+
+/*** MODULEINFO
+	<depend>vorbis</depend>
+	<depend>ogg</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <vorbis/codec.h>
+#include <vorbis/vorbisenc.h>
+
+#ifdef _WIN32
+#include <io.h>
+#endif
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+
+/*
+ * this is the number of samples we deal with. Samples are converted
+ * to SLINEAR so each one uses 2 bytes in the buffer.
+ */
+#define SAMPLES_MAX 160
+#define	BUF_SIZE	(2*SAMPLES_MAX)
+
+#define BLOCK_SIZE 4096		/* used internally in the vorbis routines */
+
+struct vorbis_desc {	/* format specific parameters */
+	/* structures for handling the Ogg container */
+	ogg_sync_state oy;
+	ogg_stream_state os;
+	ogg_page og;
+	ogg_packet op;
+	
+	/* structures for handling Vorbis audio data */
+	vorbis_info vi;
+	vorbis_comment vc;
+	vorbis_dsp_state vd;
+	vorbis_block vb;
+	
+	/*! \brief Indicates whether this filestream is set up for reading or writing. */
+	int writing;
+	
+	/*! \brief Indicates whether an End of Stream condition has been detected. */
+	int eos;
+};
+
+/*!
+ * \brief Create a new OGG/Vorbis filestream and set it up for reading.
+ * \param s File that points to on disk storage of the OGG/Vorbis data.
+ * \return The new filestream.
+ */
+static int ogg_vorbis_open(struct ast_filestream *s)
+{
+	int i;
+	int bytes;
+	int result;
+	char **ptr;
+	char *buffer;
+	struct vorbis_desc *tmp = (struct vorbis_desc *)s->_private;
+
+	tmp->writing = 0;
+
+	ogg_sync_init(&tmp->oy);
+
+	buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
+	bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
+	ogg_sync_wrote(&tmp->oy, bytes);
+
+	result = ogg_sync_pageout(&tmp->oy, &tmp->og);
+	if (result != 1) {
+		if(bytes < BLOCK_SIZE) {
+			ast_log(LOG_ERROR, "Run out of data...\n");
+		} else {
+			ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
+		}
+		ogg_sync_clear(&tmp->oy);
+		return -1;
+	}
+	
+	ogg_stream_init(&tmp->os, ogg_page_serialno(&tmp->og));
+	vorbis_info_init(&tmp->vi);
+	vorbis_comment_init(&tmp->vc);
+
+	if (ogg_stream_pagein(&tmp->os, &tmp->og) < 0) { 
+		ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
+error:
+		ogg_stream_clear(&tmp->os);
+		vorbis_comment_clear(&tmp->vc);
+		vorbis_info_clear(&tmp->vi);
+		ogg_sync_clear(&tmp->oy);
+		return -1;
+	}
+	
+	if (ogg_stream_packetout(&tmp->os, &tmp->op) != 1) { 
+		ast_log(LOG_ERROR, "Error reading initial header packet.\n");
+		goto error;
+	}
+	
+	if (vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) { 
+		ast_log(LOG_ERROR, "This Ogg bitstream does not contain Vorbis audio data.\n");
+		goto error;
+	}
+	
+	for (i = 0; i < 2 ; ) {
+		while (i < 2) {
+			result = ogg_sync_pageout(&tmp->oy, &tmp->og);
+			if (result == 0)
+				break;
+			if (result == 1) {
+				ogg_stream_pagein(&tmp->os, &tmp->og);
+				while(i < 2) {
+					result = ogg_stream_packetout(&tmp->os,&tmp->op);
+					if(result == 0)
+						break;
+					if(result < 0) {
+						ast_log(LOG_ERROR, "Corrupt secondary header.  Exiting.\n");
+						goto error;
+					}
+					vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
+					i++;
+				}
+			}
+		}
+
+		buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
+		bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
+		if (bytes == 0 && i < 2) {
+			ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
+			goto error;
+		}
+		ogg_sync_wrote(&tmp->oy, bytes);
+	}
+	
+	for (ptr = tmp->vc.user_comments; *ptr; ptr++)
+		ast_debug(1, "OGG/Vorbis comment: %s\n", *ptr);
+		ast_debug(1, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
+		ast_debug(1, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor);
+
+	if (tmp->vi.channels != 1) {
+		ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
+		goto error;
+	}
+	
+	if (tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
+		ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
+		vorbis_block_clear(&tmp->vb);
+		vorbis_dsp_clear(&tmp->vd);
+		goto error;
+	}
+	
+	vorbis_synthesis_init(&tmp->vd, &tmp->vi);
+	vorbis_block_init(&tmp->vd, &tmp->vb);
+
+	return 0;
+}
+
+/*!
+ * \brief Create a new OGG/Vorbis filestream and set it up for writing.
+ * \param s File pointer that points to on-disk storage.
+ * \param comment Comment that should be embedded in the OGG/Vorbis file.
+ * \return A new filestream.
+ */
+static int ogg_vorbis_rewrite(struct ast_filestream *s,
+						 const char *comment)
+{
+	ogg_packet header;
+	ogg_packet header_comm;
+	ogg_packet header_code;
+	struct vorbis_desc *tmp = (struct vorbis_desc *)s->_private;
+
+	tmp->writing = 1;
+
+	vorbis_info_init(&tmp->vi);
+
+	if (vorbis_encode_init_vbr(&tmp->vi, 1, DEFAULT_SAMPLE_RATE, 0.4)) {
+		ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
+		return -1;
+	}
+
+	vorbis_comment_init(&tmp->vc);
+	vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
+	if (comment)
+		vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
+
+	vorbis_analysis_init(&tmp->vd, &tmp->vi);
+	vorbis_block_init(&tmp->vd, &tmp->vb);
+
+	ogg_stream_init(&tmp->os, ast_random());
+
+	vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm,
+				  &header_code);
+	ogg_stream_packetin(&tmp->os, &header);
+	ogg_stream_packetin(&tmp->os, &header_comm);
+	ogg_stream_packetin(&tmp->os, &header_code);
+
+	while (!tmp->eos) {
+		if (ogg_stream_flush(&tmp->os, &tmp->og) == 0)
+			break;
+		fwrite(tmp->og.header, 1, tmp->og.header_len, s->f);
+		fwrite(tmp->og.body, 1, tmp->og.body_len, s->f);
+		if (ogg_page_eos(&tmp->og))
+			tmp->eos = 1;
+	}
+
+	return 0;
+}
+
+/*!
+ * \brief Write out any pending encoded data.
+ * \param s An OGG/Vorbis filestream.
+ * \param f The file to write to.
+ */
+static void write_stream(struct vorbis_desc *s, FILE *f)
+{
+	while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
+		vorbis_analysis(&s->vb, NULL);
+		vorbis_bitrate_addblock(&s->vb);
+
+		while (vorbis_bitrate_flushpacket(&s->vd, &s->op)) {
+			ogg_stream_packetin(&s->os, &s->op);
+			while (!s->eos) {
+				if (ogg_stream_pageout(&s->os, &s->og) == 0) {
+					break;
+				}
+				fwrite(s->og.header, 1, s->og.header_len, f);
+				fwrite(s->og.body, 1, s->og.body_len, f);
+				if (ogg_page_eos(&s->og)) {
+					s->eos = 1;
+				}
+			}
+		}
+	}
+}
+
+/*!
+ * \brief Write audio data from a frame to an OGG/Vorbis filestream.
+ * \param fs An OGG/Vorbis filestream.
+ * \param f A frame containing audio to be written to the filestream.
+ * \return -1 if there was an error, 0 on success.
+ */
+static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int i;
+	float **buffer;
+	short *data;
+	struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
+
+	if (!s->writing) {
+		ast_log(LOG_ERROR, "This stream is not set up for writing!\n");
+		return -1;
+	}
+
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_SLINEAR) {
+		ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n",
+				f->subclass);
+		return -1;
+	}
+	if (!f->datalen)
+		return -1;
+
+	data = (short *) f->data;
+
+	buffer = vorbis_analysis_buffer(&s->vd, f->samples);
+
+	for (i = 0; i < f->samples; i++)
+		buffer[0][i] = (double)data[i] / 32768.0;
+
+	vorbis_analysis_wrote(&s->vd, f->samples);
+
+	write_stream(s, fs->f);
+
+	return 0;
+}
+
+/*!
+ * \brief Close a OGG/Vorbis filestream.
+ * \param fs A OGG/Vorbis filestream.
+ */
+static void ogg_vorbis_close(struct ast_filestream *fs)
+{
+	struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
+
+	if (s->writing) {
+		/* Tell the Vorbis encoder that the stream is finished
+		 * and write out the rest of the data */
+		vorbis_analysis_wrote(&s->vd, 0);
+		write_stream(s, fs->f);
+	}
+
+	ogg_stream_clear(&s->os);
+	vorbis_block_clear(&s->vb);
+	vorbis_dsp_clear(&s->vd);
+	vorbis_comment_clear(&s->vc);
+	vorbis_info_clear(&s->vi);
+
+	if (s->writing) {
+		ogg_sync_clear(&s->oy);
+	}
+}
+
+/*!
+ * \brief Get audio data.
+ * \param fs An OGG/Vorbis filestream.
+ * \param pcm Pointer to a buffere to store audio data in.
+ */
+
+static int read_samples(struct ast_filestream *fs, float ***pcm)
+{
+	int samples_in;
+	int result;
+	char *buffer;
+	int bytes;
+	struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
+
+	while (1) {
+		samples_in = vorbis_synthesis_pcmout(&s->vd, pcm);
+		if (samples_in > 0) {
+			return samples_in;
+		}
+
+		/* The Vorbis decoder needs more data... */
+		/* See ifOGG has any packets in the current page for the Vorbis decoder. */
+		result = ogg_stream_packetout(&s->os, &s->op);
+		if (result > 0) {
+			/* Yes OGG had some more packets for the Vorbis decoder. */
+			if (vorbis_synthesis(&s->vb, &s->op) == 0) {
+				vorbis_synthesis_blockin(&s->vd, &s->vb);
+			}
+
+			continue;
+		}
+
+		if (result < 0)
+			ast_log(LOG_WARNING,
+					"Corrupt or missing data at this page position; continuing...\n");
+
+		/* No more packets left in the current page... */
+
+		if (s->eos) {
+			/* No more pages left in the stream */
+			return -1;
+		}
+
+		while (!s->eos) {
+			/* See ifOGG has any pages in it's internal buffers */
+			result = ogg_sync_pageout(&s->oy, &s->og);
+			if (result > 0) {
+				/* Yes, OGG has more pages in it's internal buffers,
+				   add the page to the stream state */
+				result = ogg_stream_pagein(&s->os, &s->og);
+				if (result == 0) {
+					/* Yes, got a new,valid page */
+					if (ogg_page_eos(&s->og)) {
+						s->eos = 1;
+					}
+					break;
+				}
+				ast_log(LOG_WARNING,
+						"Invalid page in the bitstream; continuing...\n");
+			}
+
+			if (result < 0)
+				ast_log(LOG_WARNING,
+						"Corrupt or missing data in bitstream; continuing...\n");
+
+			/* No, we need to read more data from the file descrptor */
+			/* get a buffer from OGG to read the data into */
+			buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
+			/* read more data from the file descriptor */
+			bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
+			/* Tell OGG how many bytes we actually read into the buffer */
+			ogg_sync_wrote(&s->oy, bytes);
+			if (bytes == 0) {
+				s->eos = 1;
+			}
+		}
+	}
+}
+
+/*!
+ * \brief Read a frame full of audio data from the filestream.
+ * \param fs The filestream.
+ * \param whennext Number of sample times to schedule the next call.
+ * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
+ */
+static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
+					 int *whennext)
+{
+	int clipflag = 0;
+	int i;
+	int j;
+	double accumulator[SAMPLES_MAX];
+	int val;
+	int samples_in;
+	int samples_out = 0;
+	struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
+	short *buf;	/* SLIN data buffer */
+
+	fs->fr.frametype = AST_FRAME_VOICE;
+	fs->fr.subclass = AST_FORMAT_SLINEAR;
+	fs->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	buf = (short *)(fs->fr.data);	/* SLIN data buffer */
+
+	while (samples_out != SAMPLES_MAX) {
+		float **pcm;
+		int len = SAMPLES_MAX - samples_out;
+
+		/* See ifVorbis decoder has some audio data for us ... */
+		samples_in = read_samples(fs, &pcm);
+		if (samples_in <= 0)
+			break;
+
+		/* Got some audio data from Vorbis... */
+		/* Convert the float audio data to 16-bit signed linear */
+
+		clipflag = 0;
+		if (samples_in > len)
+			samples_in = len;
+		for (j = 0; j < samples_in; j++)
+			accumulator[j] = 0.0;
+
+		for (i = 0; i < s->vi.channels; i++) {
+			float *mono = pcm[i];
+			for (j = 0; j < samples_in; j++)
+				accumulator[j] += mono[j];
+		}
+
+		for (j = 0; j < samples_in; j++) {
+			val = accumulator[j] * 32767.0 / s->vi.channels;
+			if (val > 32767) {
+				val = 32767;
+				clipflag = 1;
+			} else if (val < -32768) {
+				val = -32768;
+				clipflag = 1;
+			}
+			buf[samples_out + j] = val;
+		}
+
+		if (clipflag)
+			ast_log(LOG_WARNING, "Clipping in frame %ld\n", (long) (s->vd.sequence));
+		/* Tell the Vorbis decoder how many samples we actually used. */
+		vorbis_synthesis_read(&s->vd, samples_in);
+		samples_out += samples_in;
+	}
+
+	if (samples_out > 0) {
+		fs->fr.datalen = samples_out * 2;
+		fs->fr.samples = samples_out;
+		*whennext = samples_out;
+
+		return &fs->fr;
+	} else {
+		return NULL;
+	}
+}
+
+/*!
+ * \brief Trucate an OGG/Vorbis filestream.
+ * \param s The filestream to truncate.
+ * \return 0 on success, -1 on failure.
+ */
+
+static int ogg_vorbis_trunc(struct ast_filestream *s)
+{
+	ast_log(LOG_WARNING, "Truncation is not supported on OGG/Vorbis streams!\n");
+	return -1;
+}
+
+/*!
+ * \brief Seek to a specific position in an OGG/Vorbis filestream.
+ * \param s The filestream to truncate.
+ * \param sample_offset New position for the filestream, measured in 8KHz samples.
+ * \param whence Location to measure 
+ * \return 0 on success, -1 on failure.
+ */
+static int ogg_vorbis_seek(struct ast_filestream *s, off_t sample_offset, int whence)
+{
+	ast_log(LOG_WARNING, "Seeking is not supported on OGG/Vorbis streams!\n");
+	return -1;
+}
+
+static off_t ogg_vorbis_tell(struct ast_filestream *s)
+{
+	ast_log(LOG_WARNING, "Telling is not supported on OGG/Vorbis streams!\n");
+	return -1;
+}
+
+static const struct ast_format vorbis_f = {
+	.name = "ogg_vorbis",
+	.exts = "ogg",
+	.format = AST_FORMAT_SLINEAR,
+	.open = ogg_vorbis_open,
+	.rewrite = ogg_vorbis_rewrite,
+	.write = ogg_vorbis_write,
+	.seek =	ogg_vorbis_seek,
+	.trunc = ogg_vorbis_trunc,
+	.tell = ogg_vorbis_tell,
+	.read = ogg_vorbis_read,
+	.close = ogg_vorbis_close,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+	.desc_size = sizeof(struct vorbis_desc),
+};
+
+static int load_module(void)
+{
+	if (ast_format_register(&vorbis_f))
+		return AST_MODULE_LOAD_FAILURE;
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_unregister(vorbis_f.name);
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OGG/Vorbis audio");
+