asterisk/formats/format_ogg_vorbis.c

/*
 * 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.
 */

/*
 *
 * OGG/Vorbis streams.
 * 
 */
 
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdlib.h>
#include <sys/time.h>
#include <stdio.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>

#include <vorbis/codec.h>
#include <vorbis/vorbisenc.h>

#ifdef _WIN32
#include <io.h>
#include <fcntl.h>
#endif

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"

#define SAMPLES_MAX 160
#define BLOCK_SIZE 4096


struct ast_filestream {
	void *reserved[AST_RESERVED_POINTERS];

	int fd;

	/* 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 Buffer to hold audio data. */
	short buffer[SAMPLES_MAX];

	/*! \brief Asterisk frame object. */
	struct ast_frame fr;
	char waste[AST_FRIENDLY_OFFSET];
	char empty;
};

AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock);
static int glistcnt = 0;

static char *name = "ogg_vorbis";
static char *desc = "OGG/Vorbis audio";
static char *exts = "ogg";

/*!
 * \brief Create a new OGG/Vorbis filestream and set it up for reading.
 * \param fd Descriptor that points to on disk storage of the OGG/Vorbis data.
 * \return The new filestream.
 */
static struct ast_filestream *ogg_vorbis_open(int fd)
{
	int i;
	int bytes;
	int result;
	char **ptr;
	char *buffer;

	struct ast_filestream *tmp;

	if((tmp = malloc(sizeof(struct ast_filestream)))) {
		memset(tmp, 0, sizeof(struct ast_filestream));

		tmp->writing = 0;
		tmp->fd = fd;

		ogg_sync_init(&tmp->oy);

		buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
		bytes = read(tmp->fd, buffer, BLOCK_SIZE);
		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");
			}
			close(fd);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		
		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");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		
		if(ogg_stream_packetout(&tmp->os, &tmp->op) != 1) { 
			ast_log(LOG_ERROR, "Error reading initial header packet.\n");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		
		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");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		
		i = 0;
		while(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");
							close(fd);
							ogg_stream_clear(&tmp->os);
							vorbis_comment_clear(&tmp->vc);
							vorbis_info_clear(&tmp->vi);
							ogg_sync_clear(&tmp->oy);
							free(tmp);
							return NULL;
						}
						vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
						i++;
					}
				}
			}

			buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
			bytes = read(tmp->fd, buffer, BLOCK_SIZE);
			if(bytes == 0 && i < 2) {
				ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
				close(fd);
				ogg_stream_clear(&tmp->os);
				vorbis_comment_clear(&tmp->vc);
				vorbis_info_clear(&tmp->vi);
				ogg_sync_clear(&tmp->oy);
				free(tmp);
				return NULL;
			}
			ogg_sync_wrote(&tmp->oy, bytes);
		}
		
		ptr = tmp->vc.user_comments;
		while(*ptr){
			ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *ptr);
			++ptr;
		}
		ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
		ast_log(LOG_DEBUG, "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");
			ogg_stream_clear(&tmp->os);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		

		if(tmp->vi.rate != 8000) {
			ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_block_clear(&tmp->vb);
			vorbis_dsp_clear(&tmp->vd);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		
		vorbis_synthesis_init(&tmp->vd, &tmp->vi);
		vorbis_block_init(&tmp->vd, &tmp->vb);

		if(ast_mutex_lock(&ogg_vorbis_lock)) {
			ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_block_clear(&tmp->vb);
			vorbis_dsp_clear(&tmp->vd);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			ogg_sync_clear(&tmp->oy);
			free(tmp);
			return NULL;
		}
		glistcnt++;
		ast_mutex_unlock(&ogg_vorbis_lock);
		ast_update_use_count();
	}
	return tmp;
}

/*!
 * \brief Create a new OGG/Vorbis filestream and set it up for writing.
 * \param fd File descriptor that points to on-disk storage.
 * \param comment Comment that should be embedded in the OGG/Vorbis file.
 * \return A new filestream.
 */
static struct ast_filestream *ogg_vorbis_rewrite(int fd, const char *comment)
{
	ogg_packet header;
	ogg_packet header_comm;
	ogg_packet header_code;

	struct ast_filestream *tmp;

	if((tmp = malloc(sizeof(struct ast_filestream)))) {
		memset(tmp, 0, sizeof(struct ast_filestream));

		tmp->writing = 1;
		tmp->fd = fd;

		vorbis_info_init(&tmp->vi);

		if(vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) {
			ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
			free(tmp);
			return NULL;
		}

		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, rand());

		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;
			write(tmp->fd, tmp->og.header, tmp->og.header_len);
			write(tmp->fd, tmp->og.body, tmp->og.body_len);
			if(ogg_page_eos(&tmp->og))
				tmp->eos = 1;
		}

		if(ast_mutex_lock(&ogg_vorbis_lock)) {
			ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
			close(fd);
			ogg_stream_clear(&tmp->os);
			vorbis_block_clear(&tmp->vb);
			vorbis_dsp_clear(&tmp->vd);
			vorbis_comment_clear(&tmp->vc);
			vorbis_info_clear(&tmp->vi);
			free(tmp);
			return NULL;
		}
		glistcnt++;
		ast_mutex_unlock(&ogg_vorbis_lock);
		ast_update_use_count();
	}
	return tmp;
}

/*!
 * \brief Write out any pending encoded data.
 * \param s A OGG/Vorbis filestream.
 */
static void write_stream(struct ast_filestream *s)
{
	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;
				}
				write(s->fd, s->og.header, s->og.header_len);
				write(s->fd, s->og.body, s->og.body_len);
				if(ogg_page_eos(&s->og)) {
					s->eos = 1;
				}
			}
		}
	}
}

/*!
 * \brief Write audio data from a frame to an OGG/Vorbis filestream.
 * \param s A OGG/Vorbis filestream.
 * \param f An frame containing audio to be written to the filestream.
 * \return -1 ifthere was an error, 0 on success.
 */
static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
{
	int i;
	float **buffer;
	short *data;

	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] = data[i]/32768.f;
	}

	vorbis_analysis_wrote(&s->vd, f->samples);

	write_stream(s);

	return 0;
}

/*!
 * \brief Close a OGG/Vorbis filestream.
 * \param s A OGG/Vorbis filestream.
 */
static void ogg_vorbis_close(struct ast_filestream *s)
{
	if(ast_mutex_lock(&ogg_vorbis_lock)) {
		ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
		return;
	}
	glistcnt--;
	ast_mutex_unlock(&ogg_vorbis_lock);
	ast_update_use_count();

	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);
	}

	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);
	}
	
	close(s->fd);
	free(s);
}

/*!
 * \brief Get audio data.
 * \param s An OGG/Vorbis filestream.
 * \param pcm Pointer to a buffere to store audio data in.
 */

static int read_samples(struct ast_filestream *s, float ***pcm)
{
	int samples_in;
	int result;
	char *buffer;
	int bytes;

	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 = read(s->fd, buffer, BLOCK_SIZE);
			/* 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 s 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 *s, int *whennext)
{
	int clipflag = 0;
	int i;
	int j;
	float **pcm;
	float *mono;
	double accumulator[SAMPLES_MAX];
	int val;
	int samples_in;
	int samples_out = 0;

	while (1) {
		/* See ifwe have filled up an audio frame yet */
		if(samples_out == SAMPLES_MAX)
			break;

		/* See ifVorbis decoder has some audio data for us ... */
		samples_in = read_samples(s, &pcm);
		if(samples_in <= 0)
			break;

		/* Got some audio data from Vorbis... */
		/* Convert the float audio data to 16-bit signed linear */
		
		clipflag = 0;

		samples_in = samples_in < (SAMPLES_MAX - samples_out) ? samples_in : (SAMPLES_MAX - samples_out);
  
		for(j = 0; j < samples_in; j++)
			accumulator[j] = 0.0;

		for(i = 0; i < s->vi.channels; i++) {
			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;
			}
			if(val < -32768) {
				val = -32768;
				clipflag = 1;
			}
			s->buffer[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) {
		s->fr.frametype = AST_FRAME_VOICE;
		s->fr.subclass = AST_FORMAT_SLINEAR;
		s->fr.offset = AST_FRIENDLY_OFFSET;
		s->fr.datalen = samples_out * 2;
		s->fr.data = s->buffer;
		s->fr.src = name;
		s->fr.mallocd = 0;
		s->fr.samples = samples_out;
		*whennext = samples_out;
		
		return &s->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, long sample_offset, int whence) {
	ast_log(LOG_WARNING, "Seeking is not supported on OGG/Vorbis streams!\n");
	return -1;
}

static long ogg_vorbis_tell(struct ast_filestream *s) {
	ast_log(LOG_WARNING, "Telling is not supported on OGG/Vorbis streams!\n");
	return -1;
}

static char *ogg_vorbis_getcomment(struct ast_filestream *s) {
	ast_log(LOG_WARNING, "Getting comments is not supported on OGG/Vorbis streams!\n");
	return NULL;
}

int load_module()
{
	return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
				   ogg_vorbis_open,
				   ogg_vorbis_rewrite,
				   ogg_vorbis_write,
				   ogg_vorbis_seek,
				   ogg_vorbis_trunc,
				   ogg_vorbis_tell,
				   ogg_vorbis_read,
				   ogg_vorbis_close,
				   ogg_vorbis_getcomment);
}

int unload_module()
{
	return ast_format_unregister(name);
}	

int usecount()
{
	return glistcnt;
}

char *description()
{
	return desc;
}


char *key()
{
	return ASTERISK_GPL_KEY;
}

/*
Local Variables:
mode: C
c-file-style: "linux"
indent-tabs-mode: t
End:
*/