add OGG/Vorbis file format support (bug #4296)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6173 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 670 insertions, 0 deletions

diff --git a/formats/Makefile b/formats/Makefile
index cbf43f899..ea0efee80 100755
--- a/formats/Makefile
+++ b/formats/Makefile
@@ -21,6 +21,11 @@ FORMAT_LIBS+=format_jpeg.so
 #
 FORMAT_LIBS+=format_g723.so
 
+#
+# OGG/Vorbis format
+#
+FORMAT_LIBS+=$(shell if [ -f $(CROSS_COMPILE_TARGET)/usr/include/vorbis/codec.h ]; then echo "format_ogg_vorbis.so" ; fi)
+
 GSMLIB=../codecs/gsm/lib/libgsm.a
 
 CFLAGS+=-fPIC
@@ -40,6 +45,9 @@ endif
 format_mp3.so : format_mp3.o
 	$(CC) $(SOLINK) -o $@ $< -lm
 
+format_ogg_vorbis.so : format_ogg_vorbis.o
+	$(CC) $(SOLINK) -o $@ $< -logg -lvorbis -lvorbisenc -lm
+
 install: all
 	for x in $(FORMAT_LIBS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
 
diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c
new file mode 100755
index 000000000..1fc78afa6
--- /dev/null
+++ b/formats/format_ogg_vorbis.c
@@ -0,0 +1,662 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * OGG/Vorbis streams.
+ * 
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#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:
+*/