path: root/testing/asterisk/ASTERISK-18995.patch

--- /dev/null	2011-11-29 09:02:40.279581283 +0200
+++ b/formats/format_ogg_speex.c	2011-12-08 15:57:12.000000000 +0200
@@ -0,0 +1,355 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011, Timo Teräs
+ *
+ * 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/Speex streams.
+ * \arg File name extension: spx
+ * \ingroup formats
+ */
+
+/*** MODULEINFO
+	<depend>speex</depend>
+	<depend>ogg</depend>
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+
+#include <speex/speex_header.h>
+#include <ogg/ogg.h>
+
+#define BLOCK_SIZE	4096		/* buffer size for feeding OGG routines */
+#define	BUF_SIZE	200
+
+struct speex_desc {	/* format specific parameters */
+	/* structures for handling the Ogg container */
+	ogg_sync_state oy;
+	ogg_stream_state os;
+	ogg_page og;
+	ogg_packet op;
+
+	int format_id;
+	int serialno;
+
+	/*! \brief Indicates whether an End of Stream condition has been detected. */
+	int eos;
+};
+
+static int read_packet(struct ast_filestream *fs)
+{
+	struct speex_desc *s = (struct speex_desc *)fs->_private;
+	char *buffer;
+	int result;
+	size_t bytes;
+
+	while (1) {
+		/* Get one packet */
+		result = ogg_stream_packetout(&s->os, &s->op);
+		if (result > 0) {
+			if (s->op.bytes>=5 && !memcmp(s->op.packet, "Speex", 5))
+				s->serialno = s->os.serialno;
+			if (s->serialno == -1 || s->os.serialno != s->serialno)
+				continue;
+			return 0;
+		}
+
+		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 if OGG has any pages in it's internal buffers */
+			result = ogg_sync_pageout(&s->oy, &s->og);
+			if (result > 0) {
+				/* Read all streams. */
+				if (ogg_page_serialno(&s->og) != s->os.serialno)
+					ogg_stream_reset_serialno(&s->os, ogg_page_serialno(&s->og));
+				/* 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) &&
+					    ogg_page_serialno(&s->og) == s->serialno)
+						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);
+			bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
+			ogg_sync_wrote(&s->oy, bytes);
+			if (bytes == 0)
+				s->eos = 1;
+		}
+	}
+}
+
+/*!
+ * \brief Create a new OGG/Speex filestream and set it up for reading.
+ * \param fs File that points to on disk storage of the OGG/Speex data.
+ * \param expected_rate The expected Speex format (sampling rate).
+ * \return The new filestream.
+ */
+static int ogg_speex_open(struct ast_filestream *fs, int format_id, int expected_rate)
+{
+	char *buffer;
+	size_t bytes;
+	struct speex_desc *s = (struct speex_desc *)fs->_private;
+	SpeexHeader *hdr = NULL;
+	int i, result;
+
+	s->format_id = format_id;
+	s->serialno = -1;
+	ogg_sync_init(&s->oy);
+
+	buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
+	bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
+	ogg_sync_wrote(&s->oy, bytes);
+
+	result = ogg_sync_pageout(&s->oy, &s->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(&s->oy);
+		return -1;
+	}
+
+	ogg_stream_init(&s->os, ogg_page_serialno(&s->og));
+	if (ogg_stream_pagein(&s->os, &s->og) < 0) { 
+		ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
+		goto error;
+	}
+
+	if (read_packet(fs) < 0) {
+		ast_log(LOG_ERROR, "Error reading initial header packet.\n");
+		goto error;
+	}
+
+	hdr = speex_packet_to_header((char*)s->op.packet, s->op.bytes);
+	if (memcmp(hdr->speex_string, "Speex   ", 8)) {
+		ast_log(LOG_ERROR, "OGG container does not contain Speex audio!\n");
+		goto error;
+	}
+	if (hdr->frames_per_packet != 1) {
+		ast_log(LOG_ERROR, "Only one frame-per-packet OGG/Speex files are currently supported!\n");
+		goto error;
+	}
+	if (hdr->nb_channels != 1) {
+		ast_log(LOG_ERROR, "Only monophonic OGG/Speex files are currently supported!\n");
+		goto error;
+	}
+	if (hdr->rate != expected_rate) {
+		ast_log(LOG_ERROR, "Unexpected sampling rate (%d != %d)!\n",
+			hdr->rate, expected_rate);
+		goto error;
+	}
+
+	/* this packet is the comment */
+	if (read_packet(fs) < 0) {
+		ast_log(LOG_ERROR, "Error reading comment packet.\n");
+		goto error;
+	}
+	for (i = 0; i < hdr->extra_headers; i++) {
+		if (read_packet(fs) < 0) {
+			ast_log(LOG_ERROR, "Error reading extra header packet %d.\n", i+1);
+			goto error;
+		}
+	}
+	free(hdr);
+
+	return 0;
+error:
+	if (hdr)
+		free(hdr);
+	ogg_stream_clear(&s->os);
+	ogg_sync_clear(&s->oy);
+	return -1;
+}
+
+/*!
+ * \brief Close a OGG/Speex filestream.
+ * \param fs A OGG/Speex filestream.
+ */
+static void ogg_speex_close(struct ast_filestream *fs)
+{
+	struct speex_desc *s = (struct speex_desc *)fs->_private;
+
+	ogg_stream_clear(&s->os);
+	ogg_sync_clear(&s->oy);
+}
+
+/*!
+ * \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_speex_read(struct ast_filestream *fs,
+					 int *whennext)
+{
+	struct speex_desc *s = (struct speex_desc *)fs->_private;
+
+	if (read_packet(fs) < 0)
+		return NULL;
+
+	fs->fr.frametype = AST_FRAME_VOICE;
+	ast_format_set(&fs->fr.subclass.format, s->format_id, 0);
+	fs->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
+	fs->fr.datalen = s->op.bytes;
+	fs->fr.samples = *whennext = ast_codec_get_samples(&fs->fr);
+
+	return &fs->fr;
+}
+
+/*!
+ * \brief Trucate an OGG/Speex filestream.
+ * \param s The filestream to truncate.
+ * \return 0 on success, -1 on failure.
+ */
+
+static int ogg_speex_trunc(struct ast_filestream *s)
+{
+	ast_log(LOG_WARNING, "Truncation is not supported on OGG/Speex streams!\n");
+	return -1;
+}
+
+/*!
+ * \brief Seek to a specific position in an OGG/Speex 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_speex_seek(struct ast_filestream *s, off_t sample_offset, int whence)
+{
+	ast_log(LOG_WARNING, "Seeking is not supported on OGG/Speex streams!\n");
+	return -1;
+}
+
+static off_t ogg_speex_tell(struct ast_filestream *s)
+{
+	ast_log(LOG_WARNING, "Telling is not supported on OGG/Speex streams!\n");
+	return -1;
+}
+
+static int ogg_speex_open_nb(struct ast_filestream *fs) 
+{
+	return ogg_speex_open(fs, AST_FORMAT_SPEEX, 8000);
+}
+
+static struct ast_format_def speex_f = {
+	.name = "ogg_speex",
+	.exts = "spx",
+	.open = ogg_speex_open_nb,
+	.seek = ogg_speex_seek,
+	.trunc = ogg_speex_trunc,
+	.tell = ogg_speex_tell,
+	.read = ogg_speex_read,
+	.close = ogg_speex_close,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+	.desc_size = sizeof(struct speex_desc),
+};
+
+static int ogg_speex_open_wb(struct ast_filestream *fs) 
+{
+	return ogg_speex_open(fs, AST_FORMAT_SPEEX16, 16000);
+}
+
+static struct ast_format_def speex16_f = {
+	.name = "ogg_speex16",
+	.exts = "spx16",
+	.open = ogg_speex_open_wb,
+	.seek = ogg_speex_seek,
+	.trunc = ogg_speex_trunc,
+	.tell = ogg_speex_tell,
+	.read = ogg_speex_read,
+	.close = ogg_speex_close,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+	.desc_size = sizeof(struct speex_desc),
+};
+
+static int ogg_speex_open_uwb(struct ast_filestream *fs) 
+{
+	return ogg_speex_open(fs, AST_FORMAT_SPEEX32, 32000);
+}
+
+static struct ast_format_def speex32_f = {
+	.name = "ogg_speex32",
+	.exts = "spx32",
+	.open = ogg_speex_open_uwb,
+	.seek = ogg_speex_seek,
+	.trunc = ogg_speex_trunc,
+	.tell = ogg_speex_tell,
+	.read = ogg_speex_read,
+	.close = ogg_speex_close,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+	.desc_size = sizeof(struct speex_desc),
+};
+
+static int load_module(void)
+{
+	ast_format_set(&speex_f.format, AST_FORMAT_SPEEX, 0);
+	ast_format_set(&speex16_f.format, AST_FORMAT_SPEEX16, 0);
+	ast_format_set(&speex32_f.format, AST_FORMAT_SPEEX32, 0);
+
+	if (ast_format_def_register(&speex_f) ||
+	    ast_format_def_register(&speex16_f) ||
+	    ast_format_def_register(&speex32_f))
+		return AST_MODULE_LOAD_FAILURE;
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	int res = 0;
+	res |= ast_format_def_unregister(speex_f.name);
+	res |= ast_format_def_unregister(speex16_f.name);
+	res |= ast_format_def_unregister(speex32_f.name);
+	return res;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio",
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_APP_DEPEND
+);