Add support for OGG/Speex file format
authorTimo Teräs <timo.teras@iki.fi>
Fri, 3 Jun 2016 06:20:39 +0000 (09:20 +0300)
committerTimo Teräs <timo.teras@iki.fi>
Thu, 9 Jun 2016 19:01:42 +0000 (22:01 +0300)
ASTERISK-18995 #close

Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a

CHANGES
formats/format_ogg_speex.c [new file with mode: 0644]

diff --git a/CHANGES b/CHANGES
index 43dc18f..175138a 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -249,6 +249,13 @@ Functions
  * The func_odbc global option "single_db_connection" default value has been
    changed to 'no'.
 
+
+Formats
+------------------
+ * New module format_ogg_speex added which supports Speex codec inside
+   Ogg containers (filename extension .spx).
+
+
 CHANNEL
 ------------------
  * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
diff --git a/formats/format_ogg_speex.c b/formats/format_ogg_speex.c
new file mode 100644 (file)
index 0000000..6152e9c
--- /dev/null
@@ -0,0 +1,345 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011-2016, 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_REGISTER_FILE()
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+#include "asterisk/format_cache.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 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.
+ * \return The new filestream.
+ */
+static int ogg_speex_open(struct ast_filestream *fs)
+{
+       char *buffer;
+       size_t bytes;
+       struct speex_desc *s = (struct speex_desc *)fs->_private;
+       SpeexHeader *hdr = NULL;
+       int i, result, expected_rate;
+
+       expected_rate = ast_format_get_sample_rate(fs->fmt->format);
+       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;
+               }
+       }
+       speex_header_free(hdr);
+
+       return 0;
+error:
+       if (hdr) {
+               speex_header_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;
+       }
+
+       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_samples_count(&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 struct ast_format_def speex_f = {
+       .name = "ogg_speex",
+       .exts = "spx",
+       .open = ogg_speex_open,
+       .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 struct ast_format_def speex16_f = {
+       .name = "ogg_speex16",
+       .exts = "spx16",
+       .open = ogg_speex_open,
+       .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 struct ast_format_def speex32_f = {
+       .name = "ogg_speex32",
+       .exts = "spx32",
+       .open = ogg_speex_open,
+       .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)
+{
+       speex_f.format = ast_format_speex;
+       speex16_f.format = ast_format_speex16;
+       speex32_f.format = ast_format_speex32;
+
+       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
+);