[Asterisk-code-review] Add support for OGG/Speex file format (asterisk[master])
Timo Teräs
asteriskteam at digium.com
Fri Jun 3 01:28:56 CDT 2016
Timo Teräs has uploaded a new change for review.
https://gerrit.asterisk.org/2939
Change subject: Add support for OGG/Speex file format
......................................................................
Add support for OGG/Speex file format
ASTERISK-18995 #close
Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a
---
A formats/format_ogg_speex.c
1 file changed, 336 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/39/2939/1
diff --git a/formats/format_ogg_speex.c b/formats/format_ogg_speex.c
new file mode 100644
index 0000000..7a18414
--- /dev/null
+++ b/formats/format_ogg_speex.c
@@ -0,0 +1,336 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011-2014, 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;
+ }
+ }
+ 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;
+
+ 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
+);
--
To view, visit https://gerrit.asterisk.org/2939
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Timo Teräs <timo.teras at iki.fi>
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