--- /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 + speex + ogg + extended + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include "asterisk/mod_format.h" +#include "asterisk/module.h" + +#include +#include + +#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 +);