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@ -20,7 +20,7 @@
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* \arg File name extension: ogg
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* \ingroup formats
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*/
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#include <sys/types.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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@ -48,37 +48,35 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/file.h"
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#include "asterisk/logger.h"
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#include "asterisk/module.h"
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#define SAMPLES_MAX 160
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#define BLOCK_SIZE 4096
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struct ast_filestream {
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void *reserved[AST_RESERVED_POINTERS];
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FILE *f;
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/* structures for handling the Ogg container */
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ogg_sync_state oy;
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ogg_sync_state oy;
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ogg_stream_state os;
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ogg_page og;
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ogg_packet op;
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ogg_page og;
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ogg_packet op;
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/* structures for handling Vorbis audio data */
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vorbis_info vi;
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vorbis_comment vc;
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vorbis_info vi;
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vorbis_comment vc;
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vorbis_dsp_state vd;
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vorbis_block vb;
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vorbis_block vb;
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/*! \brief Indicates whether this filestream is set up for reading or writing. */
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int writing;
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/*! \brief Indicates whether an End of Stream condition has been detected. */
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int eos;
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/*! \brief Buffer to hold audio data. */
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short buffer[SAMPLES_MAX];
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/*! \brief Asterisk frame object. */
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struct ast_frame fr;
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char waste[AST_FRIENDLY_OFFSET];
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@ -86,6 +84,7 @@ struct ast_filestream {
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};
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AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock);
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static int glistcnt = 0;
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static char *name = "ogg_vorbis";
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@ -97,7 +96,7 @@ static char *exts = "ogg";
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* \param f File that points to on disk storage of the OGG/Vorbis data.
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* \return The new filestream.
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*/
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static struct ast_filestream *ogg_vorbis_open(FILE *f)
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static struct ast_filestream *ogg_vorbis_open(FILE * f)
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{
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int i;
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int bytes;
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@ -107,7 +106,7 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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struct ast_filestream *tmp;
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if((tmp = malloc(sizeof(struct ast_filestream)))) {
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if ((tmp = malloc(sizeof(struct ast_filestream)))) {
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memset(tmp, 0, sizeof(struct ast_filestream));
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tmp->writing = 0;
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@ -120,24 +119,26 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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ogg_sync_wrote(&tmp->oy, bytes);
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result = ogg_sync_pageout(&tmp->oy, &tmp->og);
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if(result != 1) {
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if(bytes < BLOCK_SIZE) {
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if (result != 1) {
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if (bytes < BLOCK_SIZE) {
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ast_log(LOG_ERROR, "Run out of data...\n");
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} else {
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ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
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ast_log(LOG_ERROR,
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"Input does not appear to be an Ogg bitstream.\n");
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}
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fclose(f);
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ogg_sync_clear(&tmp->oy);
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free(tmp);
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return NULL;
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}
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ogg_stream_init(&tmp->os, ogg_page_serialno(&tmp->og));
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vorbis_info_init(&tmp->vi);
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vorbis_comment_init(&tmp->vc);
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if(ogg_stream_pagein(&tmp->os, &tmp->og) < 0) {
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ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
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if (ogg_stream_pagein(&tmp->os, &tmp->og) < 0) {
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ast_log(LOG_ERROR,
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"Error reading first page of Ogg bitstream data.\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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vorbis_comment_clear(&tmp->vc);
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@ -146,8 +147,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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free(tmp);
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return NULL;
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}
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if(ogg_stream_packetout(&tmp->os, &tmp->op) != 1) {
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if (ogg_stream_packetout(&tmp->os, &tmp->op) != 1) {
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ast_log(LOG_ERROR, "Error reading initial header packet.\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -157,8 +158,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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free(tmp);
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return NULL;
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}
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if(vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) {
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if (vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) {
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ast_log(LOG_ERROR, "This Ogg bitstream does not contain Vorbis audio data.\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -168,20 +169,20 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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free(tmp);
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return NULL;
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}
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i = 0;
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while(i < 2) {
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while(i < 2){
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while (i < 2) {
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while (i < 2) {
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result = ogg_sync_pageout(&tmp->oy, &tmp->og);
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if(result == 0)
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if (result == 0)
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break;
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if(result == 1) {
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if (result == 1) {
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ogg_stream_pagein(&tmp->os, &tmp->og);
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while(i < 2) {
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result = ogg_stream_packetout(&tmp->os,&tmp->op);
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if(result == 0)
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while (i < 2) {
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result = ogg_stream_packetout(&tmp->os, &tmp->op);
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if (result == 0)
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break;
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if(result < 0) {
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if (result < 0) {
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ast_log(LOG_ERROR, "Corrupt secondary header. Exiting.\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -199,7 +200,7 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
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bytes = fread(buffer, 1, BLOCK_SIZE, f);
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if(bytes == 0 && i < 2) {
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if (bytes == 0 && i < 2) {
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ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -211,16 +212,18 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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}
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ogg_sync_wrote(&tmp->oy, bytes);
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}
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ptr = tmp->vc.user_comments;
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while(*ptr){
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while (*ptr) {
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ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *ptr);
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++ptr;
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}
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ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
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ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor);
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ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n",
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tmp->vi.channels, tmp->vi.rate);
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ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n",
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tmp->vc.vendor);
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if(tmp->vi.channels != 1) {
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if (tmp->vi.channels != 1) {
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ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
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ogg_stream_clear(&tmp->os);
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vorbis_comment_clear(&tmp->vc);
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@ -229,9 +232,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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free(tmp);
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return NULL;
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}
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if(tmp->vi.rate != 8000) {
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if (tmp->vi.rate != 8000) {
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ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -243,11 +245,11 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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free(tmp);
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return NULL;
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}
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vorbis_synthesis_init(&tmp->vd, &tmp->vi);
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vorbis_block_init(&tmp->vd, &tmp->vb);
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if(ast_mutex_lock(&ogg_vorbis_lock)) {
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if (ast_mutex_lock(&ogg_vorbis_lock)) {
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ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -272,7 +274,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f)
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* \param comment Comment that should be embedded in the OGG/Vorbis file.
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* \return A new filestream.
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*/
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static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
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static struct ast_filestream *ogg_vorbis_rewrite(FILE * f,
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const char *comment)
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{
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ogg_packet header;
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ogg_packet header_comm;
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@ -280,7 +283,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
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struct ast_filestream *tmp;
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if((tmp = malloc(sizeof(struct ast_filestream)))) {
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if ((tmp = malloc(sizeof(struct ast_filestream)))) {
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memset(tmp, 0, sizeof(struct ast_filestream));
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tmp->writing = 1;
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@ -288,7 +291,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
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vorbis_info_init(&tmp->vi);
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if(vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) {
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if (vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) {
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ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
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free(tmp);
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return NULL;
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@ -296,7 +299,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
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vorbis_comment_init(&tmp->vc);
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vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
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if(comment)
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if (comment)
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vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
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vorbis_analysis_init(&tmp->vd, &tmp->vi);
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@ -304,21 +307,22 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
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ogg_stream_init(&tmp->os, rand());
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vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm, &header_code);
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ogg_stream_packetin(&tmp->os, &header);
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vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm,
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&header_code);
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ogg_stream_packetin(&tmp->os, &header);
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ogg_stream_packetin(&tmp->os, &header_comm);
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ogg_stream_packetin(&tmp->os, &header_code);
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while(!tmp->eos) {
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if(ogg_stream_flush(&tmp->os, &tmp->og) == 0)
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while (!tmp->eos) {
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if (ogg_stream_flush(&tmp->os, &tmp->og) == 0)
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break;
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fwrite(tmp->og.header, 1, tmp->og.header_len, tmp->f);
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fwrite(tmp->og.body, 1, tmp->og.body_len, tmp->f);
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if(ogg_page_eos(&tmp->og))
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if (ogg_page_eos(&tmp->og))
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tmp->eos = 1;
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}
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if(ast_mutex_lock(&ogg_vorbis_lock)) {
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if (ast_mutex_lock(&ogg_vorbis_lock)) {
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ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
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fclose(f);
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ogg_stream_clear(&tmp->os);
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@ -345,16 +349,16 @@ static void write_stream(struct ast_filestream *s)
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while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
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vorbis_analysis(&s->vb, NULL);
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vorbis_bitrate_addblock(&s->vb);
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while (vorbis_bitrate_flushpacket(&s->vd, &s->op)) {
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ogg_stream_packetin(&s->os, &s->op);
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while (!s->eos) {
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if(ogg_stream_pageout(&s->os, &s->og) == 0) {
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if (ogg_stream_pageout(&s->os, &s->og) == 0) {
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break;
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}
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fwrite(s->og.header, 1, s->og.header_len, s->f);
|
|
|
|
|
fwrite(s->og.body, 1, s->og.body_len, s->f);
|
|
|
|
|
if(ogg_page_eos(&s->og)) {
|
|
|
|
|
if (ogg_page_eos(&s->og)) {
|
|
|
|
|
s->eos = 1;
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
@ -374,20 +378,21 @@ static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
|
|
|
|
|
float **buffer;
|
|
|
|
|
short *data;
|
|
|
|
|
|
|
|
|
|
if(!s->writing) {
|
|
|
|
|
if (!s->writing) {
|
|
|
|
|
ast_log(LOG_ERROR, "This stream is not set up for writing!\n");
|
|
|
|
|
return -1;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if(f->frametype != AST_FRAME_VOICE) {
|
|
|
|
|
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);
|
|
|
|
|
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)
|
|
|
|
|
if (!f->datalen)
|
|
|
|
|
return -1;
|
|
|
|
|
|
|
|
|
|
data = (short *) f->data;
|
|
|
|
@ -395,7 +400,7 @@ static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
|
|
|
|
|
buffer = vorbis_analysis_buffer(&s->vd, f->samples);
|
|
|
|
|
|
|
|
|
|
for (i = 0; i < f->samples; i++) {
|
|
|
|
|
buffer[0][i] = data[i]/32768.f;
|
|
|
|
|
buffer[0][i] = data[i] / 32768.f;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
vorbis_analysis_wrote(&s->vd, f->samples);
|
|
|
|
@ -411,7 +416,7 @@ static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
|
|
|
|
|
*/
|
|
|
|
|
static void ogg_vorbis_close(struct ast_filestream *s)
|
|
|
|
|
{
|
|
|
|
|
if(ast_mutex_lock(&ogg_vorbis_lock)) {
|
|
|
|
|
if (ast_mutex_lock(&ogg_vorbis_lock)) {
|
|
|
|
|
ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
|
|
|
|
|
return;
|
|
|
|
|
}
|
|
|
|
@ -419,7 +424,7 @@ static void ogg_vorbis_close(struct ast_filestream *s)
|
|
|
|
|
ast_mutex_unlock(&ogg_vorbis_lock);
|
|
|
|
|
ast_update_use_count();
|
|
|
|
|
|
|
|
|
|
if(s->writing) {
|
|
|
|
|
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);
|
|
|
|
@ -432,10 +437,10 @@ static void ogg_vorbis_close(struct ast_filestream *s)
|
|
|
|
|
vorbis_comment_clear(&s->vc);
|
|
|
|
|
vorbis_info_clear(&s->vi);
|
|
|
|
|
|
|
|
|
|
if(s->writing) {
|
|
|
|
|
if (s->writing) {
|
|
|
|
|
ogg_sync_clear(&s->oy);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
fclose(s->f);
|
|
|
|
|
free(s);
|
|
|
|
|
}
|
|
|
|
@ -455,28 +460,29 @@ static int read_samples(struct ast_filestream *s, float ***pcm)
|
|
|
|
|
|
|
|
|
|
while (1) {
|
|
|
|
|
samples_in = vorbis_synthesis_pcmout(&s->vd, pcm);
|
|
|
|
|
if(samples_in > 0) {
|
|
|
|
|
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) {
|
|
|
|
|
if (result > 0) {
|
|
|
|
|
/* Yes OGG had some more packets for the Vorbis decoder. */
|
|
|
|
|
if(vorbis_synthesis(&s->vb, &s->op) == 0) {
|
|
|
|
|
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");
|
|
|
|
|
|
|
|
|
|
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) {
|
|
|
|
|
if (s->eos) {
|
|
|
|
|
/* No more pages left in the stream */
|
|
|
|
|
return -1;
|
|
|
|
|
}
|
|
|
|
@ -484,22 +490,24 @@ static int read_samples(struct ast_filestream *s, float ***pcm)
|
|
|
|
|
while (!s->eos) {
|
|
|
|
|
/* See ifOGG has any pages in it's internal buffers */
|
|
|
|
|
result = ogg_sync_pageout(&s->oy, &s->og);
|
|
|
|
|
if(result > 0) {
|
|
|
|
|
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) {
|
|
|
|
|
if (result == 0) {
|
|
|
|
|
/* Yes, got a new,valid page */
|
|
|
|
|
if(ogg_page_eos(&s->og)) {
|
|
|
|
|
if (ogg_page_eos(&s->og)) {
|
|
|
|
|
s->eos = 1;
|
|
|
|
|
}
|
|
|
|
|
break;
|
|
|
|
|
}
|
|
|
|
|
ast_log(LOG_WARNING, "Invalid page in the bitstream; continuing...\n");
|
|
|
|
|
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");
|
|
|
|
|
|
|
|
|
|
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 */
|
|
|
|
@ -508,7 +516,7 @@ static int read_samples(struct ast_filestream *s, float ***pcm)
|
|
|
|
|
bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
|
|
|
|
|
/* Tell OGG how many bytes we actually read into the buffer */
|
|
|
|
|
ogg_sync_wrote(&s->oy, bytes);
|
|
|
|
|
if(bytes == 0) {
|
|
|
|
|
if (bytes == 0) {
|
|
|
|
|
s->eos = 1;
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
@ -521,7 +529,8 @@ static int read_samples(struct ast_filestream *s, float ***pcm)
|
|
|
|
|
* \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)
|
|
|
|
|
static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s,
|
|
|
|
|
int *whennext)
|
|
|
|
|
{
|
|
|
|
|
int clipflag = 0;
|
|
|
|
|
int i;
|
|
|
|
@ -535,25 +544,25 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s, int *whennext
|
|
|
|
|
|
|
|
|
|
while (1) {
|
|
|
|
|
/* See ifwe have filled up an audio frame yet */
|
|
|
|
|
if(samples_out == SAMPLES_MAX)
|
|
|
|
|
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)
|
|
|
|
|
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++)
|
|
|
|
|
|
|
|
|
|
for (j = 0; j < samples_in; j++)
|
|
|
|
|
accumulator[j] = 0.0;
|
|
|
|
|
|
|
|
|
|
for(i = 0; i < s->vi.channels; i++) {
|
|
|
|
|
for (i = 0; i < s->vi.channels; i++) {
|
|
|
|
|
mono = pcm[i];
|
|
|
|
|
for (j = 0; j < samples_in; j++) {
|
|
|
|
|
accumulator[j] += mono[j];
|
|
|
|
@ -561,27 +570,26 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s, int *whennext
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
for (j = 0; j < samples_in; j++) {
|
|
|
|
|
val = accumulator[j] * 32767.0 / s->vi.channels;
|
|
|
|
|
if(val > 32767) {
|
|
|
|
|
val = accumulator[j] * 32767.0 / s->vi.channels;
|
|
|
|
|
if (val > 32767) {
|
|
|
|
|
val = 32767;
|
|
|
|
|
clipflag = 1;
|
|
|
|
|
}
|
|
|
|
|
if(val < -32768) {
|
|
|
|
|
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));
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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) {
|
|
|
|
|
if (samples_out > 0) {
|
|
|
|
|
s->fr.frametype = AST_FRAME_VOICE;
|
|
|
|
|
s->fr.subclass = AST_FORMAT_SLINEAR;
|
|
|
|
|
s->fr.offset = AST_FRIENDLY_OFFSET;
|
|
|
|
@ -591,7 +599,7 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s, int *whennext
|
|
|
|
|
s->fr.mallocd = 0;
|
|
|
|
|
s->fr.samples = samples_out;
|
|
|
|
|
*whennext = samples_out;
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
return &s->fr;
|
|
|
|
|
} else {
|
|
|
|
|
return NULL;
|
|
|
|
@ -618,17 +626,21 @@ static int ogg_vorbis_trunc(struct ast_filestream *s)
|
|
|
|
|
* \return 0 on success, -1 on failure.
|
|
|
|
|
*/
|
|
|
|
|
|
|
|
|
|
static int ogg_vorbis_seek(struct ast_filestream *s, long sample_offset, int whence) {
|
|
|
|
|
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) {
|
|
|
|
|
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) {
|
|
|
|
|
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;
|
|
|
|
|
}
|
|
|
|
@ -650,7 +662,7 @@ int load_module()
|
|
|
|
|
int unload_module()
|
|
|
|
|
{
|
|
|
|
|
return ast_format_unregister(name);
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
int usecount()
|
|
|
|
|
{
|
|
|
|
@ -667,11 +679,3 @@ char *key()
|
|
|
|
|
{
|
|
|
|
|
return ASTERISK_GPL_KEY;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
/*
|
|
|
|
|
Local Variables:
|
|
|
|
|
mode: C
|
|
|
|
|
c-file-style: "linux"
|
|
|
|
|
indent-tabs-mode: t
|
|
|
|
|
End:
|
|
|
|
|
*/
|
|
|
|
|