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https://github.com/asterisk/asterisk.git
synced 2025-10-18 10:44:28 +00:00
Largely simplify format handlers (for file copy etc.)
collecting common functions in a single place and removing them from the individual handlers. The full description is on mantis, http://bugs.digium.com/view.php?id=6375 and only the ogg_vorbis handler needs to be converted to the new structure. As a result of this change, format_au.c and format_pcm_alaw.c should go away (in a separate commit) as their functionality (trivial) has been merged in another file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Luigi Rizzo
@@ -51,11 +51,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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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struct vorbis_desc {
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/* structures for handling the Ogg container */
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ogg_sync_state oy;
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ogg_stream_state os;
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@@ -73,14 +69,6 @@ struct ast_filestream {
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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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char empty;
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};
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AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock);
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@@ -96,176 +84,123 @@ 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 int ogg_vorbis_open(struct ast_filestream *s)
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{
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int i;
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int bytes;
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int result;
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char **ptr;
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char *buffer;
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struct vorbis_desc *tmp = (struct vorbis_desc *)s->private;
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struct ast_filestream *tmp;
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tmp->writing = 0;
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tmp->f = f;
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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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ogg_sync_init(&tmp->oy);
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tmp->writing = 0;
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tmp->f = 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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ogg_sync_wrote(&tmp->oy, bytes);
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ogg_sync_init(&tmp->oy);
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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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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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}
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ogg_sync_clear(&tmp->oy);
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return -1;
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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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error:
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ogg_stream_clear(&tmp->os);
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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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ogg_sync_clear(&tmp->oy);
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return -1;
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}
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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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goto error;
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}
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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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goto error;
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}
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for (i = 0; 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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break;
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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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break;
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if(result < 0) {
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ast_log(LOG_ERROR, "Corrupt secondary header. Exiting.\n");
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goto error;
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}
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vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
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i++;
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}
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}
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}
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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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ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
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goto error;
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}
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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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ast_log(LOG_ERROR, "Run out of data...\n");
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} else {
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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,
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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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vorbis_info_clear(&tmp->vi);
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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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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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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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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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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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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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i = 0;
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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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break;
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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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break;
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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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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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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vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
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i++;
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}
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}
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}
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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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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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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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_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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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",
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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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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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vorbis_info_clear(&tmp->vi);
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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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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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vorbis_block_clear(&tmp->vb);
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vorbis_dsp_clear(&tmp->vd);
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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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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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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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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vorbis_block_clear(&tmp->vb);
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vorbis_dsp_clear(&tmp->vd);
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vorbis_comment_clear(&tmp->vc);
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vorbis_info_clear(&tmp->vi);
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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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glistcnt++;
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ast_mutex_unlock(&ogg_vorbis_lock);
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ast_update_use_count();
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}
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return tmp;
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ptr = tmp->vc.user_comments;
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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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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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goto error;
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}
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if(tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
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ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
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vorbis_block_clear(&tmp->vb);
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vorbis_dsp_clear(&tmp->vd);
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goto error;
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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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ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
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vorbis_block_clear(&tmp->vb);
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vorbis_dsp_clear(&tmp->vd);
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goto error;
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}
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glistcnt++;
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ast_mutex_unlock(&ogg_vorbis_lock);
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ast_update_use_count();
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return 0;
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}
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/*!
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@@ -291,7 +226,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE * f,
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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, DEFAULT_SAMPLE_RATE, 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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@@ -440,9 +375,6 @@ static void ogg_vorbis_close(struct ast_filestream *s)
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if (s->writing) {
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ogg_sync_clear(&s->oy);
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}
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fclose(s->f);
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free(s);
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}
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/*!
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@@ -643,18 +575,28 @@ static char *ogg_vorbis_getcomment(struct ast_filestream *s)
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return NULL;
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}
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static struct ast_format_lock me = { .usecnt = -1 };
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static const struct ast_format vorbis_f = {
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.name =
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.ext =
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.format = AST_FORMAT_SLINEAR,
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.open = ogg_vorbis_open,
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.rewrite = ogg_vorbis_rewrite,
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.write = ogg_vorbis_write,
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.seek = ogg_vorbis_seek,
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.trunc = ogg_vorbis_trunc,
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.tell = ogg_vorbis_tell,
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.read = ogg_vorbis_read,
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.close = ogg_vorbis_close,
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.buf_sie = BUF_SIZE + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct vorbis_desc),
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.lockp = &me,
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};
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int load_module()
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{
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return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
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ogg_vorbis_open,
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ogg_vorbis_rewrite,
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ogg_vorbis_write,
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ogg_vorbis_seek,
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ogg_vorbis_trunc,
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ogg_vorbis_tell,
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ogg_vorbis_read,
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ogg_vorbis_close,
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ogg_vorbis_getcomment);
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return ast_format_register(&vorbis_f);
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}
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int unload_module()
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@@ -664,7 +606,7 @@ int unload_module()
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int usecount()
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{
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return glistcnt;
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return me.usecnt;
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}
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char *description()
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