From 5d9ed5739aab2b302efc178d21e6c75672369db3 Mon Sep 17 00:00:00 2001 From: kpfleming Date: Sun, 12 Feb 2006 04:28:58 +0000 Subject: major dialplan functions update deprecate LANGUAGE() and MUSICCLASS(), in favor of CHANNEL() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9674 f38db490-d61c-443f-a65b-d21fe96a405b --- formats/format_ogg_vorbis.c | 230 ++++++++++++++++++++++---------------------- 1 file changed, 117 insertions(+), 113 deletions(-) (limited to 'formats') diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c index ffee9ed76..408f19e13 100644 --- a/formats/format_ogg_vorbis.c +++ b/formats/format_ogg_vorbis.c @@ -20,7 +20,7 @@ * \arg File name extension: ogg * \ingroup formats */ - + #include #include #include @@ -48,37 +48,35 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/file.h" #include "asterisk/logger.h" #include "asterisk/module.h" - #define SAMPLES_MAX 160 #define BLOCK_SIZE 4096 - struct ast_filestream { void *reserved[AST_RESERVED_POINTERS]; - + FILE *f; - + /* structures for handling the Ogg container */ - ogg_sync_state oy; + ogg_sync_state oy; ogg_stream_state os; - ogg_page og; - ogg_packet op; + ogg_page og; + ogg_packet op; /* structures for handling Vorbis audio data */ - vorbis_info vi; - vorbis_comment vc; + vorbis_info vi; + vorbis_comment vc; vorbis_dsp_state vd; - vorbis_block vb; + vorbis_block vb; /*! \brief Indicates whether this filestream is set up for reading or writing. */ int writing; - + /*! \brief Indicates whether an End of Stream condition has been detected. */ int eos; - + /*! \brief Buffer to hold audio data. */ short buffer[SAMPLES_MAX]; - + /*! \brief Asterisk frame object. */ struct ast_frame fr; char waste[AST_FRIENDLY_OFFSET]; @@ -86,6 +84,7 @@ struct ast_filestream { }; AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock); + static int glistcnt = 0; static char *name = "ogg_vorbis"; @@ -97,7 +96,7 @@ static char *exts = "ogg"; * \param f File that points to on disk storage of the OGG/Vorbis data. * \return The new filestream. */ -static struct ast_filestream *ogg_vorbis_open(FILE *f) +static struct ast_filestream *ogg_vorbis_open(FILE * f) { int i; int bytes; @@ -107,7 +106,7 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) struct ast_filestream *tmp; - if((tmp = malloc(sizeof(struct ast_filestream)))) { + if ((tmp = malloc(sizeof(struct ast_filestream)))) { memset(tmp, 0, sizeof(struct ast_filestream)); tmp->writing = 0; @@ -120,24 +119,26 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) ogg_sync_wrote(&tmp->oy, bytes); result = ogg_sync_pageout(&tmp->oy, &tmp->og); - if(result != 1) { - if(bytes < BLOCK_SIZE) { + 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"); + ast_log(LOG_ERROR, + "Input does not appear to be an Ogg bitstream.\n"); } fclose(f); ogg_sync_clear(&tmp->oy); free(tmp); return NULL; } - + ogg_stream_init(&tmp->os, ogg_page_serialno(&tmp->og)); vorbis_info_init(&tmp->vi); vorbis_comment_init(&tmp->vc); - if(ogg_stream_pagein(&tmp->os, &tmp->og) < 0) { - ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n"); + if (ogg_stream_pagein(&tmp->os, &tmp->og) < 0) { + ast_log(LOG_ERROR, + "Error reading first page of Ogg bitstream data.\n"); fclose(f); ogg_stream_clear(&tmp->os); vorbis_comment_clear(&tmp->vc); @@ -146,8 +147,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) free(tmp); return NULL; } - - if(ogg_stream_packetout(&tmp->os, &tmp->op) != 1) { + + if (ogg_stream_packetout(&tmp->os, &tmp->op) != 1) { ast_log(LOG_ERROR, "Error reading initial header packet.\n"); fclose(f); ogg_stream_clear(&tmp->os); @@ -157,8 +158,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) free(tmp); return NULL; } - - if(vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) { + + if (vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) { ast_log(LOG_ERROR, "This Ogg bitstream does not contain Vorbis audio data.\n"); fclose(f); ogg_stream_clear(&tmp->os); @@ -168,20 +169,20 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) free(tmp); return NULL; } - + i = 0; - while(i < 2) { - while(i < 2){ + while (i < 2) { + while (i < 2) { result = ogg_sync_pageout(&tmp->oy, &tmp->og); - if(result == 0) + if (result == 0) break; - if(result == 1) { + if (result == 1) { ogg_stream_pagein(&tmp->os, &tmp->og); - while(i < 2) { - result = ogg_stream_packetout(&tmp->os,&tmp->op); - if(result == 0) + while (i < 2) { + result = ogg_stream_packetout(&tmp->os, &tmp->op); + if (result == 0) break; - if(result < 0) { + if (result < 0) { ast_log(LOG_ERROR, "Corrupt secondary header. Exiting.\n"); fclose(f); ogg_stream_clear(&tmp->os); @@ -199,7 +200,7 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE); bytes = fread(buffer, 1, BLOCK_SIZE, f); - if(bytes == 0 && i < 2) { + if (bytes == 0 && i < 2) { ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n"); fclose(f); ogg_stream_clear(&tmp->os); @@ -211,16 +212,18 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) } ogg_sync_wrote(&tmp->oy, bytes); } - + ptr = tmp->vc.user_comments; - while(*ptr){ + while (*ptr) { ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *ptr); ++ptr; } - ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate); - ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor); + ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", + tmp->vi.channels, tmp->vi.rate); + ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n", + tmp->vc.vendor); - if(tmp->vi.channels != 1) { + if (tmp->vi.channels != 1) { ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n"); ogg_stream_clear(&tmp->os); vorbis_comment_clear(&tmp->vc); @@ -229,9 +232,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) free(tmp); return NULL; } - - if(tmp->vi.rate != 8000) { + if (tmp->vi.rate != 8000) { ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n"); fclose(f); ogg_stream_clear(&tmp->os); @@ -243,11 +245,11 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) free(tmp); return NULL; } - + vorbis_synthesis_init(&tmp->vd, &tmp->vi); vorbis_block_init(&tmp->vd, &tmp->vb); - 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"); fclose(f); ogg_stream_clear(&tmp->os); @@ -272,7 +274,8 @@ static struct ast_filestream *ogg_vorbis_open(FILE *f) * \param comment Comment that should be embedded in the OGG/Vorbis file. * \return A new filestream. */ -static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment) +static struct ast_filestream *ogg_vorbis_rewrite(FILE * f, + const char *comment) { ogg_packet header; ogg_packet header_comm; @@ -280,7 +283,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment) struct ast_filestream *tmp; - if((tmp = malloc(sizeof(struct ast_filestream)))) { + if ((tmp = malloc(sizeof(struct ast_filestream)))) { memset(tmp, 0, sizeof(struct ast_filestream)); tmp->writing = 1; @@ -288,7 +291,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment) vorbis_info_init(&tmp->vi); - if(vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) { + if (vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) { ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n"); free(tmp); return NULL; @@ -296,7 +299,7 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment) vorbis_comment_init(&tmp->vc); vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX"); - if(comment) + if (comment) vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment); vorbis_analysis_init(&tmp->vd, &tmp->vi); @@ -304,21 +307,22 @@ static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment) ogg_stream_init(&tmp->os, rand()); - vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm, &header_code); - ogg_stream_packetin(&tmp->os, &header); + vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm, + &header_code); + ogg_stream_packetin(&tmp->os, &header); ogg_stream_packetin(&tmp->os, &header_comm); ogg_stream_packetin(&tmp->os, &header_code); - while(!tmp->eos) { - if(ogg_stream_flush(&tmp->os, &tmp->og) == 0) + while (!tmp->eos) { + if (ogg_stream_flush(&tmp->os, &tmp->og) == 0) break; fwrite(tmp->og.header, 1, tmp->og.header_len, tmp->f); fwrite(tmp->og.body, 1, tmp->og.body_len, tmp->f); - if(ogg_page_eos(&tmp->og)) + if (ogg_page_eos(&tmp->og)) tmp->eos = 1; } - 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"); fclose(f); ogg_stream_clear(&tmp->os); @@ -345,16 +349,16 @@ static void write_stream(struct ast_filestream *s) while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) { vorbis_analysis(&s->vb, NULL); vorbis_bitrate_addblock(&s->vb); - + while (vorbis_bitrate_flushpacket(&s->vd, &s->op)) { ogg_stream_packetin(&s->os, &s->op); while (!s->eos) { - if(ogg_stream_pageout(&s->os, &s->og) == 0) { + if (ogg_stream_pageout(&s->os, &s->og) == 0) { break; } 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: -*/ -- cgit v1.2.3