[svn-commits] trunk r17285 - in /trunk/formats: Makefile
format_ogg_vorbis.c format_sln.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Apr 4 08:40:51 MST 2006
Author: rizzo
Date: Tue Apr 4 10:40:47 2006
New Revision: 17285
URL: http://svn.digium.com/view/asterisk?rev=17285&view=rev
Log:
ogg_vorbis now compiles so put it back in.
On passing, remove an unnecessary initializazion in format_sln.c
Modified:
trunk/formats/Makefile
trunk/formats/format_ogg_vorbis.c
trunk/formats/format_sln.c
Modified: trunk/formats/Makefile
URL: http://svn.digium.com/view/asterisk/trunk/formats/Makefile?rev=17285&r1=17284&r2=17285&view=diff
==============================================================================
--- trunk/formats/Makefile (original)
+++ trunk/formats/Makefile Tue Apr 4 10:40:47 2006
@@ -21,8 +21,6 @@
#
# OGG/Vorbis format
# (on FreeBSD is in /usr/local/include/...
-
-MODS:=$(filter-out format_ogg_vorbis.so,$(MODS))
ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/vorbis/codec.h),)
ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/vorbis/codec.h),)
Modified: trunk/formats/format_ogg_vorbis.c
URL: http://svn.digium.com/view/asterisk/trunk/formats/format_ogg_vorbis.c?rev=17285&r1=17284&r2=17285&view=diff
==============================================================================
--- trunk/formats/format_ogg_vorbis.c (original)
+++ trunk/formats/format_ogg_vorbis.c Tue Apr 4 10:40:47 2006
@@ -48,10 +48,17 @@
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
+
+/*
+ * this is the number of samples we deal with. Samples are converted
+ * to SLINEAR so each one uses 2 bytes in the buffer.
+ */
#define SAMPLES_MAX 160
-#define BLOCK_SIZE 4096
-
-struct vorbis_desc {
+#define BUF_SIZE (2*SAMPLES_MAX)
+
+#define BLOCK_SIZE 4096 /* used internally in the vorbis routines */
+
+struct vorbis_desc { /* format specific parameters */
/* structures for handling the Ogg container */
ogg_sync_state oy;
ogg_stream_state os;
@@ -70,14 +77,6 @@
/*! \brief Indicates whether an End of Stream condition has been detected. */
int eos;
};
-
-AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock);
-
-static int glistcnt = 0;
-
-static char *name = "ogg_vorbis";
-static char *desc = "OGG/Vorbis audio";
-static char *exts = "ogg";
/*!
* \brief Create a new OGG/Vorbis filestream and set it up for reading.
@@ -94,12 +93,11 @@
struct vorbis_desc *tmp = (struct vorbis_desc *)s->private;
tmp->writing = 0;
- tmp->f = f;
ogg_sync_init(&tmp->oy);
buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
- bytes = fread(buffer, 1, BLOCK_SIZE, f);
+ bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
ogg_sync_wrote(&tmp->oy, bytes);
result = ogg_sync_pageout(&tmp->oy, &tmp->og);
@@ -159,29 +157,25 @@
}
buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
- bytes = fread(buffer, 1, BLOCK_SIZE, f);
- if(bytes == 0 && i < 2) {
+ bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
+ if (bytes == 0 && i < 2) {
ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
goto error;
}
ogg_sync_wrote(&tmp->oy, bytes);
}
- ptr = tmp->vc.user_comments;
- while(*ptr){
+ for (ptr = tmp->vc.user_comments; *ptr; 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);
- if(tmp->vi.channels != 1) {
+ if (tmp->vi.channels != 1) {
ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
goto error;
}
-
- if(tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
+ if (tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
vorbis_block_clear(&tmp->vb);
vorbis_dsp_clear(&tmp->vd);
@@ -191,16 +185,7 @@
vorbis_synthesis_init(&tmp->vd, &tmp->vi);
vorbis_block_init(&tmp->vd, &tmp->vb);
- if(ast_mutex_lock(&ogg_vorbis_lock)) {
- ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
- vorbis_block_clear(&tmp->vb);
- vorbis_dsp_clear(&tmp->vd);
- goto error;
- }
- glistcnt++;
- ast_mutex_unlock(&ogg_vorbis_lock);
- ast_update_use_count();
-return 0;
+ return 0;
}
/*!
@@ -209,77 +194,56 @@
* \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,
+static int ogg_vorbis_rewrite(struct ast_filestream *s,
const char *comment)
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
-
- struct ast_filestream *tmp;
-
- if ((tmp = malloc(sizeof(struct ast_filestream)))) {
- memset(tmp, 0, sizeof(struct ast_filestream));
-
- tmp->writing = 1;
- tmp->f = f;
-
- vorbis_info_init(&tmp->vi);
-
- if (vorbis_encode_init_vbr(&tmp->vi, 1, DEFAULT_SAMPLE_RATE, 0.4)) {
- ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
- free(tmp);
- return NULL;
- }
-
- vorbis_comment_init(&tmp->vc);
- vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
- if (comment)
- vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
-
- vorbis_analysis_init(&tmp->vd, &tmp->vi);
- vorbis_block_init(&tmp->vd, &tmp->vb);
-
- ogg_stream_init(&tmp->os, rand());
-
- 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)
- 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))
- tmp->eos = 1;
- }
-
- 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);
- vorbis_block_clear(&tmp->vb);
- vorbis_dsp_clear(&tmp->vd);
- vorbis_comment_clear(&tmp->vc);
- vorbis_info_clear(&tmp->vi);
- free(tmp);
- return NULL;
- }
- glistcnt++;
- ast_mutex_unlock(&ogg_vorbis_lock);
- ast_update_use_count();
- }
- return tmp;
+ struct vorbis_desc *tmp = (struct vorbis_desc *)s->private;
+
+ tmp->writing = 1;
+
+ vorbis_info_init(&tmp->vi);
+
+ if (vorbis_encode_init_vbr(&tmp->vi, 1, DEFAULT_SAMPLE_RATE, 0.4)) {
+ ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
+ return -1;
+ }
+
+ vorbis_comment_init(&tmp->vc);
+ vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
+ if (comment)
+ vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
+
+ vorbis_analysis_init(&tmp->vd, &tmp->vi);
+ vorbis_block_init(&tmp->vd, &tmp->vb);
+
+ ogg_stream_init(&tmp->os, rand());
+
+ 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)
+ break;
+ fwrite(tmp->og.header, 1, tmp->og.header_len, s->f);
+ fwrite(tmp->og.body, 1, tmp->og.body_len, s->f);
+ if (ogg_page_eos(&tmp->og))
+ tmp->eos = 1;
+ }
+
+ return 0;
}
/*!
* \brief Write out any pending encoded data.
* \param s A OGG/Vorbis filestream.
*/
-static void write_stream(struct ast_filestream *s)
+static void write_stream(struct vorbis_desc *s, FILE *f)
{
while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
vorbis_analysis(&s->vb, NULL);
@@ -291,8 +255,8 @@
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);
+ fwrite(s->og.header, 1, s->og.header_len, f);
+ fwrite(s->og.body, 1, s->og.body_len, f);
if (ogg_page_eos(&s->og)) {
s->eos = 1;
}
@@ -307,11 +271,12 @@
* \param f An frame containing audio to be written to the filestream.
* \return -1 ifthere was an error, 0 on success.
*/
-static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
+static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f)
{
int i;
float **buffer;
short *data;
+ struct vorbis_desc *s = (struct vorbis_desc *)fs->private;
if (!s->writing) {
ast_log(LOG_ERROR, "This stream is not set up for writing!\n");
@@ -334,13 +299,12 @@
buffer = vorbis_analysis_buffer(&s->vd, f->samples);
- for (i = 0; i < f->samples; i++) {
- buffer[0][i] = data[i] / 32768.f;
- }
+ for (i = 0; i < f->samples; i++)
+ buffer[0][i] = (double)data[i] / 32768.0;
vorbis_analysis_wrote(&s->vd, f->samples);
- write_stream(s);
+ write_stream(s, fs->f);
return 0;
}
@@ -349,21 +313,15 @@
* \brief Close a OGG/Vorbis filestream.
* \param s A OGG/Vorbis filestream.
*/
-static void ogg_vorbis_close(struct ast_filestream *s)
-{
- if (ast_mutex_lock(&ogg_vorbis_lock)) {
- ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
- return;
- }
- glistcnt--;
- ast_mutex_unlock(&ogg_vorbis_lock);
- ast_update_use_count();
+static void ogg_vorbis_close(struct ast_filestream *fs)
+{
+ struct vorbis_desc *s = (struct vorbis_desc *)fs->private;
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);
- write_stream(s);
+ write_stream(s, fs->f);
}
ogg_stream_clear(&s->os);
@@ -383,12 +341,13 @@
* \param pcm Pointer to a buffere to store audio data in.
*/
-static int read_samples(struct ast_filestream *s, float ***pcm)
+static int read_samples(struct ast_filestream *fs, float ***pcm)
{
int samples_in;
int result;
char *buffer;
int bytes;
+ struct vorbis_desc *s = (struct vorbis_desc *)fs->private;
while (1) {
samples_in = vorbis_synthesis_pcmout(&s->vd, pcm);
@@ -445,7 +404,7 @@
/* get a buffer from OGG to read the data into */
buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE);
/* read more data from the file descriptor */
- bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
+ bytes = fread(buffer, 1, BLOCK_SIZE, fs->f);
/* Tell OGG how many bytes we actually read into the buffer */
ogg_sync_wrote(&s->oy, bytes);
if (bytes == 0) {
@@ -461,26 +420,30 @@
* \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,
+static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
int *whennext)
{
int clipflag = 0;
int i;
int j;
- float **pcm;
- float *mono;
double accumulator[SAMPLES_MAX];
int val;
int samples_in;
int samples_out = 0;
-
- while (1) {
- /* See ifwe have filled up an audio frame yet */
- if (samples_out == SAMPLES_MAX)
- break;
+ struct vorbis_desc *s = (struct vorbis_desc *)fs->private;
+ short *buf = (short *)(fs->fr.data); /* SLIN data buffer */
+
+ fs->fr.frametype = AST_FRAME_VOICE;
+ fs->fr.subclass = AST_FORMAT_SLINEAR;
+ fs->fr.mallocd = 0;
+ FR_SET_BUF(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+
+ while (samples_out != SAMPLES_MAX) {
+ float **pcm;
+ int len = SAMPLES_MAX - samples_out;
/* See ifVorbis decoder has some audio data for us ... */
- samples_in = read_samples(s, &pcm);
+ samples_in = read_samples(fs, &pcm);
if (samples_in <= 0)
break;
@@ -488,17 +451,15 @@
/* 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);
-
+ if (samples_in > len)
+ samples_in = len;
for (j = 0; j < samples_in; j++)
accumulator[j] = 0.0;
for (i = 0; i < s->vi.channels; i++) {
- mono = pcm[i];
- for (j = 0; j < samples_in; j++) {
+ float *mono = pcm[i];
+ for (j = 0; j < samples_in; j++)
accumulator[j] += mono[j];
- }
}
for (j = 0; j < samples_in; j++) {
@@ -506,12 +467,11 @@
if (val > 32767) {
val = 32767;
clipflag = 1;
- }
- if (val < -32768) {
+ } else if (val < -32768) {
val = -32768;
clipflag = 1;
}
- s->buffer[samples_out + j] = val;
+ buf[samples_out + j] = val;
}
if (clipflag)
@@ -522,17 +482,11 @@
}
if (samples_out > 0) {
- s->fr.frametype = AST_FRAME_VOICE;
- s->fr.subclass = AST_FORMAT_SLINEAR;
- s->fr.offset = AST_FRIENDLY_OFFSET;
- s->fr.datalen = samples_out * 2;
- s->fr.data = s->buffer;
- s->fr.src = name;
- s->fr.mallocd = 0;
- s->fr.samples = samples_out;
+ fs->fr.datalen = samples_out * 2;
+ fs->fr.samples = samples_out;
*whennext = samples_out;
- return &s->fr;
+ return &fs->fr;
} else {
return NULL;
}
@@ -557,8 +511,8 @@
* \param whence Location to measure
* \return 0 on success, -1 on failure.
*/
-
-static int ogg_vorbis_seek(struct ast_filestream *s, off_t sample_offset, int whence) {
+static int ogg_vorbis_seek(struct ast_filestream *s, off_t sample_offset, int whence)
+{
ast_log(LOG_WARNING, "Seeking is not supported on OGG/Vorbis streams!\n");
return -1;
}
@@ -578,8 +532,8 @@
static struct ast_format_lock me = { .usecnt = -1 };
static const struct ast_format vorbis_f = {
- .name =
- .ext =
+ .name = "ogg_vorbis",
+ .exts = "ogg",
.format = AST_FORMAT_SLINEAR,
.open = ogg_vorbis_open,
.rewrite = ogg_vorbis_rewrite,
@@ -589,7 +543,7 @@
.tell = ogg_vorbis_tell,
.read = ogg_vorbis_read,
.close = ogg_vorbis_close,
- .buf_sie = BUF_SIZE + AST_FRIENDLY_OFFSET,
+ .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
.desc_size = sizeof(struct vorbis_desc),
.lockp = &me,
};
@@ -601,7 +555,7 @@
int unload_module()
{
- return ast_format_unregister(name);
+ return ast_format_unregister(vorbis_f.name);
}
int usecount()
@@ -611,7 +565,7 @@
char *description()
{
- return desc;
+ return "OGG/Vorbis audio";
}
Modified: trunk/formats/format_sln.c
URL: http://svn.digium.com/view/asterisk/trunk/formats/format_sln.c?rev=17285&r1=17284&r2=17285&view=diff
==============================================================================
--- trunk/formats/format_sln.c (original)
+++ trunk/formats/format_sln.c Tue Apr 4 10:40:47 2006
@@ -53,7 +53,6 @@
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
- s->fr.offset = AST_FRIENDLY_OFFSET;
s->fr.mallocd = 0;
FR_SET_BUF(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
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