ogg_vorbis now compiles so put it back in.
authorLuigi Rizzo <rizzo@icir.org>
Tue, 4 Apr 2006 15:40:47 +0000 (15:40 +0000)
committerLuigi Rizzo <rizzo@icir.org>
Tue, 4 Apr 2006 15:40:47 +0000 (15:40 +0000)
On passing, remove an unnecessary initializazion in format_sln.c

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

formats/Makefile
formats/format_ogg_vorbis.c
formats/format_sln.c

index 4c857ea..85e488e 100644 (file)
@@ -22,8 +22,6 @@ MODS:=$(filter-out format_au.so,$(MODS))
 # 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 $(CROSS_COMPILE_TARGET)/usr/local/include/vorbis/codec.h),)
   MODS:=$(filter-out format_ogg_vorbis.so,$(MODS))
index 061684f..013751c 100644 (file)
@@ -48,10 +48,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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
+#define        BUF_SIZE        (2*SAMPLES_MAX)
 
-struct vorbis_desc {
+#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;
@@ -71,14 +78,6 @@ struct vorbis_desc {
        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.
  * \param f File that points to on disk storage of the OGG/Vorbis data.
@@ -94,12 +93,11 @@ static int ogg_vorbis_open(struct ast_filestream *s)
        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 @@ error:
                }
 
                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 @@ error:
        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 @@ return 0;
  * \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 vorbis_desc *tmp = (struct vorbis_desc *)s->private;
 
-       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);
+       tmp->writing = 1;
 
-               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_info_init(&tmp->vi);
 
-               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);
+       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_analysis_init(&tmp->vd, &tmp->vi);
-               vorbis_block_init(&tmp->vd, &tmp->vb);
+       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);
 
-               ogg_stream_init(&tmp->os, rand());
+       vorbis_analysis_init(&tmp->vd, &tmp->vi);
+       vorbis_block_init(&tmp->vd, &tmp->vb);
 
-               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);
+       ogg_stream_init(&tmp->os, rand());
 
-               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;
-               }
+       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);
 
-               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();
+       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 tmp;
+
+       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 @@ static void write_stream(struct ast_filestream *s)
                                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 @@ static void write_stream(struct ast_filestream *s)
  * \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 @@ 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;
-       }
+       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 @@ static int ogg_vorbis_write(struct ast_filestream *s, struct ast_frame *f)
  * \brief Close a OGG/Vorbis filestream.
  * \param s A OGG/Vorbis filestream.
  */
-static void ogg_vorbis_close(struct ast_filestream *s)
+static void ogg_vorbis_close(struct ast_filestream *fs)
 {
-       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();
+       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 @@ static void ogg_vorbis_close(struct ast_filestream *s)
  * \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 @@ static int read_samples(struct ast_filestream *s, float ***pcm)
                        /* 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 @@ 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,
+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;
+       struct vorbis_desc *s = (struct vorbis_desc *)fs->private;
+       short *buf = (short *)(fs->fr.data);    /* SLIN data buffer */
 
-       while (1) {
-               /* See ifwe have filled up an audio frame yet */
-               if (samples_out == SAMPLES_MAX)
-                       break;
+       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 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s,
                /* 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 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s,
                        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 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s,
        }
 
        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 @@ static int ogg_vorbis_trunc(struct ast_filestream *s)
  * \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 char *ogg_vorbis_getcomment(struct ast_filestream *s)
 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 @@ static const struct ast_format vorbis_f = {
        .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 load_module()
 
 int unload_module()
 {
-       return ast_format_unregister(name);
+       return ast_format_unregister(vorbis_f.name);
 }
 
 int usecount()
@@ -611,7 +565,7 @@ int usecount()
 
 char *description()
 {
-       return desc;
+       return "OGG/Vorbis audio";
 }
 
 
index d3a7591..98a9ad5 100644 (file)
@@ -53,7 +53,6 @@ static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
 
        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) {