diff options
author | rizzo <rizzo@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-04-04 15:40:47 +0000 |
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committer | rizzo <rizzo@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-04-04 15:40:47 +0000 |
commit | 0a1c1c5ec275d544db20b6b3d17a54ef29e147b5 (patch) | |
tree | 661df038d96b3150d3f5284733b36e4c2d5ff093 | |
parent | 16bd637be21a6cd0274ba576c9f03c16d0b04425 (diff) |
ogg_vorbis now compiles so put it back in.
On passing, remove an unnecessary initializazion in format_sln.c
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17285 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | formats/Makefile | 2 | ||||
-rw-r--r-- | formats/format_ogg_vorbis.c | 220 | ||||
-rw-r--r-- | formats/format_sln.c | 1 |
3 files changed, 87 insertions, 136 deletions
diff --git a/formats/Makefile b/formats/Makefile index 4c857ea3e..85e488ee9 100644 --- a/formats/Makefile +++ b/formats/Makefile @@ -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,$(MODS)) diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c index 061684f74..013751c48 100644 --- a/formats/format_ogg_vorbis.c +++ b/formats/format_ogg_vorbis.c @@ -48,10 +48,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$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"; } diff --git a/formats/format_sln.c b/formats/format_sln.c index d3a759131..98a9ad5f5 100644 --- a/formats/format_sln.c +++ b/formats/format_sln.c @@ -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) { |