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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-06-14 19:39:12 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-06-14 19:39:12 +0000
commitf0424318479863edeed2a6a128e6d1ac926966e3 (patch)
tree4fd144260e58c9a66d331fc81b46473723452479 /formats
parentc8ca7d2f263f5161ce1b29ec2c9de835131fc950 (diff)
Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'formats')
-rw-r--r--formats/format_g726.c4
-rw-r--r--formats/format_ogg_vorbis.c6
-rw-r--r--formats/format_wav.c2
3 files changed, 5 insertions, 7 deletions
diff --git a/formats/format_g726.c b/formats/format_g726.c
index 41cd87a08..09504bf6c 100644
--- a/formats/format_g726.c
+++ b/formats/format_g726.c
@@ -78,9 +78,7 @@ static int g726_open(struct ast_filestream *tmp, int rate)
{
struct g726_desc *s = (struct g726_desc *)tmp->private;
s->rate = rate;
- if (option_debug)
- ast_log(LOG_DEBUG, "Created filestream G.726-%dk.\n",
- 40 - s->rate * 8);
+ ast_debug(1, "Created filestream G.726-%dk.\n", 40 - s->rate * 8);
return 0;
}
diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c
index 9d80b63d4..bc2cc79ac 100644
--- a/formats/format_ogg_vorbis.c
+++ b/formats/format_ogg_vorbis.c
@@ -175,9 +175,9 @@ error:
}
for (ptr = tmp->vc.user_comments; *ptr; ptr++)
- ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *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_debug(1, "OGG/Vorbis comment: %s\n", *ptr);
+ ast_debug(1, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
+ ast_debug(1, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor);
if (tmp->vi.channels != 1) {
ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
diff --git a/formats/format_wav.c b/formats/format_wav.c
index 194f82458..d37b55cd4 100644
--- a/formats/format_wav.c
+++ b/formats/format_wav.c
@@ -364,7 +364,7 @@ static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
bytes = fs->maxlen - here;
if (bytes < 0)
bytes = 0;
-/* ast_log(LOG_DEBUG, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */
+/* ast_debug(1, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
s->fr.mallocd = 0;