Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 1 insertions, 1 deletions

diff --git a/res/res_mutestream.c b/res/res_mutestream.c
index 4faf46478..41b2fd831 100644
--- a/res/res_mutestream.c
+++ b/res/res_mutestream.c
@@ -173,7 +173,7 @@ static struct ast_datastore *initialize_mutehook(struct ast_channel *chan)
 		ast_datastore_free(datastore);
 		return NULL;
 	}
-	ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute");
+	ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
 	mute->audiohook.manipulate_callback = mute_callback;
 	datastore->data = mute;
 	return datastore;