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: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 10 insertions, 2 deletions

diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h
index 75e2c8763..798a6d6e0 100644
--- a/include/asterisk/audiohook.h
+++ b/include/asterisk/audiohook.h
@@ -65,7 +65,12 @@ enum ast_audiohook_flags {
 	AST_AUDIOHOOK_MUTE_WRITE = (1 << 5),    /*!< audiohook should be mute frames written */
 };
 
-#define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*< Tolerance in milliseconds for audiohooks synchronization */
+enum ast_audiohook_init_flags {
+	/*! Audiohook manipulate callback is capable of handling slinear at any sample rate.
+	 * Without enabling this flag on initialization the manipulation callback is guaranteed
+	 * 8khz audio only. */
+	AST_AUDIOHOOK_MANIPULATE_ALL_RATES = (1 << 0),
+};
 
 struct ast_audiohook;
 
@@ -97,6 +102,7 @@ struct ast_audiohook {
 	ast_cond_t trigger;                                    /*!< Trigger condition (if enabled) */
 	enum ast_audiohook_type type;                          /*!< Type of audiohook */
 	enum ast_audiohook_status status;                      /*!< Status of the audiohook */
+	enum ast_audiohook_init_flags init_flags;              /*!< Init flags */
 	const char *source;                                    /*!< Who this audiohook ultimately belongs to */
 	unsigned int flags;                                    /*!< Flags on the audiohook */
 	struct ast_slinfactory read_factory;                   /*!< Factory where frames read from the channel, or read from the whisper source will go through */
@@ -107,6 +113,7 @@ struct ast_audiohook {
 	struct ast_trans_pvt *trans_pvt;                       /*!< Translation path for reading frames */
 	ast_audiohook_manipulate_callback manipulate_callback; /*!< Manipulation callback */
 	struct ast_audiohook_options options;                  /*!< Applicable options */
+	unsigned int hook_internal_samp_rate;                           /*!< internal read/write sample rate on the audiohook.*/
 	AST_LIST_ENTRY(ast_audiohook) list;                    /*!< Linked list information */
 };
 
@@ -116,9 +123,10 @@ struct ast_audiohook_list;
  * \param audiohook Audiohook structure
  * \param type Type of audiohook to initialize this as
  * \param source Who is initializing this audiohook
+ * \param init flags
  * \return Returns 0 on success, -1 on failure
  */
-int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source);
+int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags flags);
 
 /*! \brief Destroys an audiohook structure
  * \param audiohook Audiohook structure