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

10 files changed, 270 insertions, 97 deletions

diff --git a/include/asterisk/_private.h b/include/asterisk/_private.h
index 560c8c169..a0b171254 100644
--- a/include/asterisk/_private.h
+++ b/include/asterisk/_private.h
@@ -90,4 +90,16 @@ int ast_xmldoc_load_documentation(void);
  */
 int ast_plc_reload(void);
 
+/*!
+ * \brief Init the ast_format attribute interface register container.
+ */
+int ast_format_attr_init(void);
+
+/*!
+ * \brief Init the Asterisk global format list after all format attribute modules have been loaded
+ */
+int ast_format_list_init(void);
+
+/*! \brief initializes the rtp engine arrays */
+int ast_rtp_engine_init(void);
 #endif /* _ASTERISK__PRIVATE_H */
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
diff --git a/include/asterisk/format.h b/include/asterisk/format.h
index 09212abc8..67e4178a2 100644
--- a/include/asterisk/format.h
+++ b/include/asterisk/format.h
@@ -26,8 +26,9 @@
 #ifndef _AST_FORMAT_H_
 #define _AST_FORMAT_H_
 
+#include "asterisk/astobj2.h"
+#include "asterisk/silk.h"
 #define AST_FORMAT_ATTR_SIZE 128
-
 #define AST_FORMAT_INC 100000
 
 /*! This is the value that ends a var list of format attribute
@@ -55,32 +56,49 @@ enum ast_format_id {
 	AST_FORMAT_G726_AAL2        = 5 + AST_FORMAT_TYPE_AUDIO,
 	/*! ADPCM (IMA) */
 	AST_FORMAT_ADPCM            = 6 + AST_FORMAT_TYPE_AUDIO,
-	/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
-	AST_FORMAT_SLINEAR          = 7 + AST_FORMAT_TYPE_AUDIO,
 	/*! LPC10, 180 samples/frame */
-	AST_FORMAT_LPC10            = 8 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_LPC10            = 7 + AST_FORMAT_TYPE_AUDIO,
 	/*! G.729A audio */
-	AST_FORMAT_G729A            = 9 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_G729A            = 8 + AST_FORMAT_TYPE_AUDIO,
 	/*! SpeeX Free Compression */
-	AST_FORMAT_SPEEX            = 10 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_SPEEX            = 9 + AST_FORMAT_TYPE_AUDIO,
 	/*! iLBC Free Compression */
-	AST_FORMAT_ILBC             = 11 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_ILBC             = 10 + AST_FORMAT_TYPE_AUDIO,
 	/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
-	AST_FORMAT_G726             = 12 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_G726             = 11 + AST_FORMAT_TYPE_AUDIO,
 	/*! G.722 */
-	AST_FORMAT_G722             = 13 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_G722             = 12 + AST_FORMAT_TYPE_AUDIO,
 	/*! G.722.1 (also known as Siren7, 32kbps assumed) */
-	AST_FORMAT_SIREN7           = 14 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_SIREN7           = 13 + AST_FORMAT_TYPE_AUDIO,
 	/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
-	AST_FORMAT_SIREN14          = 15 + AST_FORMAT_TYPE_AUDIO,
-	/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
-	AST_FORMAT_SLINEAR16        = 16 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_SIREN14          = 14 + AST_FORMAT_TYPE_AUDIO,
 	/*! G.719 (64 kbps assumed) */
-	AST_FORMAT_G719             = 17 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_G719             = 15 + AST_FORMAT_TYPE_AUDIO,
 	/*! SpeeX Wideband (16kHz) Free Compression */
-	AST_FORMAT_SPEEX16          = 18 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_SPEEX16          = 16 + AST_FORMAT_TYPE_AUDIO,
 	/*! Raw mu-law data (G.711) */
-	AST_FORMAT_TESTLAW          = 19 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_TESTLAW          = 17 + AST_FORMAT_TYPE_AUDIO,
+	/*! SILK format */
+	AST_FORMAT_SILK             = 18 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
+	AST_FORMAT_SLINEAR          = 19 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (12000 Hz) PCM */
+	AST_FORMAT_SLINEAR12        = 20 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
+	AST_FORMAT_SLINEAR16        = 21 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (24000 Hz) PCM */
+	AST_FORMAT_SLINEAR24        = 22 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (32000 Hz) PCM */
+	AST_FORMAT_SLINEAR32        = 23 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (44100 Hz) PCM just because we can. */
+	AST_FORMAT_SLINEAR44        = 24 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (48000 Hz) PCM */
+	AST_FORMAT_SLINEAR48        = 25 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (96000 Hz) PCM */
+	AST_FORMAT_SLINEAR96        = 26 + AST_FORMAT_TYPE_AUDIO,
+	/*! Raw 16-bit Signed Linear (192000 Hz) PCM.  maybe we're taking this too far. */
+	AST_FORMAT_SLINEAR192       = 27 + AST_FORMAT_TYPE_AUDIO,
+	AST_FORMAT_SPEEX32          = 28 + AST_FORMAT_TYPE_AUDIO,
 
 	/*! H.261 Video */
 	AST_FORMAT_H261             = 1 + AST_FORMAT_TYPE_VIDEO,
@@ -107,6 +125,7 @@ enum ast_format_id {
 /*! Determine what type of media a ast_format_id is. */
 #define AST_FORMAT_GET_TYPE(id) (((int) (id / AST_FORMAT_INC)) * AST_FORMAT_INC)
 
+
 /*! \brief This structure contains the buffer used for format attributes */
 struct ast_format_attr {
 	/*! The buffer formats can use to represent attributes */
@@ -133,6 +152,22 @@ enum ast_format_cmp_res {
 	AST_FORMAT_CMP_SUBSET,
 };
 
+/*! \brief Definition of supported media formats (codecs) */
+struct ast_format_list {
+	struct ast_format format; /*!< The unique format. */
+	char name[64];	/*!< short name */
+	unsigned int samplespersecond; /*!< Number of samples per second (8000/16000) */
+	char desc[128];	/*!< Description */
+	int fr_len;	/*!< Single frame length in bytes */
+	int min_ms;	/*!< Min value */
+	int max_ms;	/*!< Max value */
+	int inc_ms;	/*!< Increment */
+	int def_ms;	/*!< Default value */
+	unsigned int flags;	/*!< Smoother flags */
+	int cur_ms;	/*!< Current value */
+	int custom_entry;
+};
+
 /*! \brief A format must register an attribute interface if it requires the use of the format attributes void pointer */
 struct ast_format_attr_interface {
 	/*! format type */
@@ -154,6 +189,34 @@ struct ast_format_attr_interface {
 	/*! \brief Set format capabilities from a list of key value pairs ending with AST_FORMAT_ATTR_END.
 	 * \note This function does not need to call va_end of the va_list. */
 	void (* const format_attr_set)(struct ast_format_attr *format_attr, va_list ap);
+
+	/*!
+	 * \brief Find out if format capabilities in va_list are in format.
+	 * \note This function does not need to call va_end of the va_list.
+	 *
+	 * \note This function is optional.  In many cases the format_attr_cmp
+	 * function can be used to derive these results.  If it is possible
+	 * that some format attributes have no bearing on the equality of two formats, this
+	 * function must exist.
+	 *
+	 * \retval 0 if all attributes exist
+	 * \retval -1 if any of the attributes not present
+	 */
+	int (* const format_attr_isset)(const struct ast_format_attr *format_attr, va_list ap);
+
+	/*
+	 * \brief Return a value for a specific format key.   Return that value in the void pointer.
+	 *
+	 * \note It is not expected that all key value pairs can be returned, but those that can should
+	 * be documented as such.
+	 *
+	 * \note This function is optional if key value pairs are not allowed to be accessed.  This
+	 * will result in -1 always being returned.
+	 *
+	 * \retval 0 Success, value was found and copied into void pointer.
+	 * \retval -1 failure, Value was either not found, or not allowed to be accessed.
+	 */
+	int (* const format_attr_get_val)(const struct ast_format_attr *format_attr, int key, void *val);
 };
 
 /*!
@@ -218,7 +281,18 @@ void ast_format_clear(struct ast_format *format);
  * \return 0, The format key value pairs are within the capabilities defined in this structure.
  * \return -1, The format key value pairs are _NOT_ within the capabilities of this structure.
  */
-int ast_format_isset(struct ast_format *format, ... );
+int ast_format_isset(const struct ast_format *format, ... );
+
+/*!
+ * \brief Get a value from a format containing attributes.
+ * \note The key represents the format attribute to be retrieved, and the void pointer
+ * is to the structure that value will be stored in.  It must be known what structure a
+ * key represents.
+ *
+ * \retval 0, success
+ * \retval -1, failure
+ */
+int ast_format_get_value(const struct ast_format *format, int key, void *value);
 
 /*!
  * \brief Compare ast_formats structures
@@ -287,6 +361,52 @@ struct ast_format *ast_format_from_old_bitfield(struct ast_format *dst, uint64_t
 enum ast_format_id ast_format_id_from_old_bitfield(uint64_t src);
 
 /*!
+ * \brief Retrieve the global format list in a read only array.
+ * \note ast_format_list_destroy must be called on every format
+ * list retrieved from this function.
+ */
+const struct ast_format_list *ast_format_list_get(size_t *size);
+
+/*!
+ * \brief Destroy an ast_format_list gotten from ast_format_list_get()
+ */
+const struct ast_format_list *ast_format_list_destroy(const struct ast_format_list *list);
+
+/*! \brief Get the name of a format
+ * \param format id of format
+ * \return A static string containing the name of the format or "unknown" if unknown.
+ */
+const char* ast_getformatname(const struct ast_format *format);
+
+/*! \brief Returns a string containing all formats pertaining to an format id.
+ * \param buf a buffer for the output string
+ * \param size size of buf (bytes)
+ * \param format id.
+ * \return The return value is buf.
+ */
+char* ast_getformatname_multiple_byid(char *buf, size_t size, enum ast_format_id id);
+
+/*!
+ * \brief Gets a format from a name.
+ * \param name string of format
+ * \param format structure to return the format in.
+ * \return This returns the format pointer given to it on success and NULL on failure
+ */
+struct ast_format *ast_getformatbyname(const char *name, struct ast_format *format);
+
+/*!
+ * \brief Get a name from a format 
+ * \param format to get name of
+ * \return This returns a static string identifying the format on success, 0 on error.
+ */
+const char *ast_codec2str(struct ast_format *format);
+
+/*!
+ * \brief Get the sample rate for a given format.
+ */
+int ast_format_rate(const struct ast_format *format);
+
+/*!
  * \brief register ast_format_attr_interface with core.
  *
  * \retval 0 success
@@ -303,8 +423,12 @@ int ast_format_attr_reg_interface(const struct ast_format_attr_interface *interf
 int ast_format_attr_unreg_interface(const struct ast_format_attr_interface *interface);
 
 /*!
- * \brief Init the ast_format attribute interface register container.
+ * \brief Determine if a format is 16bit signed linear of any sample rate. 
  */
-int ast_format_attr_init(void);
+int ast_format_is_slinear(const struct ast_format *format);
 
+/*!
+ * \brief Get the best slinear format id for a given sample rate
+ */
+enum ast_format_id ast_format_slin_by_rate(unsigned int rate);
 #endif /* _AST_FORMAT_H */
diff --git a/include/asterisk/format_cap.h b/include/asterisk/format_cap.h
index cdb5421f9..234767685 100644
--- a/include/asterisk/format_cap.h
+++ b/include/asterisk/format_cap.h
@@ -70,7 +70,7 @@ void *ast_format_cap_destroy(struct ast_format_cap *cap);
  * what is placed in the ast_format_cap structure.  The actual
  * input format ptr is not stored.
  */
-void ast_format_cap_add(struct ast_format_cap *cap, struct ast_format *format);
+void ast_format_cap_add(struct ast_format_cap *cap, const struct ast_format *format);
 
 /*!
  * \brief Add all formats Asterisk knows about for a specific type to
@@ -155,6 +155,15 @@ void ast_format_cap_remove_all(struct ast_format_cap *cap);
 void ast_format_cap_set(struct ast_format_cap *cap, struct ast_format *format);
 
 /*!
+ * \brief Find if input ast_format is within the capabilities of the ast_format_cap object
+ * then return the compatible format from the capabilities structure in the result.
+ *
+ * \retval 1 format is compatible with formats held in ast_format_cap object.
+ * \retval 0 format is not compatible with any formats in ast_format_cap object.
+ */
+int ast_format_cap_get_compatible_format(const struct ast_format_cap *cap, const struct ast_format *format, struct ast_format *result);
+
+/*!
  * \brief Find if ast_format is within the capabilities of the ast_format_cap object.
  *
  * retval 1 format is compatible with formats held in ast_format_cap object.
@@ -163,6 +172,14 @@ void ast_format_cap_set(struct ast_format_cap *cap, struct ast_format *format);
 int ast_format_cap_iscompatible(const struct ast_format_cap *cap, const struct ast_format *format);
 
 /*!
+ * \brief Finds the best quality audio format for a given format id and returns it in result.
+ *
+ * \retval 1 format found and set to result structure.
+ * \retval 0 no format found, result structure is cleared.
+ */
+int ast_format_cap_best_byid(const struct ast_format_cap *cap, enum ast_format_id, struct ast_format *result);
+
+/*!
  * \brief is cap1 identical to cap2
  *
  * retval 1 true, identical
@@ -278,4 +295,14 @@ uint64_t ast_format_cap_to_old_bitfield(const struct ast_format_cap *cap);
  */
 void ast_format_cap_from_old_bitfield(struct ast_format_cap *dst, uint64_t src);
 
+/*! \brief Get the names of a set of formats
+ * \param buf a buffer for the output string
+ * \param size size of buf (bytes)
+ * \param format the format (combined IDs of codecs)
+ * Prints a list of readable codec names corresponding to "format".
+ * ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
+ * \return The return value is buf.
+ */
+char* ast_getformatname_multiple(char *buf, size_t size, struct ast_format_cap *cap);
+
 #endif /* _AST_FORMATCAP_H */
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 63cbb952f..e02df42ed 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -427,23 +427,6 @@ struct ast_option_header {
 		uint8_t data[0];
 };
 
-
-/*! \brief Definition of supported media formats (codecs) */
-struct ast_format_list {
-	enum ast_format_id id;    /*!< The format unique id */
-	char *name;	/*!< short name */
-	int samplespersecond; /*!< Number of samples per second (8000/16000) */
-	char *desc;	/*!< Description */
-	int fr_len;	/*!< Single frame length in bytes */
-	int min_ms;	/*!< Min value */
-	int max_ms;	/*!< Max value */
-	int inc_ms;	/*!< Increment */
-	int def_ms;	/*!< Default value */
-	unsigned int flags;	/*!< Smoother flags */
-	int cur_ms;	/*!< Current value */
-};
-
-
 /*! \brief  Requests a frame to be allocated 
  * 
  * \param source 
@@ -505,37 +488,6 @@ void ast_swapcopy_samples(void *dst, const void *src, int samples);
  */
 int ast_parse_allow_disallow(struct ast_codec_pref *pref, struct ast_format_cap *cap, const char *list, int allowing);
 
-/*! \brief Get the name of a format
- * \param format id of format
- * \return A static string containing the name of the format or "unknown" if unknown.
- */
-char* ast_getformatname(struct ast_format *format);
-
-/*! \brief Get the names of a set of formats
- * \param buf a buffer for the output string
- * \param size size of buf (bytes)
- * \param format the format (combined IDs of codecs)
- * Prints a list of readable codec names corresponding to "format".
- * ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
- * \return The return value is buf.
- */
-char* ast_getformatname_multiple(char *buf, size_t size, struct ast_format_cap *cap);
-
-/*!
- * \brief Gets a format from a name.
- * \param name string of format
- * \param format structure to return the format in.
- * \return This returns the format pointer given to it on success and NULL on failure
- */
-struct ast_format *ast_getformatbyname(const char *name, struct ast_format *format);
-
-/*! \brief Get a name from a format 
- * Gets a name from a format
- * \param format to get name of
- * \return This returns a static string identifying the format on success, 0 on error.
- */
-char *ast_codec2str(struct ast_format *format);
-
 /*! \name AST_Smoother 
 */
 /*@{ */
@@ -582,8 +534,6 @@ struct ast_frame *ast_smoother_read(struct ast_smoother *s);
 #endif
 /*@} Doxygen marker */
 
-const struct ast_format_list *ast_get_format_list_index(int index);
-const struct ast_format_list *ast_get_format_list(size_t *size);
 void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix);
 
 /*! \brief Returns the number of samples contained in the frame */
@@ -622,26 +572,6 @@ int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
 int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
 
 /*!
- * \brief Get the sample rate for a given format.
- */
-static force_inline int ast_format_rate(struct ast_format *format)
-{
-	switch (format->id) {
-	case AST_FORMAT_G722:
-	case AST_FORMAT_SLINEAR16:
-	case AST_FORMAT_SIREN7:
-	case AST_FORMAT_SPEEX16:
-		return 16000;
-	case AST_FORMAT_SIREN14:
-		return 32000;
-	case AST_FORMAT_G719:
-		return 48000;
-	default:
-		return 8000;
-	}
-}
-
-/*!
  * \brief Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR 
  */
 int ast_frame_clear(struct ast_frame *frame);
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index f13538321..4c5753e84 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -1048,6 +1048,19 @@ void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp
 struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload);
 
 /*!
+ * \brief Retrieve the actual ast_format stored on the codecs structure for a specific payload
+ *
+ * \param codecs Codecs structure to look in
+ * \param payload Numerical payload to look up
+ *
+ * \retval pointer to format structure on success
+ * \retval NULL on failure
+ *
+ * \since 1.10
+ */
+struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload);
+
+/*!
  * \brief Get the sample rate associated with known RTP payload types
  *
  * \param asterisk_format True if the value in format is to be used.
@@ -1798,6 +1811,15 @@ struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy);
 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
 
+/*! \brief Custom formats declared in codecs.conf at startup must be communicated to the rtp_engine
+ * so their mime type can payload number can be initialized. */
+int ast_rtp_engine_load_format(const struct ast_format *format);
+
+/*! \brief Formats requiring the use of a format attribute interface must have that
+ * interface registered in order for the rtp engine to handle it correctly.  If an
+ * attribute interface is unloaded, this function must be called to notify the rtp_engine. */
+int ast_rtp_engine_unload_format(const struct ast_format *format);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif
diff --git a/include/asterisk/silk.h b/include/asterisk/silk.h
new file mode 100644
index 000000000..5da827e7e
--- /dev/null
+++ b/include/asterisk/silk.h
@@ -0,0 +1,44 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011, Digium, Inc.
+ *
+ * David Vossel <dvossel@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief SILK Format Attributes
+ *
+ * \author David Vossel <dvossel@digium.com>
+ */
+#ifndef _AST_FORMAT_SILK_H_
+#define _AST_FORMAT_SILK_H_
+
+/*! SILK format attribute key value pairs, all are accessible through ast_format_get_value()*/
+enum silk_attr_keys {
+	SILK_ATTR_KEY_SAMP_RATE, /*!< value is silk_attr_vals enum */
+	SILK_ATTR_KEY_DTX, /*!< value is an int, 1 dtx is enabled, 0 dtx not enabled. */
+	SILK_ATTR_KEY_FEC, /*!< value is an int, 1 encode with FEC, 0 do not use FEC. */
+	SILK_ATTR_KEY_PACKETLOSS_PERCENTAGE, /*!< value is an int (0-100), Represents estimated packetloss in uplink direction.*/
+	SILK_ATTR_KEY_MAX_BITRATE, /*!< value is an int */
+};
+
+enum silk_attr_vals {
+	SILK_ATTR_VAL_SAMP_8KHZ = (1 << 0),
+	SILK_ATTR_VAL_SAMP_12KHZ = (1 << 1),
+	SILK_ATTR_VAL_SAMP_16KHZ = (1 << 2),
+	SILK_ATTR_VAL_SAMP_24KHZ = (1 << 3),
+};
+
+#endif /* _AST_FORMAT_SILK_H */
diff --git a/include/asterisk/slinfactory.h b/include/asterisk/slinfactory.h
index 003c6ac28..324c0ae28 100644
--- a/include/asterisk/slinfactory.h
+++ b/include/asterisk/slinfactory.h
@@ -56,11 +56,11 @@ void ast_slinfactory_init(struct ast_slinfactory *sf);
  * \brief Initialize a slinfactory
  *
  * \param sf The slinfactory to initialize
- * \param sample_rate The output sample rate desired
+ * \param slin_out the slinear output format desired.
  *
  * \return 0 on success, non-zero on failure
  */
-int ast_slinfactory_init_rate(struct ast_slinfactory *sf, unsigned int sample_rate);
+int ast_slinfactory_init_with_format(struct ast_slinfactory *sf, const struct ast_format *slin_out);
 
 /*!
  * \brief Destroy the contents of a slinfactory
diff --git a/include/asterisk/time.h b/include/asterisk/time.h
index 2ffc691b8..c78ff2db0 100644
--- a/include/asterisk/time.h
+++ b/include/asterisk/time.h
@@ -171,7 +171,7 @@ struct timeval ast_tv(ast_time_t sec, ast_suseconds_t usec),
 AST_INLINE_API(
 struct timeval ast_samp2tv(unsigned int _nsamp, unsigned int _rate),
 {
-	return ast_tv(_nsamp / _rate, ((_nsamp % _rate) * (4000000 / _rate)) / 4); /* this calculation is accurate up to 32000Hz. */
+	return ast_tv(_nsamp / _rate, (_nsamp % _rate) * (1000000 / (float) _rate));
 }
 )
 
diff --git a/include/asterisk/translate.h b/include/asterisk/translate.h
index 7e73cd1b1..8545f0ae5 100644
--- a/include/asterisk/translate.h
+++ b/include/asterisk/translate.h
@@ -133,7 +133,7 @@ enum ast_trans_cost_table {
  * Generic plc is only available for dstfmt = SLINEAR
  */
 struct ast_translator {
-	const char name[80];                   /*!< Name of translator */
+	char name[80];                         /*!< Name of translator */
 	struct ast_format src_format;          /*!< Source format */
 	struct ast_format dst_format;          /*!< Destination format */
 
@@ -204,6 +204,12 @@ struct ast_translator {
 struct ast_trans_pvt {
 	struct ast_translator *t;
 	struct ast_frame f;         /*!< used in frameout */
+	/*! If a translation path using a format with attributes requires the output
+	 * to be a specific set of attributes, this variable will be set describing those
+	 * attributes to the translator.  Otherwise, the translator must choose a set
+	 * of format attributes for the destination that preserves the quality of the
+	 * audio in the best way possible. */
+	struct ast_format explicit_dst;
 	int samples;                /*!< samples available in outbuf */
 	/*! \brief actual space used in outbuf */
 	int datalen;
@@ -213,7 +219,7 @@ struct ast_trans_pvt {
 		unsigned char *uc;      /*!< the useful portion of the buffer */
 		int16_t *i16;
 		uint8_t *ui8;
-	} outbuf; 
+	} outbuf;
 	plc_state_t *plc;           /*!< optional plc pointer */
 	struct ast_trans_pvt *next; /*!< next in translator chain */
 	struct timeval nextin;