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Added a new file CHANGES which currently includes the changes made si…
…nce UniMRCP 1.0.0 version.
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| Changes for UniMRCP-1.1.0 | ||
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| * Added missing recognizer header fields used for voice enrollment (speaker-dependent recognition). | ||
| * Added support for custom MRCP header fields. | ||
| * Added init.d script. The script was originally submitted by pdeschen. Thanks. | ||
| * The function apt_log_file_open() makes a copy of dir_pass and file_name variables passed from a user application. | ||
| * Set the default profile name in umcscenarios.xml to "uni2". | ||
| * In the PocketSphinx plugin, instead of using one common timeout for detection of speech activity and inactivity, use two different timeouts: one for activity and the other for inactivity detection. | ||
| * In the recognition results sent from the PocketSphinx plugin, set both <instance> and <input> elements. | ||
| * Added support for SIP redirection with 300 Multiple Choices used by the Nuance Resource Manager. | ||
| * Added support for feature-tags set in the SIP Accept-Contact header field in an outgoing SIP INVITE message sent to the Nuance Resource Manager. | ||
| * Use strcasecmp() instead of apr_strnatcasecmp() to match RTSP resource names (Issue-94). | ||
| * Added support for speaker verification and identification resource. | ||
| * Fixed a race condition in the PocketSphinx plugin which caused the server to crash. | ||
| * Added sample 8kHz and 16 kHz voiceprints in the data directory which are used by the umc application for a sample verification scenario. | ||
| * Added a new accessor function to the client API to get an audio stream associated with the specified channel. The function name is mrcp_application_audio_stream_get(). | ||
| * Added missing recognizer methods used for voice enrollment and interpretation. | ||
| * Updated the state machine of the recognizer resource to take into consideration requests, responses and events used for interpretation. | ||
| * Fixed the use of recognition timer in the PocketSphinx plugin. | ||
| * Applied a patch to the apt_log routine which allows the log file to be appended instead of being overwritten. The patch was submitted by Dani. Thanks. | ||
| * Added a new parameter to the function apt_log_file_open() which specifies whether the log file should be appended or overwritten. | ||
| * Set an MRCP version specific completion cause in the PocketSphinx plugin. | ||
| * Tweaked DTMF detector's energy thresholds to eliminate false positives during in-band (from audio) DTMF detection. Thanks Vali. | ||
| * Added the ability to retrieve an external object associated with the MRCP session through the log handler (apt_log_ext_handler_f). | ||
| * Fixed the formatting of float values in the header fields. Applied a patch submitted by Randy (Issue-108). Thanks. | ||
| * Fixed an interoperability issue with AVP. The mid attribute is not required when the SDP contains only one m-line. | ||
| * Fixed the processing of a response to the SIP OPTIONS request used for resource discovery (Issue-112). | ||
| * Added mandatory attributes for the SSML <speak> element in the sample speak.xml file. | ||
| * Took into consideration the RTP marker in order to re-sync the jitter buffer on a new talkspurt. Audio data loss could be experienced in the RTP receiver in case of consecutive SPEAK (for client) or RECOGNIZE (for server) requests. | ||
| * Modified the "prepare" utility project to use the new location of PthreadVC2.dll which is now built from source with other dependencies. | ||
| * Fixed the build of C++ MRCP plugins for platforms other than Win32. Thanks Vali. | ||
| * Instead of discarding a non-aligned RTP packet, adjust the timestamp and write available frames to the jitter buffer (Issue-122). | ||
| * Fixed a crash in the RTSP client stack when the server closes a TCP connection while the associated RTSP session is being destroyed (Issue-124). | ||
| * Fixed the processing of RTSP TEARDOWN requests being timed out. Applied a patch submitted to Issue-125 by Chris. Thanks. | ||
| * Took out unused tags (variables) to compile with Sofia-SIP 1.12.11. | ||
| * Added support for Visual Studio 2010. | ||
| * Fixed apt_log_output_mode_check() which returned TRUE if any mode was enabled or checked regardless their correspondence. Thanks Vali. | ||
| * Added a new constructor function unimrcp_client_create2() which allows to pass the client XML configuration not by a file, but rather by a string parameter. Thanks Vali. | ||
| * Added the ability to take and use parameters set by the plugin in a response to the GET-PARAMS request. Thanks Vali. | ||
| * Fixed the processing of more than one pending application requests upon reception of a SIP BYE message from the server. | ||
| * Enhanced the debug output by adding task message identifier to the log statements "Signal Message" and "Process Message". | ||
| * Fixed a potential crash related to the use of pollsets. | ||
| * Added support for Sofia-SIP's TPTAG_LOG() and TPTAG_DUMP() tags which can be enabled from the client and server configuration to print out and/or dump SIP messages. | ||
| * Fixed the loading of the client configuration parameter <offer-new-connection>. | ||
| * Added support for the adaptive jitter buffer. Applied a reworked patch submitted by Erik. Thanks. | ||
| * Enhanced the detection of a new RTP talkspurt by implicitly setting the RTP marker if a gap between two RTP packets is more than the specified threshold (INTER_TALSKPUSRT_GAP = 1000 msec). | ||
| * Allow the initial playout delay in the jitter buffer to be set to 0. | ||
| * Implemented a time skew detection algorithm for RTP streams. The detection can be enabled and used for both the adaptive and static jitter buffer. | ||
| * Added support for redirection of RTP traces (RTP_TRACE, JB_TRACE) to the debug output window of Visual Studio. | ||
| * Modified the MPF test application to read a raw PCM data from one file, transmit it over RTP, and write the data back to another file. | ||
| * Enhanced helper functions which operate on the MRCP header to properly set, get and inherit header fields (Issue-110). | ||
| * Set the libtool parameters link_all_deplibs and link_all_deplibs_CXX to "yes" by default, with an option to disable them (--disable-interlib-deps), if ever needed. This fixes a link error on recent Debian/Ubuntu distributions. | ||
| * Enhanced the processing of the RTP named events. | ||
| * Enhanced the UniMRCP Windows service manager. Thanks Vali. | ||
| * Added support for a binary recognition grammar used in RecogScenario by the sample umc application. Thanks Vali. | ||
| * Fixed a potential buffer overflow in apt_text_pair_array_insert(). Thanks Vali. | ||
| * Modified the apr.m4 and apu.m4 macros to use '--link-ld' instead of '--link-libtool --libs' for the APR library dependencies. This addresses the problem with a wrong -L path to the expat library. | ||
| * Set prerequisite version for autoconf to 2.59. | ||
| * Added a checking for pkg-config to the configure script. | ||
| * For logging purposes, pass a string identifier of the RTSP/MRCPv1 signaling agent to the RTSP client and server stacks. | ||
| * Remove a socket descriptor from the pollset only if the descriptor has been properly added to the pollset. Otherwise, this operation could cause a crash. | ||
| * Respond to client user application requests with failure if a new session couldn't be created due to an error in initialization of the SIP stack (Issue-127). | ||
| * Added a sample SRGS ABNF grammar to the data directory. | ||
| * When originating an offer from the client, take into account capabilities of an audio stream created by the client user application. | ||
| * Added a new function to the client API to retrieve a SIP/RTSP response code received from the server (Issue-90). The support is incomplete. | ||
| * Added a new option (-v or --version) to the unimrcpserver as well as the sample umc and unimrcpclient applications. | ||
| * Corrected FileType in Windows resources from DLL to APP. Thanks Vali. | ||
| * Added a Windows resource file for the unimrcpservice application. |