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    221 rsdk speex trabalhos encontrados, preços em USD

    i am looking for the follwing requirements : 1- swift Microphone class ( similar to Android AudioRecord) with following functionality : - record audio voice data and store it into Int16 ( cshort) and int8 (byte) arrays so...tcpip network. - it must use sample rate of 8000 hz( so i can use it for android devices ) 2- play back audio on IOS device from Int16 array or Int8 array ( similar to Andoid AudioTrack) - audio play back must use sample rate of 8000 hz (8k) ( audio data will be comming from both android or iphone through the network) - i needed it to be in int16 array as i am going to use it in speex encoder / decoder , that work i am going to do once the this work is done - you can use swift ( avaudioengine tech, or audioqueue tech what ever suit...

    $329 (Avg Bid)
    $329 Média
    19 ofertas

    Thank you for reading this project. We are using PulseAudio on an Openwrt project and we are implementing echo cancel for a speakerphone application. We are having some difficulty getting the echo cancel to work (there is a distorted echo at the distant end), we believe this may be related to timing of the echo cancel stream with the microphone stream. I have an engineer on the job who needs some help and advice. Please let me know if there is someone who is an expert to help us resolve this issue. Thank you

    $38 / hr (Avg Bid)
    $38 / hr Média
    3 ofertas
    Speex wrapper Encerrado left

    looking for someone who can create a Speex codec wrapper for a voip app .. the app will send and receive voice messages need to pass it through a speex codex..

    $562 (Avg Bid)
    $562 Média
    4 ofertas

    ...SIMPLE protocol Session timers Network diagnostics Auto Answer and Do Not Disturb Modes File transfer (compatibile with any SIP server) File sharing (compatibile with any SIP server) Remote Desktop over SIP Fax (beta version) Call and Chat History Audio and video recording Audio Codecs: G.711-Alaw, G.711-uLaw, G.723.1, G729, GSM, iLBC, L16, Speex, OPUS Video Codecs: MPEG1, MPEG4, Theora, DIV3, MJPG, H263, H264, VP8 WideBand and Ultra WideBand codec (speex, opus) Audio tuning wizard Dynamic Jitter Buffer Packet loss concealment (PLC) Automatic Gain Control (AGC) Acoustic Echo Cancellation (AEC) Voice activity detection (VAD) Noise supression Push notifications Auto QoS Dynamic Threshold Algorithm for Silence Detection Network handling: UPNP, STUN, ICE, IP Translation, Fire...

    $3000 - $5000
    $3000 - $5000
    0 ofertas
    QT Voice Over IP Encerrado left

    would like Jamulus to be turned into a voice over IP application for multiple users. mostly gui and tray applet work probably because it already works. would be neat to incorporate midi with it aswell. speex might be better option ? Audio Jamulus Midi qmidinet ninjam multimidicast v1.3 linux is preferred development environment.

    $1176 (Avg Bid)
    $1176 Média
    5 ofertas

    Basic Linphone App with Customization with Own branding App to App calling App to mobile calling Multiple calls management Call transfer, pause and resume Audio con...(merge calls into a conference) Instant Messaging Pictures and files sharing Address Book Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications: zRTP, DTLS, SRTP Bluetooth headset support Languages: English, French, Japanese, Arabic Account creation assistant Dedicated tablet user interface Audio  using codecs such as G722,G729,SILK, SPEEX Video calls using codecs such as VP8, H264, MPEG4 Integration with push notification (requires compatible SIP server, sip service has push enabled) Peer to Peer Audio and Video calling using ICE Sup...

    $1600 (Avg Bid)
    $1600 Média
    22 ofertas
    MP3BookHelper Update Encerrado left

    I have for years been using a abandoned software to manage my audiobook collections' tagging and renaming. I'm seeking someone who is willing to update the software to include some additional file types (MP3, Ogg/Vorbis, FLAC, MPC, MP4/AAC, MP2, Speex, TrueAudio, WavPack, WMA, WAV and AIFF files, etc) and to update for 64 bit and windows 10., even though they did not write it originally. I can get more into specifics as to functions and features. I do plan to continue to offer the update version as freeware/opensource. Original sourceforge information: This takes it to 2.3.4 version. There is a 2.3.5.2 version available but I cannot locate the source files for it. I you are interested

    $16 / hr (Avg Bid)
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    2 ofertas

    We have a Linux based IOT which has handsfree speaker and 6 MIC. We use PJSIP to make telephone calls. The receiving party hears an Echo. We need to developed an echo cancellation for our PJSIP base applications on Linux IOT which currently using speex echo cancellation which does not work well. We need to hire a person who have experience on write AEC improve on speex AEC or to provide better AEC canceller so it will work well on our Linux platform You should be able to work remotely with our team and our hardware and are well versed with PJSIP and AEC.

    $1706 (Avg Bid)
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    5 ofertas

    I have a code in which speex codec is not some modifications on code to make it work

    $7 - $18
    $7 - $18
    0 ofertas

    hello i have voipswitch server , i want to get android mobile dialer for my customer , so they can use my app to sign up from the application dialer , check balance , recharge i donot want any payment solution as i wi...payment solution as i will just sell the voucher card to my customer Faster call connectivity and registration with SIP Server Runs behind NAT or private IP and in blocked country Supports 3G/4G, GPRS and Wi-Fi for internet connectivity Displays my customers account Balance Syncs with your contacts in your device Address book Displays Call Logs and Call Duration Auto Signal Switching Supports Speex, G.723, G.722, G.729 and AMR codecs Compatible with SIP standard voipswitch my logo is ready i will puplish the app to the play store pleas send me the cost and detail...

    $709 (Avg Bid)
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    12 ofertas

    we have a voice portal using web socket for connection and wish to build a story / ai robot using XR 897 to build and to write the complete programming - so that at the end PCM file send to cloud and voice from cloud streaming back. PCM files can be compressed using SpeeX.

    $2250 (Avg Bid)
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    2 ofertas

    I'd like a c++ console application that can take raw pcm audio data (i can provide a file) which is in the form of an array of floats (float[48000]) then use Speex () to encode, then to decode, and display to console the size of the encoded byte array...and also to verify that the decoded float array matches the input. Is this something you can do? I have one day Thank you

    $50 (Avg Bid)
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    3 ofertas

    ...User shall be able to check the his commission on the web and receive an sms once every day with his daily made commission and accumulated balance in total - We shall be able to execute any USSD string and read its answer remotely by IP - We shall be able to send SMS and read any answer remotely by IP - Compression used to send to smartphone has to be ultra low bandwith consuming, perhaps using Speex - Signalling of SIP has to be matched to GSM on the phone - The user shall be able to see whether WiFi can be used for making calls (maybe ports are blocked or phone is on WiFi, but still needs to login to the WiFi internet page to use internet) - We shall be able to turn on and turn off caller ID to be shown to called party - We shall be able to change IMEI number on the phon...

    $544 (Avg Bid)
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    1 ofertas

    We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone who has hands-on practical experience with Linphone who can properly address this issue.

    $451 (Avg Bid)
    $451 Média
    2 ofertas

    A microphone is attached to a standard Adobe NetStream that is connected to a local UDP socket so that loopback can intercept the Speex formatted output. That output needs to be pushed in to another Netstream (through a second UDP loopback to which that netstream is connected). That's it. If you know the internal protocol of NetStreams/NetConnections, you know how to handle the order of commands and it's just a 1 hour job. If you don't know what I'm talking about, please don't bid! It should be FULLY AS3 based functions (which is not that difficult, basically it's just catching some packets!) to catch and forward. FIXED PRICE: $200 IMPORTANT: To exclude automatic bids, everything that doesn't start with "Hi Peter" is deleted.

    $250 (Avg Bid)
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    1 ofertas

    ...speed 4 Mbytes Flash, 16 Mbytes SDRAM 4 Ohm Speaker Driver, and 2 UARTs 2 channel Audio Codec with sampling rates up to 48kHz Industrial Temperature: -40°C to +85°C Software Features Fusion RTOS TFTP Client Telnet Server Voice Engine with RTP (includes support for RoIP products) Voice Engine Codecs: G.711 (µ-law, A-law), G.722, G.726 (16/24/33/40 kbps), G.729, DVI4 (narrow/HD/Ultra HD), iLBC, Speex (narrow/HD/Ultra HD), SILK, L16 PCM (narrow/HD/Ultra HD/CD) Data Flash Management (Multi-Partition File System, Advanced Firmware Management, Bootloader, Secure Area) SIP Fusion TCP/IPv4/IPv6 DHCP Client Extensible Command Line Interface Web Server with Extensible Configuration Web App and Configuration and Control Web Services VoIP Call Manager (includes support for in...

    $3061 (Avg Bid)
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    5 ofertas

    ...IOS iPhone and android mobile. Mobile Dialer FEATURES 1-User friendly graphical interface 2- Advanced Phone Book (Contact manager with name number,profile pic and searching ) and Integration with Mobile Phonebook with option to show the mobile phonebook or no 3-login with ext number and password 4-Call Transfer, Call Hold, record calls 5-Connect through 3G, 4G, GPRS and Wi-Fi 6-G729, GSM, iLBC, Speex, G711, G722, AMR Codec Support 7- Making and receiving calls 8-Call Log 9- Call Status Indicator 10-Registration Status Indicator 11- Silence Suppression 12- noise and echo Cancellation 13-Make call with extremely low bandwidth situation 14-Voice quality is calibrated with CNG (Comfort Noise Generation), VAD (Voice Activity Detection) and PLC (Packet Loss Concealment) to ensure cr...

    $670 (Avg Bid)
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    7 ofertas

    This project is only for those freelancer that has existing similar app to amend for us. Do you hav...Excellent audio quality, even on older devices - Supports calling over 2G, 3G, 4G and WIFI - Multiprotocol with SIP and IAX support, compatible with all RFC compliant PBXs - Background / multitasking support - Native dialer integration - Integration with the native android contact list - Speakerphone mute and hold - UDP and TCP transports (use TCP for better battery life!) - Supports g711 (ulaw, alaw), speex, iLBC and gsm codecs - Supports sending of DTMF - DNS SRV - Built-in echo cancellation - STUN support - Change ringtone per account - call waiting - Video support (gold users only) - ZRTP / TLS support (gold users only) - Call Transfer (gold users only) - Wideband audio (gold u...

    $2579 (Avg Bid)
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    16 ofertas

    i want a video and audio player for smartphones and tablets with hardware acceleration support, to watch your videos online and offline anywhere and listen to your music. ★ Play Videos. Supports most containers (aac, avi, asf, divx, flv, m3u8, mkv, mov, mp3, mp4, mpg, mts, ogg, rm, rmvb, ts, wmv), formats (aac, ac3, amr, divx , h263, h264, hevc, mpeg2, mpeg4, speex, vp9, wmv) and protocols (http, https, mms, rtmp, rtsp). ★ Video lists. You can upload video lists in .w3u format directly from file, through links the list can get them in the forum. ★ TV. Send content compatible with Dragons Feel to your TV using Chromecast or DLNA devices, without having to worry about the video format and without having to use the hdmi would be more remote. ★ PARENTAL CONTROL: Use the parental contr...

    $554 (Avg Bid)
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    14 ofertas

    We need to developed a better echo cancellation for our PJSIP base applications on PC which currently using speex echo cancellation which is not working very well. We need to hire a person who have experience on write AEC for us or improve on speex AEC so it will work well on PC applications.

    $4240 (Avg Bid)
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    6 ofertas

    HI There, I am looking help for Magento 2 Speed Optimization, code level changes required in Theme/modules, HTACCESS level optimization will be not enough and looking only magneto experience gigs. Thanks

    $325 (Avg Bid)
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    42 ofertas

    Call functions like mute, conference, hold, transfer, and call recording should be available. These options should be visible during a call. When you transfer a call, it will ask for your permission to move the call to another person first. Custom ringtones Echo cancellation an...another person first. Custom ringtones Echo cancellation and noise suppression Pop-up notifications Presence/online status Call history Voicemail messages and missed calls notifications Contact integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 RAW SIP technology, not WebRTC. Similar Projects: Zoiper SI...

    $529 (Avg Bid)
    $529 Média
    7 ofertas

    Call functions like mute, conference, hold, transfer, and call recording should be available. These options should be visible during a call. When you transfer a call, it will ask for your permission to move the call to another person first. Custom ringtones Echo cancellation and noise suppression Pop-up notific...another person first. Custom ringtones Echo cancellation and noise suppression Pop-up notifications Presence/online status Call history Voicemail messages and missed calls notifications Contact integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SI...

    $552 (Avg Bid)
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    12 ofertas

    We require a predictive dialer module to be built for FreePBX 13 Asterisk PABX. This will need to have full predictive dialer capability and monitoring. For an outbound call centre with 200 agents. We have the following standard codecs: • Alaw, • Ulaw, • G729A, • Opus, • Speex, • GSM, • G723

    $1324 (Avg Bid)
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    15 ofertas

    ...altitude, distance and current location (similar option can be found in Vidometer app by hitcase). 4. Streaming compatibility inputs with Wowza Streaming requirements, for Amazon cloudfront RTMP Video: H.264, VP8, VP6, Sorenson Spark®, Screen Video v1 & v2 Audio: AAC, AAC-LC, HE-AAC+ v1 & v2, MP3, Speex, Opus, Vorbis ...

    $762 (Avg Bid)
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    33 ofertas
    Write some Software Encerrado left

    - We need a simple android application which could send voice to RTMP server - The application will read the audio from mic and transmit it to the server in real time - The application must encode the audio in real time using speex codec - The server’s credentials would be provided after the project is awarded. - Keep in mind the most important thing is to keep the latency low, and make sure it can run fine even on low bandwith internets.

    $547 (Avg Bid)
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    9 ofertas

    We are looking for experienced developer/development team with prior experience in developing SIP audio/video clients. Features required: - TCP or UDP SIP transport - Codecs: G.722, G.711u, G729, Speex - Remote phonebook synchronization - Remote provisioning - Branding - VoiceMail and VoiceMail Message Waiting indication - DTMF modes - STUN support for NAT traversal - SRTP audio and TLS encryption -openvpn encryption

    $193 (Avg Bid)
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    9 ofertas

    ...User shall be able to check the his commission on the web and receive an sms once every day with his daily made commission and accumulated balance in total - We shall be able to execute any USSD string and read its answer remotely by IP - We shall be able to send SMS and read any answer remotely by IP - Compression used to send to smartphone has to be ultra low bandwith consuming, perhaps using Speex - Signalling of SIP has to be matched to GSM on the phone - The user shall be able to see whether WiFi can be used for making calls (maybe ports are blocked or phone is on WiFi, but still needs to login to the WiFi internet page to use internet) - We shall be able to turn on and turn off caller ID to be shown to called party - We shall be able to change IMEI number on the phon...

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    15 ofertas

    - Integrating web service APIs via JSON/XML . (Amazon Echo, Google Home, IP cameras, Sonos) - Building and Creating Restful Web service API documentation with JSON/XML - Improving Authentication flow. - Being well versed with Network protocols such as HTTP and TCP/IP Skills below will be a plus but not required - video/audio transmission protoco...JSON/XML . (Amazon Echo, Google Home, IP cameras, Sonos) - Building and Creating Restful Web service API documentation with JSON/XML - Improving Authentication flow. - Being well versed with Network protocols such as HTTP and TCP/IP Skills below will be a plus but not required - video/audio transmission protocols(rtp, rtsp, http, rtmp, sip,…) - video/audio codecs(vp9, vp8, h264, aac, speex, opus, ..) - Lan...

    $54 / hr (Avg Bid)
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    19 ofertas
    Write some Software Encerrado left

    Need an android ndk library (to be run on android devices) to decode audio files that are encoded using Opus or Speex audio codec (Opus preferred) and encrypted using a stream cipher (aes preferred). Provided a file pointer and a decryption key, the library should be able to: 1) Decrypt and decode the contents at the file pointer in chunks. 2) Support decrypting+decoding files that are partially downloaded, while they are being downloaded. 3) Support seeking to a given timestamp lying within the downloaded part of the file.

    $18 - $152
    $18 - $152
    0 ofertas

    Need an android ndk library (to be run on android devices) to decode audio files that are encrypted using Speex codec and encrypted using a stream cipher (aes preferred). Provided a file pointer and a decryption key, the library should be able to: 1) Decrypt and decode the contents at the file pointer in chunks. 2) Support decrypting+decoding files that are partially downloaded, while they are being downloaded. 3) Support seeking to a given timestamp lying within the downloaded part of the file.

    $148 (Avg Bid)
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    2 ofertas

    Hi kushal002, I noticed your profile and would like to offer you my project. We need firmware for our new product - audio player based based on STM32L151. Firmware content: SPEEX audio decoder, USB MSC, FATfs, touch sensing, LED RGB PWM control. Are you interested in? What price and time?

    $150 (Avg Bid)
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    1 ofertas

    We provide: - schematics under Altium Designer - 3D STEP of PCB (localization of key parts) - description of functionality - final prototype (for firmware developing/testing) We need (contractor provides): - PCB 3D under Altium Designer - firmware (source code) under GCC/ECLIPSE Payments: a) 20% after verification of PCB design b) 80% after verification of firmware at final prototype

    $588 (Avg Bid)
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    5 ofertas

    ...on the home screen (list users, channels, text messaging, P2P users The app should have volume control for microphone and speaker linkt to the device Vol Up and Down Button. The app should be connected to the hardware PTT button on the side of the device. the app should have a "settings" menu te configure options, serveradres, username and pasword to connect to the server The app should use Speex or Opus voice codec and must be able to run when the screen is OFF. For the GPS part inside the app, the option must exist inside the app to put the GPS on or off. When ON, the app needs to push its location to the GPS-Gate server, connection to that server is also based on a username and password (may be the same as the connection to the voice server). We prefer the con...

    $3967 (Avg Bid)
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    27 ofertas

    ...Multi-user chats File transfer SECURITY Encrypted password storage Password protection with a master password Encrypted Instant Messaging with Off-the-Record Messaging (OTRv4) Chat authentication with the Socialist Millionaire Protocol over OTR Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery via DHCP and mDNS (Bonjour) IPv6 fully supported by SIP and XMPP Call history Missed call notifications Support for Google Contacts On-line contact lis...

    $3079 (Avg Bid)
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    50 ofertas

    -- Problem: Adobe Air provides raw audio samples only, we need to apply an audio codec and transmit ourselves as we are bound to use our own UDP sockets. -- Solution: We provide you with the sound samples, you handle the codec stuff. We prefer to use OGG, but Speex, Vorbis, AAC or even MP3 are fine too. -- Requirements: No separate application, we need to implement this in OUR source

    $351 (Avg Bid)
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    3 ofertas

    We are looking to integrate PortSIP SDK into our mobile app. Phase I 1. 1 to 1 Audio Call Support G.729, Opus, AMR, AMR-WB, G.722.1, G.711a/u, GSM, iLBC, SPEEX, SPEEWX-WB codecs 2. 1 to 1 Video Call Support H.263, H.263+, H.264 and VP8 video call 3. Call Transfer 4. Call Hold 5. Call Waiting 6. Call History 7. Presence Phase II 8. Audio Call Conference 9. Video Call Conference 10. DTMF

    $2380 (Avg Bid)
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    16 ofertas

    I need a very basic implementation of the opus codec. ( formerly speex ) for Android. This will be a stand alone app that can communicate with other users using PTT ( push to talk ) and continuous telephone-like conversation. Basic Function. 1. Register User 2. Find User 3. Speak to user via PTT or Continuous. 4. Add user to group so one can address many users at once. This app will eventually be integrated into another app of mine currently under developments

    $721 (Avg Bid)
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    14 ofertas

    Take the following open source Android SDK and convert to iOS: Basically you are just porting the Android SDK to Objective C Here are the iOS SDK Requirements: -Speech Synthesis Request -Viseme request -Word Timing Markers -Speech recognition -Freeform -Commands Record Audio Speex Encoding Display optional popup dialog

    $734 (Avg Bid)
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    11 ofertas

    Hi I am looking for a programmer who can do integrating speex codec to make voice recording & playing in Cygwin environment/ linux environment . C program should record voice for 2 sec ( Raw PCM samples of 8Khz sampling rate ) , encode the voice , decode the voice & play it back using speex encoder & echo cancellation functionality . You can download speex encoder from --venu

    $222 (Avg Bid)
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    1 ofertas

    h264 flash video encoder with speex audio - wowza transcoder to acc Need a live flash RTMP- H264 /speex encoder to be able to Stream on rtmp to Wowza ( using transcoder to transcode from speex -to aac , able to stream on other formats ( HLS) Full source required, minimum latency for the flash encoder. A wowza server is already setup for testing . Wowza transcoder need to be setup for transcoding from rtmp H264/Speex - H264/aac . Please send similar references , dealy ASAP ...

    $30 - $250
    $30 - $250
    0 ofertas
    VOIP Dialer Encerrado left

    SIP Dialer for Android, IOS, Windows 1. Voice call 2. Message ...12. Shows balance on the top of the dialer 13. Capable of saving 5 caller IDs and can use any by manual selecting 14. Echo cancellation (with various backends : webRTC, speex, simple) 15. Conference calls up to 4 16. Callback Option (On selecting this option dialer will call on both registered number and destination number) 17. Call transfer option(example: if I want ). 18. Local Access dial (Call on local access number using mobile phone dialer not through internet) 19. Invite friends option(this option will send web link on invited persons mobile as text through server) *** Codecs: pcmu/a (aka g711u/a); speex; g722; gsm; iSAC; SILK; G729; AMR (depending on device) and as extra plugin : OPUS; g726;...

    $3083 (Avg Bid)
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    8 ofertas

    To develop a mobile phone softphone app with specifications as below    Technical Requirements: The application should support the following codecs: G.722.1, G.723.1, G.726, G.728, G.711, G.729, G.723.1, iLBC,: G.711 (A and m-law),G.729A/B/D/E, AMR, GSM 6.10/EFR, iLBC, Speex · Voice: Real time Quality Monitoring (MOS) 3 independent phone lines Auto-Answer/Do Not Disturb Call Forwarding Full Duplex Audio Recording compress the audio data. · Support Video call and Video on demand and online TV  using H264 codec · Fully SIP compatible soft phone/dialer for PC2Phone and Mobile2Phone (iPhone, Android) application.  · Compatible with Windows 2000/XP/2003/Vista/Mac · NAT/Firewall traversal feature · ...

    $555 (Avg Bid)
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    1 ofertas

    To develop a mobile phone softphone app with specifications as below    Technical Requirements: The application should support the following codecs: G.722.1, G.723.1, G.726, G.728, G.711, G.729, G.723.1, iLBC,: G.711 (A and m-law),G.729A/B/D/E, AMR, GSM 6.10/EFR, iLBC, Speex · Voice: Real time Quality Monitoring (MOS) 3 independent phone lines Auto-Answer/Do Not Disturb Call Forwarding Full Duplex Audio Recording compress the audio data. · Support Video call and Video on demand and online TV  using H264 codec · Fully SIP compatible soft phone/dialer for PC2Phone and Mobile2Phone (iPhone, Android) application.  · Compatible with Windows 2000/XP/2003/Vista/Mac · NAT/Firewall traversal feature · ...

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    15 ofertas

    ...conferencing, with ability to chat, share files, change Image or Avatars 4 Would have control via Admin side to modify the BITRATE perhaps 5 levels of control, with ability to have Video chat resolutions of: 1024 x 576 pixels @ 1500 kbps 11MB/minute 1280x 720 pixels @ 2500 kbps 19MB/minute 1920x1080 pixels @ 4000 kbps 30MB/minute 2160p 4K 4320p 8K 5 High Defintion sound: G.722, Opus, Speex and other Codecs use involved in producing high quality sound 6 Ability to create Multiple rooms and multiple viewers who can video chat with a main moderator, and viewers are NOT able to see each other within the SAME ROOM 7 Ability to work with , WebRTC, SIP and XMPP protocols if doable and feasible 8 Ability to integrate seamlessly with a Payment Gateway. Choices are

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    9 ofertas
    IM Messenger Encerrado left

    ...Detailed user profiles (Client/Server) VB.NET 4.5 Desktop Application SQL Server Database (Operating Systems) Client: Microsoft Windows XP,7,8 and letter Server: Microsoft Windows Server 2003,2008, 2012 and letter Technology’s: (You can give us your ideas to implement a very good quality) -Low bitrate High resolution ffmpeg Video encoding/decoding -Low bitrate very clear speex voice encoding/decoding -Secure and fast WCF Duplex service ...

    $4242 (Avg Bid)
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    8 ofertas
    Python PyAudio Speex Encerrado left

    I have a python script that requires pyaudio and can not get it working on my ubuntu 14.04 box or my raspberry pi and want to get away from pyaudio and use the speex python modules

    $263 (Avg Bid)
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    4 ofertas

    Need to create a application using ffmpeg libraries which can merge multiple live streams and output single stream. Basically code will do equivalent work of using ffmpeg tool with complex filter example -filter_complex "nullsrc=size=640x480 [base]; [0:v] setpts=PTS-STARTPTS, scale=320x480 [left]; [1:v] setpts=PTS-STARTPTS, scale=320x480 [right]; [base][left] overlay...overlay=shortest=0:x=320:y=0" The code should be able to handle adding and removing of input streams at runtime. ie for example if at start only one stream stream is active simply output that stream's data to a diff endpoint. Now if a new stream comes online app should be able to merge the two streams and use 50% of output video area for each stream and so on. Audio codecs support needed - SPEEX ...

    $13 / hr (Avg Bid)
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    1 ofertas

    - For iPhone 5,5s and 6 - Dial list of fixed numbers. - Include codecs G.711 (uLaw &ALaw) - SILK - SPEEX - I will provide my SIP server configuration for testing. - Using PJSIP Lib last version. - I will extend the project then to Android version. For more info: For more info.:

    $724 (Avg Bid)
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    Risk Risk is a java version of the classic Risk board game,with a simple map format,network play single player mode,hotseat,mapedit...standard operating environments and cloud services as well because of its royalty-free pricing and excellent cross platform support industry standard up to the current TLS 12 mediastreamer mediastreamer is a portable c library that allows you to create and run audio and video is designed for any kind of voice over IP features RTP connectivity,audio codecs(speex,ILBC,GSM,and AMR),video codes(MPG4,H263,H264,and Theora),sound card I/O,wav file streaming,webcam videocapture,echo-cancellation,conferencing,parametric equalization,and various other media streaming compontent of linephone,a GPL SIP video phone.

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