transcoding

Hello, the IDecodedFrameInterceptor works only for transcoded frames. Is there a way to capture frames from all streams regardless if the frame has been transcoded or not? Thanks, P

Hello, we are using Android SDK 1.1.0.57 and are experiencing a problem where WCS 5.2.1279 wouldn't republish video with RTMP - only audio gets restreamed and with a significant delay - around 15 seconds. We also tried doing this using WebSDK - works fine. Thanks, P

Howdy, I was curious if WCS can be used for recorded files transcoding. For example, on some Android devices, files are stored as VP8/webm. We would like to transcode those files to H264/AAC. Is this possible with WCS or should we go with ffmpeg? Thanks, P

Hi Max, We are facing a sudden error of TRANSCODER_RESULT_NULL from Flashphoner side during room creation which was working fine earlier. on server log I found the following errors - 10:36:16,548 ERROR 01-987d-af6d21c8eeeb -...

Is there a way to crop video streams that are sent to Flashphoner? What I want to do is screencast a window from the remote machine with webrtc and crop a certain area of that window (so top/right/left/bottom coordinates are known) before it's restreamed by flashphoner. Thank you!

Hello, We implemented VP8 stream transcoding, but faced another problem: A new session for transcoding starts with the parameters that we specified in the API request (480x640), but in fact, when recording, we get 640x640 resolutions. When examining the logs, we found that at some point the...

Добрый день. Скажите, если в нашей схеме есть origin + edge на edge сервере используются параметры cdn_point_of_entry = ORIGIN_SRV_IP cdn_origin_allowed_to_transcode=true force_client_requested_video_resolution = false disable_streaming_proxy = false...

Здравствуйте. Скажите пожалуйста, есть ли возможность реализации инструмента "Лупа" средствами вашего ПО. В случае, если нет, можете ли предложить варианты реализации такого функционала? Возможно ли это в принципе?

The first problem is; When i issue this command : "ssh ip@localhost -p 2001" in aws ec2 hourly license, a password is required but these passwords did not work: admin,demo,password The other problem is; even though im using amazon ec2 10 gb/s server , it freezes on wlan networks. It's ok on...

Hello, we are building a solution that uses Mixer heavily (Audio only). Part of our requirement is to change properties of incoming streams to the mixer. Stream1, Stream2, Stream3 are fed to Mixer1, Stream 4, Stream 5 are fed to Mixer2. So output of mixer2 will be Stream1, Stream2, Stream3...

Dear Max, We are running well with 720p -> Input RTMP and Output is HLS edge. On this scenario, if we watch WEBRTC stream from source server (rtmp input) and output HLS edge, the quality is the same. If we attempt to move to 1080p, including this on cdn_profiles, we have GOOD quality on...

Hi ! I have a wierd thing happening. I publish with OBS WebRTC to my AWS Server. On top of that there is a WebRTC 1 way video and 2 way audio that I publish from my computer at the office. When There is more than 3 peoples at the office watching the stream via their browser in WebRTC there is...

Hi Team, We are using Flashphoner for last several months and one common complaint from our clients is the video streaming does not work with low connectivity. Is there anything we can do to improve this experience. What are the network considerations or minimum bandwidth required to have smooth...

Hi, I m publishing the streams to origin servers and playing using HLS by edge servers. When a first user tries to view the stream, the stream can't be played in player and buffer gets stalled out. If it is played again, then streams gets played. Based on this link...

Hello Max, We are confused about HLS. How to define 720p quality to HLS on edge? We start the HLS by API, but how to call an specific transcoder to them?

Hello! We are streaming from VMIX to Youtube and FlashPhoner. On Youtube we got SYNC audio x video. On our structure, we got Audio and Video after some time, lost the sync and do not recover. We have an ORIGIN server and EDGE on Google Cloud. We can see the lost of SYNC on WebRTC and M3MU...

My source is an RTSP stream with stereo AAC audio. I want to achieve high quality stereo audio encoding in the direction of server to browser. The AAC and Opus bitrates are configurable in flashphoner.properties, but it's still mono. How can I set the Opus and AAC encoders to stereo? And it...

Hi, 1. When and what type of GPU transcoding will be available since we are looking for the streaming server that can use large scale GPU transcoding such as Nvidia CUDA 2. Do you have any plan to allow custom transcoder via native java interface Thanks and Best Regards,

Добрый день! У нас задача объединить два webrtc стрима. На входе у нас есть видео+аудио от одного спикера и аудио от другого. На выходе хотим получить два видео, одно с оригинальным звуком, второе тоже с оригинальным звуком, плюс звук от второго (в таком случае оригинальная звуковая дорожка...

I am using screen sharing feature of flashphoner and after 20 sharing screen it has increased my server load. There is no other application running on this server only flashphone is installed. kindly provide the solution. Server is 4 core and 16gb ram 8gb is allocated as you advised but load...