On Wed, Jun 18, 2014 at 11:03 AM, AirMike <[email protected]> wrote:
> I have task of recording webrtc local stream audio and webrtc remote > stream audio. > I succeeded in recording audio with MediaRecorder (using timeSlice) in > which I get recorder chunk Blob (audio/ogg) for local and remote audio. > > Now, before I send this to server I would like to mix recorded local and > remote audio chunks to one chunk using Web Audio API and this is where I > have some problems. > It sounds like you're using MediaRecorder to compress the local and remote audio chunks on the client, and then trying to uncompress them on the client, mix them, recompress them and send the result to the server. Is that right? If so, why are you doing the first compression step instead of just leaving them uncompressed? I used this steps: > > 1. when both local and remote audio blobs are available I'm using > FileReader to get ArrayBuffer for each > > 2. using AudioContext decodeData to get AudioBuffer (here I get error: > The buffer passed to decodeAudioData contains an unknown content type. and > The buffer passed to decodeAudioData contains invalid content which cannot > be decoded successfully.) > Is your initial compression step using timeSlice to produce multiple Blobs from a single MediaRecorder? If so, those Blobs must be concatenated to get a single resource which you can decode successfully. E.g. you can't pass just the second Blob created by a MediaRecorder to AudioContext.decodeAudioData and expect it to work. Rob -- Jtehsauts tshaei dS,o n" Wohfy Mdaon yhoaus eanuttehrotraiitny eovni le atrhtohu gthot sf oirng iyvoeu rs ihnesa.r"t sS?o Whhei csha iids teoa stiheer :p atroa lsyazye,d 'mYaonu,r "sGients uapr,e tfaokreg iyvoeunr, 'm aotr atnod sgaoy ,h o'mGee.t" uTph eann dt hwea lmka'n? gBoutt uIp waanndt wyeonut thoo mken.o w _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
