similar to: Is it possible to change the quality with encoding ?

The link is http://www.adobe.com/devnet/rtmp/. TC Message stands for TinCan message. It is 11 bytes long, first byte is message type, three bytes of payload length four bytes of timestamp and three bytes of stream ID. The first byte of the payload for audio message is the format byte and the rest of the byte is the payload. Jozsef ----- Original Message ---- From: Jeff Ramin <jeff.ramin

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

FLV contains TC messages? TC message payload contains a format byte and speex frames (up to eight). In the format byte 0xb0 indicates speex. Speex is always 16 kHz, 16 bit, mono. Jozsef Message: 1 Date: Mon, 16 Nov 2009 14:40:20 -0600 From: Jeff Ramin <jeff.ramin at singlewire.com> Subject: [Speex-dev] jspeex question To: speex-dev at xiph.org Message-ID: <4B01B8B4.8020904 at

First of all, Flash Player can only publish Speex at 16 kHz. 20 ms of audio (320 samples) will result in compressed payload size of 106 bytes (42.4 kbps). When Flash Player sends a TC message, there is 11 bytes TC message header and a single byte of audio message header. For more information, please see ActionScript 3 reference http://help.adobe.com/en_US/AS3LCR/Flash_10.0/ Jozsef

Hi Jean-Marc I am using speex in my voip app (wideband mode). I have noticed that when VAD kicks in, CPU considerably increases (bitrate decreases to 4 kbps). It takes about 7 ms to encode one frame versus 0.5 ms (for speech). This happens regardless preprocessor is used. Although the issue seems to be driver dependent, it occurs on XP, Vista and Mac OS X. Besides complexity and quality, this

Flash player encodes speex at 16 kHz, mono, 16 bit. Fields in the format byte should be ignored if the format is speex. You can set the quality by Microphone.encodeQuality (default 6). You can also set the number of speex frames per tc message using Microphone.framesPerPacket. Flash player can only decode speex at 16 kHz, so make sure you have the proper sample rate. Jozsef > > Message:

Hi Jozsef, this approach sounds interesting. Do have you have some source code available ? Thanks Mark -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of Jozsef Vass Sent: Friday, September 25, 2009 11:43 PM To: mark_schilling at gmx.de Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] High CPU usage I have run into the same

What are you trying to accomplish? My code sample was about how Flash Player microphone input in speex. If you want to do decode, please see speexdec.c Jozsef -----Original Message----- From: Max Lapshin [mailto:max.lapshin at gmail.com] Sent: Tuesday, March 30, 2010 8:53 AM To: Jozsef Vass Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] Speex in flash player: how to work with? On Sat, Mar

I have run into the same issue. Before sending a frame to encoder, I calculate the energy. If it is less than a small threshold, I simply replace this frame with "silence frame," which is 320 random samples of values smaller than 3 (16 kHz). BTW, I have only experience this problem with certain USB headsets that provide you all 0 samples when muted. Jozsef

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in

Since r12660, the speex_jitter_get with high latency doesn?t works, I have no sound. Before this release, the speex_jitter_get works in all conditions. speex_jitter_get return void, then I cannot know the reason of this problem. Regards Ouss -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 23:07 To: Ouss Cc:

What is the best way to pick a prebuffering length for a streaming audio application using UDP transport? I'm using Speex in a VoIP application with RTP transport, currently with a fixed 500ms prebuffer on the playback side. However, I'd like something a bit more adaptive to accomodate high-jitter connections. For example, in one test configuration there is a very low average

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

I use the speex version of your jitter, and in speex_jitter_get, you always call the jitter_buffer_update_delay. -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 13:06 To: Ouss Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Problem with the svn jitter buffer > I think that the new Jitter Buffer have a problem. > >

Ok, this is a silly question, but what does the jitter buffer do? I'm really new to audio, so please bear with me. From what I gather (primarily from the list archive), the jitter buffer is a wrapper around the Speex decoder. I give it the packets I receive, in whatever order I receive them, and then it gives me back a clean stream of audio samples. But what I don't entirely

Hi, Our project is using the jitter buffer feature built in Speex. We noticed there are some latency when using the jitter buffer. Does anyone know what is the "worst case" latency inherent in the jitter buffer algorithm? I believe someone already mentioned that it's adaptive but is there a worst case hard number (in terms of 20ms Speex frames)? I'm not familiar with the

Is is possible to give a short hint about how the jitter buffer would "catch up" when network condition have been bad and then get better? I'm using the jitter buffer with success now, but sometimes I have a long delay that's caused by bad network conditions and then later when the conditions get better, I would think we would want the audio to gradually catch up with real-time

Hi, I'd like know about anyone using the current jitter buffer in Speex. I'm planning on changing it to make it more general and I'd like some feedback about how to make it better. Also, let me know if you're doing anything serious with it and want to make sure I don't break your stuff. Basically, I want to make the jitter buffer easier to use with other codecs and reduce the

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

I am trying to trouble shoot one of my ISP's network and compare to my other ISPs offering. Although network 1 is reasonably fast and has low enough latency, voice quality is not good and the reason for this is not readily apparent using standard network tools. What tools can be used to assess the quality of the network in terms of it's suitability for voice? I am using ping, mtr,