The programing API for opus only uses 48khz signed linear audio. We add the codec to the system with that as the native rate and it translates for us. I don't know what the cost is really. Its not really useful to use if you plan to bridge to the PSTN anyway but it will work fine in FS at probably some noticeable cpu usage. On Mon, Aug 5, 2013 at 9:59 AM, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Is there any info how much bandwidth is required for 48kHz audio? How CPU hungry is downsampling to 8kHz G711? According to opus-codec.org, opus can use sampling rates from 8kHz, not just 48kHz. In cases when the call ends in PSTN, won't be appropriate to support 8kHz sampling rate in FS too?
On Thu, Aug 1, 2013 at 9:18 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Opus always uses 48khz On Aug 1, 2013 2:15 PM, "Dimosthenis Arkoudeas" < dimosthenis@arkoudeas.com> wrote:
Hi all,****
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Playing with mod_opus came up with the following.****
I am trying to recompile the codec with 8Khz and 40ms ptime. I am not familiar with the codec internals and not really sure what parameters should I change…****
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In mod_opus.c :****
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Line 59:****
int bitrate_bps = codec->implementation->bits_per_second;****
int use_vbr = 1;****
int complexity = 10;****
int use_inbandfec = 1;****
int use_dtx = 1;****
int bandwidth = OPUS_BANDWIDTH_FULLBAND;****
int err;****
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I suppose that the bandwidth must be changed to OPUS_BANDWIDTH_NARROWBAND and use_vbr=0 and I haven’t found yet whether the FEC parameter is passed to opus encoder.****
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And in Line 184 , I think the following parameters must be changed****
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switch_codec_interface_t *codec_interface;****
int samples = 480;****
int bytes = 960;****
int mss = 10000;****
int x = 0;****
int rate = 48000;****
int bits = 32000;****
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Can you propose any values about these parameters to achieve 8khz with 40ms packet time? ****
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Thanks for any help,****
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Dimosthenis****
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