Re: [discuss-webrtc] How to modify the audio sample rate used by WebRTC (Android)?

[José Carlos Pujol]

You need to mangle the SDP to include a line with the sprop-maxcapturerate attribute [1]. Here is an example taken from there:

m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000

[1] https://tools.ietf.org/html/rfc7587

On Thu, Jan 5, 2017 at 7:49 AM, Kushtrim Pacaj <kushtr...@gmail.com> wrote:

We're making a video-conferencing device, and the hardware guys said that they optimized the mic for 16 khz sample rate, and that we should use that.

Where can I specify what sample rate WebRTC uses for calls.

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[Kushtrim Pacaj]

Thanks you. I'll try this.

On Thursday, January 5, 2017 at 8:29:35 PM UTC+1, José Carlos Pujol wrote:

You need to mangle the SDP to include a line with the sprop-maxcapturerate attribute [1]. Here is an example taken from there:

m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000

[1] https://tools.ietf.org/html/rfc7587

On Thu, Jan 5, 2017 at 7:49 AM, Kushtrim Pacaj <kushtr...@gmail.com> wrote:

We're making a video-conferencing device, and the hardware guys said that they optimized the mic for 16 khz sample rate, and that we should use that.

Where can I specify what sample rate WebRTC uses for calls.

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[Matteo C.]

Hi Jose,

do you know if the sprop-maxcapturerate sdp parameter is actually implemented in WebRTC?

Last time I checked (some months ago) I didn't find any reference of sprop-maxcapturerate in the code.

Thanks,

Matteo

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[Henrik Andreasson]

AFAIK, using SDP will not lead to a modified native input sample rate.

Is it a web based client or a native client?

What platform is it?

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[Kushtrim Pacaj]

Android

On Monday, January 9, 2017 at 3:15:21 PM UTC+1, Henrik Andreassson wrote:

AFAIK, using SDP will not lead to a modified native input sample rate.

Is it a web based client or a native client?

What platform is it?

On Mon, Jan 9, 2017 at 2:24 PM, Matteo C. <matte...@gmail.com> wrote:

Hi Jose,

do you know if the sprop-maxcapturerate sdp parameter is actually implemented in WebRTC?

Last time I checked (some months ago) I didn't find any reference of sprop-maxcapturerate in the code.

Thanks,

Matteo

On Thu, Jan 5, 2017 at 8:29 PM, José Carlos Pujol <jpu...@tokbox.com> wrote:

You need to mangle the SDP to include a line with the sprop-maxcapturerate attribute [1]. Here is an example taken from there:

m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000

[1] https://tools.ietf.org/html/rfc7587

On Thu, Jan 5, 2017 at 7:49 AM, Kushtrim Pacaj <kushtr...@gmail.com> wrote:

We're making a video-conferencing device, and the hardware guys said that they optimized the mic for 16 khz sample rate, and that we should use that.

Where can I specify what sample rate WebRTC uses for calls.

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[Henrik Andreasson]

Is it a web based client or a native client?

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[Kushtrim Pacaj]

Native Android

[José Carlos Pujol]

To my understanding, modifying the SDP will affect the parameters also for native clients. Kushtrim, please let us know if it worked for you this way

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[Matteo C.]

Yes you're right, for instance we're passing maxaveragebitrate and maxplaybackrate in the SDP and they are correctly handled in native clients.

My point is that looking at the code these two constants are defined in mediaconstants.cc and used in webrtcvoiceengine.cc, while sprop-maxcapturerate is not present at all in the code (M52 version, I didn't check the last code) so I think it's not taken into account.

Cheers,

Matteo

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[Matteo C.]

FYI I filed this bug about sprop-maxcapturerate: https://bugs.chromium.org/p/webrtc/issues/detail?id=6970

Cheers,

Matteo

[Bang Nguyen]

I'm not sure about Android, but on iOS I found out that we can change sample rate by 

RTCAudioSessionConfiguration

[Eric Davies]

In the commit I'm using, which is a few months old, its
           WebRtcAudioUtils.setDefaultSampleRateHz(value);