Update WebRTC transceivers dynamically

[Busayo Famutimi]

Thanks Matthew.

Got it working.

On Sun, 20 Feb 2022, 13:22 Matthew Waters, <ystreet00 at gmail.com> wrote:

> Nothing has to be unlinked to not send data.  It is entirely possible to
> have things linked and not data flowing.  You can drop data manually with a
> pad probe, or you could change the element states to not have them
> producing data.  It all depends on what your exact requirements and
> elements involved are.
>
> Cheers
> -Matt
>
> On 18/2/22 20:37, Busayo Famutimi wrote:
>
> Thanks Matthew.
>
> Just to understand clearly, I have the following in a pipeline,
>
> "wasapisrc role=comms use-audioclient3=true low-latency=true ! queue !
> audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay !
> queue name=audio_queue !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=111 ! webrtcbin ".
>
> To mute, the queue (named audio_queue) src pad and the webrtcbin sink_pad
> (sink_%u) have to be unlinked and
>
> to unmute, both pads must be linked back together.
>
> Is my understanding correct?
>
> Thanks.
>
>
>
>
> On Fri, Feb 18, 2022 at 2:53 AM Matthew Waters <ystreet00 at gmail.com>
> wrote:
>
>> Setting the sendrecv/recvonly/sendonly does not change what data you push
>> into webrtcbin.  If you want 'mute' functionality, stop pushing audio/video
>> into webrtcbin.
>>
>> Cheers
>> -Matt
>>
>> On 18/2/22 05:15, Busayo Famutimi via gstreamer-devel wrote:
>>
>> Hello.
>>
>> Please I have a working webrtc peer connection using. However, I am
>> unable to set the transceivers (both audio and video) dynamically.
>>
>> I have the following code:
>>
>> GArray *transceivers;
>>     g_signal_emit_by_name(voip_data->webrtc, "get-transceivers", &
>> transceivers);
>>     g_assert(transceivers != NULL && transceivers->len > 1);
>>     for (guint i = 0; i < transceivers->len; i++)
>>     {
>>         GstWebRTCRTPTransceiver *transceiver = g_array_index(transceivers,
>> GstWebRTCRTPTransceiver *, i);
>>         GstWebRTCKind kind;
>>         g_object_get(transceiver, "kind", &kind, NULL);
>>         if (kind == GST_WEBRTC_KIND_AUDIO)
>>         {
>>             if (mic_action == MUTE_MIC)
>>                 g_object_set(transceiver, "direction",
>> GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, NULL);
>>             else if (mic_action == UNMUTE_MIC)
>>                 g_object_set(transceiver, "direction",
>> GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, NULL);
>>             else
>>                 g_assert_not_reached();
>>         }
>>     }
>>     g_array_unref(transceivers);
>>
>> but the other peer still receives audio.
>>
>> I am trying to create the mute/unmute microphone functionality.
>>
>> How can this be achieved?
>>
>> Thanks.
>>
>>
>>
>
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