Hi Tobias,
There seems to be an easier workaround: i.e. to connect the audio to a
sink during the early media instead. It can use null audio device or
mute it if it doesn't want to play it at first.
The patches seem to assume that: - the decoding is not connected
during the silence frames, or - when we are sending silence frames,
the remote are sending ones as well, which is not always true.
Regards,
Ming
On Mon, Jul 23, 2018 at 11:01 PM, Tobias Schneider
<tobias.schneider@xxxxxxxxxxxxxxxxxxxx> wrote:
> This is variant 2 to fix that issue by directly calling
> pjmedia_port_get_frame in stream.c after packet was sent via RTP to
> ensure that there are "snychronous" calls of put- and get-frame.
>
> ---
> pjmedia/src/pjmedia/stream.c | 14 ++++++++++++++
> 1 file changed, 14 insertions(+)
>
> diff --git a/pjmedia/src/pjmedia/stream.c b/pjmedia/src/pjmedia/stream.c
> index adcaa628..1f2d0457 100644
> --- a/pjmedia/src/pjmedia/stream.c
> +++ b/pjmedia/src/pjmedia/stream.c
> @@ -240,6 +240,12 @@ static const char digitmap[17] = { '0', '1', '2', '3',
> /* Zero audio frame samples */
> static pj_int16_t zero_frame[2 * 30 * 16000 / 1000];
>
> +/* null buffer to hold result of get_frame after transmit of NULL frame.
> + * Make it big enough to hold any kind of media port size connected to conf
> + * bridge, e.g. up to 30ms ptime and 16kHz samplerate
> + */
> +static pj_int16_t null_buffer[2 * 30 * 16000 / 1000];
> +
> /*
> * Print error.
> */
> @@ -1318,6 +1324,14 @@ static pj_status_t put_frame_imp( pjmedia_port *port,
> (const void**)&rtphdr,
> &rtphdrlen);
>
> + /* directly call get_frame of stream to avoid filling of JB during
> + * silence period */
> + pjmedia_frame null_frame;
> + pj_bzero(&null_frame, sizeof(null_frame));
> + null_frame.buf = null_buffer;
> + null_frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
> + null_frame.size = stream->enc_samples_per_pkt * 2;
> + pjmedia_port_get_frame(&stream->port, &null_frame);
>
> /* Encode audio frame */
> } else if ((frame->type == PJMEDIA_FRAME_TYPE_AUDIO &&
> --
> 2.13.7
>
>
>
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