Skip to content

Allow to configure frame rate and audio duration #261

New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Open
LouisAE wants to merge 1 commit into
base: main
Choose a base branch
from
Open

Allow to configure frame rate and audio duration #261

Show file tree
Hide file tree
Changes from all commits
Commits
Show all changes
1 commit
Select commit
File filter

Filter by extension

Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
33 changes: 33 additions & 0 deletions src/peer_connection.c
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -165,6 +165,34 @@ PeerConnection* peer_connection_create(PeerConfiguration* config) {
memset(&pc->sctp, 0, sizeof(pc->sctp));

if (pc->config.audio_codec) {

switch(pc->config.audio_codec)
{
case CODEC_PCMA:
case CODEC_PCMU:
if (pc->config.audio_duration)
{
pc->artp_encoder.timestamp_increment = pc->config.audio_duration * 8000 / 1000;
}
else
{
pc->artp_encoder.timestamp_increment = CONFIG_AUDIO_DURATION * 8000 / 1000;
}
break;
case CODEC_OPUS:
if (pc->config.audio_duration)
{
pc->artp_encoder.timestamp_increment = pc->config.audio_duration * 48000 / 1000;
}
else
{
pc->artp_encoder.timestamp_increment = CONFIG_AUDIO_DURATION * 48000 / 1000;
}
break;
default:
break;
}

rtp_encoder_init(&pc->artp_encoder, pc->config.audio_codec,
peer_connection_outgoing_rtp_packet, (void*)pc);

Expand All @@ -173,6 +201,11 @@ PeerConnection* peer_connection_create(PeerConfiguration* config) {
}

if (pc->config.video_codec) {
if (pc->config.video_frame_rate)
{
pc->vrtp_encoder.timestamp_increment = 90000 / (pc->config.video_frame_rate);
}

rtp_encoder_init(&pc->vrtp_encoder, pc->config.video_codec,
peer_connection_outgoing_rtp_packet, (void*)pc);

Expand Down
2 changes: 2 additions & 0 deletions src/peer_connection.h
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -75,6 +75,8 @@ typedef struct PeerConfiguration {
MediaCodec audio_codec;
MediaCodec video_codec;
DataChannelType datachannel;
uint32_t video_frame_rate;
uint16_t audio_duration;

void (*onaudiotrack)(uint8_t* data, size_t size, void* userdata);
void (*onvideotrack)(uint8_t* data, size_t size, void* userdata);
Expand Down
4 changes: 0 additions & 4 deletions src/rtp.c
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -190,25 +190,21 @@ void rtp_encoder_init(RtpEncoder* rtp_encoder, MediaCodec codec, RtpOnPacket on_
case CODEC_H264:
rtp_encoder->type = PT_H264;
rtp_encoder->ssrc = SSRC_H264;
rtp_encoder->timestamp_increment = 90000 / 30; // 30 FPS.
rtp_encoder->encode_func = rtp_encoder_encode_h264;
break;
case CODEC_PCMA:
rtp_encoder->type = PT_PCMA;
rtp_encoder->ssrc = SSRC_PCMA;
rtp_encoder->timestamp_increment = CONFIG_AUDIO_DURATION * 8000 / 1000;
rtp_encoder->encode_func = rtp_encoder_encode_generic;
break;
case CODEC_PCMU:
rtp_encoder->type = PT_PCMU;
rtp_encoder->ssrc = SSRC_PCMU;
rtp_encoder->timestamp_increment = CONFIG_AUDIO_DURATION * 8000 / 1000;
rtp_encoder->encode_func = rtp_encoder_encode_generic;
break;
case CODEC_OPUS:
rtp_encoder->type = PT_OPUS;
rtp_encoder->ssrc = SSRC_OPUS;
rtp_encoder->timestamp_increment = CONFIG_AUDIO_DURATION * 48000 / 1000;
rtp_encoder->encode_func = rtp_encoder_encode_generic;
break;
default:
Expand Down