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Improve audio RTP header data to better match GFE

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This commit is contained in:
Cameron Gutman 2021-07-03 16:05:18 -05:00
parent e169259f6f
commit 118707f37a
1 changed files with 13 additions and 7 deletions
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20
sunshine/stream.cpp
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@ -181,7 +181,8 @@ struct session_t {
} video;
struct {
std::uint16_t frame;
std::uint16_t sequenceNumber;
std::uint32_t timestamp;
udp::endpoint peer;
} audio;
@ -706,20 +707,24 @@ void audioBroadcastThread(udp::socket &sock) {
TUPLE_2D_REF(channel_data, packet_data, *packet);
auto session = (session_t *)channel_data;
auto frame = session->audio.frame++;
auto sequenceNumber = session->audio.sequenceNumber;
auto timestamp = session->audio.timestamp;
audio_packet_t audio_packet { (audio_packet_raw_t *)malloc(sizeof(audio_packet_raw_t) + packet_data.size()) };
audio_packet->rtp.header = 0;
audio_packet->rtp.header = 0x80;
audio_packet->rtp.packetType = 97;
audio_packet->rtp.sequenceNumber = util::endian::big(frame);
audio_packet->rtp.timestamp = 0;
audio_packet->rtp.sequenceNumber = util::endian::big(sequenceNumber);
audio_packet->rtp.timestamp = util::endian::big(timestamp);
audio_packet->rtp.ssrc = 0;
session->audio.sequenceNumber++;
session->audio.timestamp += session->config.audio.packetDuration;
std::copy(std::begin(packet_data), std::end(packet_data), audio_packet->payload());
sock.send_to(asio::buffer((char *)audio_packet.get(), sizeof(audio_packet_raw_t) + packet_data.size()), session->audio.peer);
BOOST_LOG(verbose) << "Audio ["sv << frame << "] :: send..."sv;
BOOST_LOG(verbose) << "Audio ["sv << sequenceNumber << "] :: send..."sv;
}
shutdown_event->raise(true);
@ -947,7 +952,8 @@ std::shared_ptr<session_t> alloc(config_t &config, crypto::aes_t &gcm_key, crypt
session->video.idr_events = mail->event<video::idr_t>(mail::idr);
session->video.lowseq = 0;
session->audio.frame = 1;
session->audio.sequenceNumber = 0;
session->audio.timestamp = 0;
session->control.peer = nullptr;
session->state.store(state_e::STOPPED, std::memory_order_relaxed);