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RTC: Supports Firefox and mobile access. Also fixes some memory leak bugs. #1998

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RTC: Supports Firefox and mobile access. Also fixes some memory leak bugs. #1998

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a75964a
Merge pull request #2 from ossrs/feature/rtc
duiniuluantanqin Sep 27, 2020
d9f5f7d
Merge branch 'feature/rtc' of https://github.com/duiniuluantanqin/srs…
Sep 27, 2020
eada513
Merge pull request #4 from ossrs/feature/rtc
duiniuluantanqin Oct 13, 2020
17ccfe1
Merge branch 'feature/rtc' of https://github.com/duiniuluantanqin/srs…
Oct 19, 2020
61c7bf2
Merge pull request #5 from ossrs/feature/rtc
duiniuluantanqin Oct 19, 2020
44b6838
Merge branch 'feature/rtc' of https://github.com/duiniuluantanqin/srs…
Oct 19, 2020
dded3b2
rtc:support play webrtc by firefox and mobile phone
Oct 19, 2020
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68 changes: 57 additions & 11 deletions trunk/src/app/srs_app_rtc_conn.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -646,6 +646,8 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
if (pkt->is_audio()) {
// TODO: FIXME: Any simple solution?
SrsRtcAudioSendTrack* audio_track = audio_tracks_[pkt->header.get_ssrc()];
// update pt with sdp payload type, for each player
pkt->header.set_payload_type(audio_track->get_track_media_pt());
if ((err = audio_track->on_rtp(pkt, info)) != srs_success) {
return srs_error_wrap(err, "audio track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
}
Expand All @@ -654,6 +656,8 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
} else {
// TODO: FIXME: Any simple solution?
SrsRtcVideoSendTrack* video_track = video_tracks_[pkt->header.get_ssrc()];
// update pt with sdp payload type, for each player
pkt->header.set_payload_type(video_track->get_track_media_pt());
if ((err = video_track->on_rtp(pkt, info)) != srs_success) {
return srs_error_wrap(err, "video track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
}
Expand Down Expand Up @@ -1815,6 +1819,14 @@ srs_error_t SrsRtcConnection::add_player(SrsRequest* req, const SrsSdp& remote_s
return srs_error_wrap(err, "create player");
}

// release resource
for (std::map<uint32_t, SrsRtcTrackDescription*>::iterator iter = play_sub_relations.begin(); iter != play_sub_relations.end(); ++iter)
{
SrsRtcTrackDescription* track = iter->second;
srs_freep(track);
}
play_sub_relations.clear();

return err;
}

Expand Down Expand Up @@ -2917,6 +2929,7 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S

bool nack_enabled = _srs_config->get_rtc_nack_enabled(req->vhost);
bool twcc_enabled = _srs_config->get_rtc_twcc_enabled(req->vhost);
bool has_42e01f = srs_sdp_has_h264_profile(remote_sdp, "42e01f");

SrsRtcStream* source = NULL;
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
Expand All @@ -2925,6 +2938,10 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S

for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];

SrsRtcTrackDescription* track_desc = new SrsRtcTrackDescription();
SrsAutoFree(SrsRtcTrackDescription, track_desc);

// Whether feature enabled in remote extmap.
int remote_twcc_id = 0;
if (true) {
Expand All @@ -2938,34 +2955,65 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
}

std::vector<SrsRtcTrackDescription*> track_descs;
std::vector<std::string> remote_rtcp_fb;
if (remote_media_desc.is_audio()) {
SrsRtcTrackDescription* source_track_desc = source->get_track_desc("audio", "opus").at(0);
track_desc->ssrc_ = source_track_desc->ssrc_;

// TODO: check opus format specific param
std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
if (payloads.empty()) {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
}

SrsMediaPayloadType payload = payloads.at(0);
remote_rtcp_fb = payload.rtcp_fb_;
for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
// if the playload is opus, and the encoding_param_ is channel
SrsAudioPayload* audio_payload = new SrsAudioPayload(iter->payload_type_, iter->encoding_name_, iter->clock_rate_, ::atol(iter->encoding_param_.c_str()));
audio_payload->set_opus_param_desc(iter->format_specific_param_);
audio_payload->rtcp_fbs_ = iter->rtcp_fb_;

track_descs = source->get_track_desc("audio", "opus");
track_desc->type_ = "audio";
track_desc->set_codec_payload((SrsCodecPayload*)audio_payload);
// Only choose one match opus codec.
break;
}
track_descs.push_back(track_desc);
} else if (remote_media_desc.is_video()) {
// TODO: check opus format specific param
SrsRtcTrackDescription* source_track_desc = source->get_track_desc("video", "H264").at(0);
track_desc->ssrc_ = source_track_desc->ssrc_;

// TODO: check H264 format specific param
std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
if (payloads.empty()) {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found h264 payload type");
}

SrsMediaPayloadType payload = payloads.at(0);
remote_rtcp_fb = payload.rtcp_fb_;
for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
H264SpecificParam h264_param;
if ((err = srs_parse_h264_fmtp(iter->format_specific_param_, h264_param)) != srs_success) {
srs_error_reset(err); continue;
}

// If not exists 42e01f, we pick up any profile such as 42001f.
bool profile_matched = (!has_42e01f || h264_param.profile_level_id == "42e01f");

track_descs = source->get_track_desc("video", "H264");
// Try to pick the "best match" H.264 payload type.
if (h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1" && profile_matched) {
SrsVideoPayload* video_payload = new SrsVideoPayload(iter->payload_type_, iter->encoding_name_, iter->clock_rate_);
video_payload->set_h264_param_desc(iter->format_specific_param_);
video_payload->rtcp_fbs_ = iter->rtcp_fb_;

track_desc->type_ = "video";
track_desc->set_codec_payload((SrsCodecPayload*)video_payload);
// Only choose first match H.264 payload type.
break;
}
}
track_descs.push_back(track_desc);
}

for (int i = 0; i < (int)track_descs.size(); ++i) {
SrsRtcTrackDescription* track = track_descs[i]->copy();
track->mid_ = remote_media_desc.mid_;
track->set_mid(remote_media_desc.mid_);
uint32_t publish_ssrc = track->ssrc_;

vector<string> rtcp_fb;
Expand All @@ -2984,8 +3032,6 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
}
}

track->ssrc_ = SrsRtcSSRCGenerator::instance()->generate_ssrc();

// TODO: FIXME: set audio_payload rtcp_fbs_,
// according by whether downlink is support transport algorithms.
// TODO: FIXME: if we support downlink RTX, MUST assign rtx_ssrc_, rtx_pt, rtx_apt
Expand Down
21 changes: 19 additions & 2 deletions trunk/src/app/srs_app_rtc_source.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -1248,7 +1248,7 @@ SrsMediaPayloadType SrsVideoPayload::generate_media_payload_type()

std::ostringstream format_specific_param;
if (!h264_param_.level_asymmerty_allow.empty()) {
format_specific_param << "level-asymmetry-allowed=" << h264_param_.level_asymmerty_allow;
format_specific_param << ";level-asymmetry-allowed=" << h264_param_.level_asymmerty_allow;
}
if (!h264_param_.packetization_mode.empty()) {
format_specific_param << ";packetization-mode=" << h264_param_.packetization_mode;
Expand All @@ -1258,6 +1258,7 @@ SrsMediaPayloadType SrsVideoPayload::generate_media_payload_type()
}

media_payload_type.format_specific_param_ = format_specific_param.str();
media_payload_type.format_specific_param_ = media_payload_type.format_specific_param_.substr(1);

return media_payload_type;
}
Expand Down Expand Up @@ -1345,8 +1346,14 @@ SrsMediaPayloadType SrsAudioPayload::generate_media_payload_type()
media_payload_type.rtcp_fb_ = rtcp_fbs_;

std::ostringstream format_specific_param;
if (opus_param_.stereo) {
format_specific_param << ";stereo=" << opus_param_.stereo;
}
if (opus_param_.maxplaybackrate) {
format_specific_param << ";maxplaybackrate=" << opus_param_.maxplaybackrate;
}
if (opus_param_.minptime) {
format_specific_param << "minptime=" << opus_param_.minptime;
format_specific_param << ";minptime=" << opus_param_.minptime;
}
if (opus_param_.use_inband_fec) {
format_specific_param << ";useinbandfec=1";
Expand All @@ -1355,6 +1362,7 @@ SrsMediaPayloadType SrsAudioPayload::generate_media_payload_type()
format_specific_param << ";usedtx=1";
}
media_payload_type.format_specific_param_ = format_specific_param.str();
media_payload_type.format_specific_param_ = media_payload_type.format_specific_param_.substr(1);
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@duiniuluantanqin duiniuluantanqin Nov 12, 2020 •
edited by winlinvip
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This line of code is used to remove the first semicolon. The purpose of having the first semicolon is to avoid formatting errors that may occur when the first parameter is missing.

TRANS_BY_GPT3

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yes


return media_payload_type;
}
Expand All @@ -1372,6 +1380,10 @@ srs_error_t SrsAudioPayload::set_opus_param_desc(std::string fmtp)
opus_param_.use_inband_fec = (kv[1] == "1") ? true : false;
} else if (kv[0] == "usedtx") {
opus_param_.usedtx = (kv[1] == "1") ? true : false;
} else if (kv[0] == "stereo") {
opus_param_.stereo = (int)::atol(kv[1].c_str());
} else if (kv[0] == "maxplaybackrate") {
opus_param_.maxplaybackrate = (int)::atol(kv[1].c_str());
}
} else {
return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid opus param=%s", vec[i].c_str());
Expand Down Expand Up @@ -1731,6 +1743,11 @@ std::string SrsRtcRecvTrack::get_track_id()
return track_desc_->id_;
}

int SrsRtcSendTrack::get_track_media_pt()
{
return track_desc_->media_->pt_;
}

srs_error_t SrsRtcRecvTrack::on_nack(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
Expand Down
5 changes: 5 additions & 0 deletions trunk/src/app/srs_app_rtc_source.hpp
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Original file line number Diff line number Diff line change
Expand Up @@ -322,11 +322,15 @@ class SrsAudioPayload : public SrsCodecPayload
{
struct SrsOpusParameter
{
int stereo;
int maxplaybackrate;
int minptime;
bool use_inband_fec;
bool usedtx;

SrsOpusParameter() {
stereo = 0; // 0:mono; 1:stereo
maxplaybackrate = 0;
minptime = 0;
use_inband_fec = false;
usedtx = false;
Expand Down Expand Up @@ -566,6 +570,7 @@ class SrsRtcSendTrack
bool set_track_status(bool active);
bool get_track_status();
std::string get_track_id();
int get_track_media_pt();
public:
virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info) = 0;
virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt) = 0;
Expand Down