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Squash: Merge SRS 4.0, regression test for RTMP.

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This commit is contained in:
winlin 2021-10-12 08:36:24 +08:00
parent a81aa2edc5
commit b874d9c9ba
32 changed files with 9977 additions and 131 deletions
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50
trunk/src/app/srs_app_rtc_source.cpp
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@ -707,8 +707,8 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
source_ = source;
format = new SrsRtmpFormat();
codec_ = new SrsAudioTranscoder();
discard_aac = false;
discard_bframe = false;
rtmp_to_rtc = false;
keep_bframe = false;
merge_nalus = false;
meta = new SrsMetaCache();
audio_sequence = 0;
@ -743,22 +743,24 @@ srs_error_t SrsRtcFromRtmpBridger::initialize(SrsRequest* r)
srs_error_t err = srs_success;
req = r;
rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(req->vhost);
if ((err = format->initialize()) != srs_success) {
return srs_error_wrap(err, "format initialize");
if (rtmp_to_rtc) {
if ((err = format->initialize()) != srs_success) {
return srs_error_wrap(err, "format initialize");
}
int bitrate = 48000; // The output bitrate in bps.
if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate,
bitrate)) != srs_success) {
return srs_error_wrap(err, "init codec");
}
}
int bitrate = 48000; // The output bitrate in bps.
if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
return srs_error_wrap(err, "init codec");
}
// TODO: FIXME: Support reload.
discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
keep_bframe = _srs_config->get_rtc_keep_bframe(req->vhost);
merge_nalus = _srs_config->get_rtc_server_merge_nalus();
srs_trace("RTC bridge from RTMP, discard_aac=%d, discard_bframe=%d, merge_nalus=%d",
discard_aac, discard_bframe, merge_nalus);
srs_trace("RTC bridge from RTMP, rtmp2rtc=%d, keep_bframe=%d, merge_nalus=%d",
rtmp_to_rtc, keep_bframe, merge_nalus);
return err;
}
@ -767,6 +769,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_publish()
{
srs_error_t err = srs_success;
if (!rtmp_to_rtc) {
return err;
}
// TODO: FIXME: Should sync with bridger?
if ((err = source_->on_publish()) != srs_success) {
return srs_error_wrap(err, "source publish");
@ -781,6 +787,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_publish()
void SrsRtcFromRtmpBridger::on_unpublish()
{
if (!rtmp_to_rtc) {
return;
}
// Reset the metadata cache, to make VLC happy when disable/enable stream.
// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
meta->update_previous_vsh();
@ -795,6 +805,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
if (!rtmp_to_rtc) {
return err;
}
// TODO: FIXME: Support parsing OPUS for RTC.
if ((err = format->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "format consume audio");
@ -813,7 +827,7 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
}
// When drop aac audio packet, never transcode.
if (discard_aac && acodec == SrsAudioCodecIdAAC) {
if (acodec != SrsAudioCodecIdAAC) {
return err;
}
@ -905,6 +919,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
if (!rtmp_to_rtc) {
return err;
}
// cache the sequence header if h264
bool is_sequence_header = SrsFlvVideo::sh(msg->payload, msg->size);
if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
@ -993,7 +1011,7 @@ srs_error_t SrsRtcFromRtmpBridger::filter(SrsSharedPtrMessage* msg, SrsFormat* f
// Because RTC does not support B-frame, so we will drop them.
// TODO: Drop B-frame in better way, which not cause picture corruption.
if (discard_bframe) {
if (!keep_bframe) {
if ((err = sample->parse_bframe()) != srs_success) {
return srs_error_wrap(err, "parse bframe");
}