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[cast] Make fake announcement of video stream (fixes #1092)

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For unknown reasons some Chromecast devices disconnect causing a TLS error if
we don't OFFER a video stream. Seems actually sending video isn't required to
fix the issue.
This commit is contained in:
ejurgensen 2020-11-04 12:20:16 +01:00
parent db43d51568
commit a11ab808b2
1 changed files with 10 additions and 1 deletions
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11
src/outputs/cast.c
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@ -120,6 +120,12 @@
// See cast_packet_header_make()
#define CAST_HEADER_SIZE 11
// These limits are defined in components/mirroring/service/session.cc
#define CAST_SSRC_AUDIO_MIN 1
#define CAST_SSRC_AUDIO_MAX 500000
#define CAST_SSRC_VIDEO_MIN 500001
#define CAST_SSRC_VIDEO_MAX 1000000
/* Notes
* OFFER/ANSWER <-webrtc
* RTCP/RTP
@ -383,7 +389,7 @@ struct cast_msg_basic cast_msg[] =
// sampleRate seems to be ignored
// storeTime unknown meaning - perhaps size of buffer?
// targetDelay - should be RTP delay in ms, but doesn't seem to change anything?
.payload = "{'type':'OFFER','seqNum':%u,'offer':{'castMode':'mirroring','supportedStreams':[{'index':0,'type':'audio_source','codecName':'opus','rtpProfile':'cast','rtpPayloadType':127,'ssrc':%" PRIu32 ",'storeTime':400,'targetDelay':400,'bitRate':128000,'sampleRate':48000,'timeBase':'1/48000','channels':2,'receiverRtcpEventLog':false}]}}",
.payload = "{'type':'OFFER','seqNum':%u,'offer':{'castMode':'mirroring','supportedStreams':[{'index':0,'type':'audio_source','codecName':'opus','rtpProfile':'cast','rtpPayloadType':127,'ssrc':%" PRIu32 ",'storeTime':400,'targetDelay':400,'bitRate':128000,'sampleRate':48000,'timeBase':'1/48000','channels':2,'receiverRtcpEventLog':false},{'codecName':'vp8','index':1,'maxBitRate':5000000,'maxFrameRate':'30000/1000','receiverRtcpEventLog':false,'renderMode':'video','resolutions':[{'height':900,'width':1600}],'rtpPayloadType':96,'rtpProfile':'cast','ssrc':999999,'targetDelay':400,'timeBase':'1/90000','type':'video_source'}]}}",
.flags = USE_TRANSPORT_ID | USE_REQUEST_ID,
},
{
@ -1503,6 +1509,9 @@ master_session_make(struct media_quality *quality)
return NULL;
}
// Change the SSRC to be in the interval [CAST_SSRC_AUDIO_MIN, CAST_SSRC_AUDIO_MAX]
cms->rtp_session->ssrc_id = ((cms->rtp_session->ssrc_id + CAST_SSRC_AUDIO_MIN) % CAST_SSRC_AUDIO_MAX) + CAST_SSRC_AUDIO_MIN;
cms->quality = *quality;
cms->samples_per_packet = CAST_SAMPLES_PER_PACKET;
cms->rawbuf_size = STOB(cms->samples_per_packet, quality->bits_per_sample, quality->channels);