gopal_apprik
New Member
Hi,
Conference Screen(Android Sample App): The app is not responding on click of leave button in Android 10 version mobile(Xiaomi Note 7 Pro).
Steps to produce the bug:
Scenario-1:
i. Join 2 users in one room.
ii. publish the video for another mobile (2nd user).
iii. click leave button in first mobile(1st user, Xiaomi Note 7 Pro mobile)
Scenario-2:
i. Join 2 users in one room.
ii. publish the video for another mobile (2nd user) and stop - this step do multiple times
iii. If step-ii is happening multiple times, the app is not responding in 1st mobile(1st user, Xiaomi Note 7 Pro mobile)
Note:
Using Android SDK 1.1(i.e., wcs-android-sdk-1.1.0.16-release@aar)
Any help will be very helpful.
Log:
2021-01-04 21:40:40.691 7078-7078/com.vcdemo D/MainActivity::: onCreate: mJoinButtonL toLeave
2021-01-04 21:40:40.698 7078-7162/com.vcdemo I/MediaConnection: D1 - Thread-6
2021-01-04 21:40:40.699 7078-7162/com.vcdemo I/MediaConnection: D3 - Thread-6
2021-01-04 21:40:40.699 7078-7162/com.vcdemo D/WCSAudioManager: close
2021-01-04 21:40:40.701 7078-7139/com.vcdemo I/MediaConnection: D2 - Thread-3
2021-01-04 21:40:40.710 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 2541): VideoReceiveStream stats: 613871322, {ssrc: 155127907, total_bps: 568440, width: 360, height: 480, key: 3, delta: 83, network_fps: 8, decode_fps: 8, render_fps: 8, decode_ms: 7, max_decode_ms: 13, cur_delay_ms: 58, targ_delay_ms: 107, jb_delay_ms: 84, min_playout_delay_ms: 0, discarded: 0, sync_offset_ms: 623, cum_loss: 58, max_ext_seq: 6977, nack: 14, fir: 0, pli: 6}
2021-01-04 21:40:40.710 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 1339): Call stats: 613871322, {send_bw_bps: 0, recv_bw_bps: 613113, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
2021-01-04 21:40:40.711 7078-7142/com.vcdemo I/peerconnection.cc: (line 3711): Session: 4459859850551373494 Old state: kStable New state: kClosed
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: IceConnectionState: CLOSED
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: ConnectionState: CLOSED
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: SignalingState: CLOSED
2021-01-04 21:40:40.712 7078-7141/com.vcdemo I/webrtcvoiceengine.cc: (line 1986): SetOutputVolume() to 0 for recv stream with ssrc 1314681839
2021-01-04 21:40:40.712 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 1293): SetSink: ssrc:155127907 nullptr
2021-01-04 21:40:40.713 7078-7141/com.vcdemo I/channel.cc: (line 564): Channel disabled
2021-01-04 21:40:40.714 7078-7141/com.vcdemo I/channel.cc: (line 939): Changing video state, send=0
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/call.cc: (line 1079): UpdateAggregateNetworkState: aggregate_state=up
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/send_side_congestion_controller.cc: (line 334): SignalNetworkState Up
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/video_receive_stream.cc: [1/2] (line 198): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {}}, {payload_type: 100, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}], rtp: {remote_ssrc: 155127907, local_ssrc: 1, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: off, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 0, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/video_receive_stream.cc: [2/2] , {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietfarams:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: default, target_delay_ms: 0}
2021-01-04 21:40:40.719 7078-7141/com.vcdemo I/AndroidVideoDecoder: release
2021-01-04 21:40:40.746 7078-7309/com.vcdemo I/AndroidVideoDecoder: Releasing MediaCodec on output thread
2021-01-04 21:40:40.765 7078-7078/com.vcdemo D/WCSProximitySensor: Proximity sensor => FAR state
2021-01-04 21:40:40.765 7078-7304/com.vcdemo D/SurfaceUtils: disconnecting from surface 0x78ce70e010, reason disconnectFromSurface
2021-01-04 21:40:40.782 7078-7309/com.vcdemo I/AndroidVideoDecoder: Release on output thread done
2021-01-04 21:40:40.783 7078-7141/com.vcdemo I/SurfaceTextureHelper: stopListening()
2021-01-04 21:40:40.783 7078-7141/com.vcdemo I/SurfaceTextureHelper: dispose()
2021-01-04 21:40:40.787 7078-7141/com.vcdemo I/videodecoderwrapper.cc: (line 136): release: 0
2021-01-04 21:40:40.788 7078-7141/com.vcdemo I/video_render_frames.cc: (line 47): WebRTC.Video.DroppedFrames.RenderQueue 0
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/receive_statistics_proxy.cc: (line 487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 13
Frames decoded 72
WebRTC.Video.DroppedFrames.Receiver 14
WebRTC.Video.ReceivedPacketsLostInPercent 5
WebRTC.Video.InterframeDelay95PercentileInMs 226
WebRTC.Video.MediaBitrateReceivedInKbps 755
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/video_quality_observer.cc: (line 121): WebRTC.Video.MeanTimeBetweenFreezesMs 5026
WebRTC.Video.MeanFreezeDurationMs 745
WebRTC.Video.TimeInHdPercentage 0
WebRTC.Video.TimeInBlockyVideoPercentage 0
WebRTC.Video.NumberResolutionDownswitchesPerMinute 5
WebRTC.Video.NumberFreezesPerMinute 5
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/channel.cc: (line 133): Destroyed channel: video
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/messagequeue.cc: (line 518): Message took 76ms to dispatch. Posted from: DestroyVideoChannel@../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/pc/channelmanager.cc:272
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/channel.cc: (line 564): Channel disabled
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_impl.cc: (line 801): StopPlayout
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_template.h: (line 194): Playing
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_template.h: (line 188): StopPlayout
2021-01-04 21:40:40.790 7078-7141/com.vcdemo D/OpenSLESPlayer: StopPlayout[tid=7141]
2021-01-04 21:40:40.790 7078-7141/com.vcdemo D/AudioTrack: stop(9649): called with 647040 frames delivered
2021-01-04 21:40:40.792 7078-7141/com.vcdemo D/OpenSLESPlayer: DestroyAudioPlayer
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_buffer.cc: (line 137): StopPlayout
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_buffer.cc: (line 143): total playout time: 13582
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_impl.cc: (line 805): output: 0
2021-01-04 21:40:40.799 7078-7141/com.vcdemo I/channel.cc: (line 808): Changing voice state, recv=0 send=0
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/webrtcvoiceengine.cc: (line 1924): RemoveRecvStream: 1314681839
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/call.cc: (line 1079): UpdateAggregateNetworkState: aggregate_state=down
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/send_side_congestion_controller.cc: (line 334): SignalNetworkState Down
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/audio_receive_stream.cc: (line 135): ~AudioReceiveStream: 1314681839
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/channel.cc: (line 133): Destroyed channel: audio
2021-01-04 21:40:40.801 7078-7140/com.vcdemo I/opensslstreamadapter.cc: (line 917): Cleanup
2021-01-04 21:40:40.802 7078-7140/com.vcdemo I/NetworkMonitor: Stop monitoring with native observer 520379029248
2021-01-04 21:40:40.802 7078-7140/com.vcdemo I/NetworkMonitorAutoDetect: Unregister network callback
2021-01-04 21:40:40.803 7078-7140/com.vcdemo I/NetworkMonitorAutoDetect: Unregister network callback
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/call.cc: (line 590): WebRTC.Call.AudioBitrateReceivedInBps, periodic_samples:6, {min:14088, avg:20520, max:24880}
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/call.cc: (line 606): WebRTC.Call.BitrateReceivedInBps, periodic_samples:6, {min:658104, avg:793304, max:948976}
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/paced_sender.cc: (line 385): ProcessThreadAttached 0x0
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/paced_sender.cc: (line 385): ProcessThreadAttached 0x0
2021-01-04 21:40:40.810 7078-7141/com.vcdemo I/rtc_event_log_impl.cc: (line 202): Stopping WebRTC event log.
2021-01-04 21:40:40.810 7078-7141/com.vcdemo I/rtc_event_log_impl.cc: (line 219): WebRTC event log successfully stopped.
2021-01-04 21:40:40.813 7078-7142/com.vcdemo I/messagequeue.cc: (line 518): Message took 110ms to dispatch. Posted from: Close@../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/api/peerconnectionproxy.h:135
2021-01-04 21:40:40.814 7078-7142/com.vcdemo I/peerconnection.cc: (line 860): Session: 4459859850551373494 is destroyed.
2021-01-04 21:40:40.821 7078-7139/com.vcdemo D/MediaConnection: Closing the audio manager...
continuing...
Conference Screen(Android Sample App): The app is not responding on click of leave button in Android 10 version mobile(Xiaomi Note 7 Pro).
Steps to produce the bug:
Scenario-1:
i. Join 2 users in one room.
ii. publish the video for another mobile (2nd user).
iii. click leave button in first mobile(1st user, Xiaomi Note 7 Pro mobile)
Scenario-2:
i. Join 2 users in one room.
ii. publish the video for another mobile (2nd user) and stop - this step do multiple times
iii. If step-ii is happening multiple times, the app is not responding in 1st mobile(1st user, Xiaomi Note 7 Pro mobile)
Note:
Using Android SDK 1.1(i.e., wcs-android-sdk-1.1.0.16-release@aar)
Any help will be very helpful.
Log:
2021-01-04 21:40:40.691 7078-7078/com.vcdemo D/MainActivity::: onCreate: mJoinButtonL toLeave
2021-01-04 21:40:40.698 7078-7162/com.vcdemo I/MediaConnection: D1 - Thread-6
2021-01-04 21:40:40.699 7078-7162/com.vcdemo I/MediaConnection: D3 - Thread-6
2021-01-04 21:40:40.699 7078-7162/com.vcdemo D/WCSAudioManager: close
2021-01-04 21:40:40.701 7078-7139/com.vcdemo I/MediaConnection: D2 - Thread-3
2021-01-04 21:40:40.710 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 2541): VideoReceiveStream stats: 613871322, {ssrc: 155127907, total_bps: 568440, width: 360, height: 480, key: 3, delta: 83, network_fps: 8, decode_fps: 8, render_fps: 8, decode_ms: 7, max_decode_ms: 13, cur_delay_ms: 58, targ_delay_ms: 107, jb_delay_ms: 84, min_playout_delay_ms: 0, discarded: 0, sync_offset_ms: 623, cum_loss: 58, max_ext_seq: 6977, nack: 14, fir: 0, pli: 6}
2021-01-04 21:40:40.710 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 1339): Call stats: 613871322, {send_bw_bps: 0, recv_bw_bps: 613113, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}
2021-01-04 21:40:40.711 7078-7142/com.vcdemo I/peerconnection.cc: (line 3711): Session: 4459859850551373494 Old state: kStable New state: kClosed
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: IceConnectionState: CLOSED
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: ConnectionState: CLOSED
2021-01-04 21:40:40.711 7078-7142/com.vcdemo D/MediaConnection: SignalingState: CLOSED
2021-01-04 21:40:40.712 7078-7141/com.vcdemo I/webrtcvoiceengine.cc: (line 1986): SetOutputVolume() to 0 for recv stream with ssrc 1314681839
2021-01-04 21:40:40.712 7078-7141/com.vcdemo I/webrtcvideoengine.cc: (line 1293): SetSink: ssrc:155127907 nullptr
2021-01-04 21:40:40.713 7078-7141/com.vcdemo I/channel.cc: (line 564): Channel disabled
2021-01-04 21:40:40.714 7078-7141/com.vcdemo I/channel.cc: (line 939): Changing video state, send=0
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/call.cc: (line 1079): UpdateAggregateNetworkState: aggregate_state=up
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/send_side_congestion_controller.cc: (line 334): SignalNetworkState Up
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/video_receive_stream.cc: [1/2] (line 198): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {}}, {payload_type: 100, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}], rtp: {remote_ssrc: 155127907, local_ssrc: 1, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: off, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 0, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}
2021-01-04 21:40:40.716 7078-7141/com.vcdemo I/video_receive_stream.cc: [2/2] , {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietfarams:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: default, target_delay_ms: 0}
2021-01-04 21:40:40.719 7078-7141/com.vcdemo I/AndroidVideoDecoder: release
2021-01-04 21:40:40.746 7078-7309/com.vcdemo I/AndroidVideoDecoder: Releasing MediaCodec on output thread
2021-01-04 21:40:40.765 7078-7078/com.vcdemo D/WCSProximitySensor: Proximity sensor => FAR state
2021-01-04 21:40:40.765 7078-7304/com.vcdemo D/SurfaceUtils: disconnecting from surface 0x78ce70e010, reason disconnectFromSurface
2021-01-04 21:40:40.782 7078-7309/com.vcdemo I/AndroidVideoDecoder: Release on output thread done
2021-01-04 21:40:40.783 7078-7141/com.vcdemo I/SurfaceTextureHelper: stopListening()
2021-01-04 21:40:40.783 7078-7141/com.vcdemo I/SurfaceTextureHelper: dispose()
2021-01-04 21:40:40.787 7078-7141/com.vcdemo I/videodecoderwrapper.cc: (line 136): release: 0
2021-01-04 21:40:40.788 7078-7141/com.vcdemo I/video_render_frames.cc: (line 47): WebRTC.Video.DroppedFrames.RenderQueue 0
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/receive_statistics_proxy.cc: (line 487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 13
Frames decoded 72
WebRTC.Video.DroppedFrames.Receiver 14
WebRTC.Video.ReceivedPacketsLostInPercent 5
WebRTC.Video.InterframeDelay95PercentileInMs 226
WebRTC.Video.MediaBitrateReceivedInKbps 755
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/video_quality_observer.cc: (line 121): WebRTC.Video.MeanTimeBetweenFreezesMs 5026
WebRTC.Video.MeanFreezeDurationMs 745
WebRTC.Video.TimeInHdPercentage 0
WebRTC.Video.TimeInBlockyVideoPercentage 0
WebRTC.Video.NumberResolutionDownswitchesPerMinute 5
WebRTC.Video.NumberFreezesPerMinute 5
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/channel.cc: (line 133): Destroyed channel: video
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/messagequeue.cc: (line 518): Message took 76ms to dispatch. Posted from: DestroyVideoChannel@../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/pc/channelmanager.cc:272
2021-01-04 21:40:40.789 7078-7141/com.vcdemo I/channel.cc: (line 564): Channel disabled
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_impl.cc: (line 801): StopPlayout
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_template.h: (line 194): Playing
2021-01-04 21:40:40.790 7078-7141/com.vcdemo I/audio_device_template.h: (line 188): StopPlayout
2021-01-04 21:40:40.790 7078-7141/com.vcdemo D/OpenSLESPlayer: StopPlayout[tid=7141]
2021-01-04 21:40:40.790 7078-7141/com.vcdemo D/AudioTrack: stop(9649): called with 647040 frames delivered
2021-01-04 21:40:40.792 7078-7141/com.vcdemo D/OpenSLESPlayer: DestroyAudioPlayer
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_buffer.cc: (line 137): StopPlayout
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_buffer.cc: (line 143): total playout time: 13582
2021-01-04 21:40:40.798 7078-7141/com.vcdemo I/audio_device_impl.cc: (line 805): output: 0
2021-01-04 21:40:40.799 7078-7141/com.vcdemo I/channel.cc: (line 808): Changing voice state, recv=0 send=0
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/webrtcvoiceengine.cc: (line 1924): RemoveRecvStream: 1314681839
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/call.cc: (line 1079): UpdateAggregateNetworkState: aggregate_state=down
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/send_side_congestion_controller.cc: (line 334): SignalNetworkState Down
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/audio_receive_stream.cc: (line 135): ~AudioReceiveStream: 1314681839
2021-01-04 21:40:40.800 7078-7141/com.vcdemo I/channel.cc: (line 133): Destroyed channel: audio
2021-01-04 21:40:40.801 7078-7140/com.vcdemo I/opensslstreamadapter.cc: (line 917): Cleanup
2021-01-04 21:40:40.802 7078-7140/com.vcdemo I/NetworkMonitor: Stop monitoring with native observer 520379029248
2021-01-04 21:40:40.802 7078-7140/com.vcdemo I/NetworkMonitorAutoDetect: Unregister network callback
2021-01-04 21:40:40.803 7078-7140/com.vcdemo I/NetworkMonitorAutoDetect: Unregister network callback
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/call.cc: (line 590): WebRTC.Call.AudioBitrateReceivedInBps, periodic_samples:6, {min:14088, avg:20520, max:24880}
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/call.cc: (line 606): WebRTC.Call.BitrateReceivedInBps, periodic_samples:6, {min:658104, avg:793304, max:948976}
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/paced_sender.cc: (line 385): ProcessThreadAttached 0x0
2021-01-04 21:40:40.809 7078-7141/com.vcdemo I/paced_sender.cc: (line 385): ProcessThreadAttached 0x0
2021-01-04 21:40:40.810 7078-7141/com.vcdemo I/rtc_event_log_impl.cc: (line 202): Stopping WebRTC event log.
2021-01-04 21:40:40.810 7078-7141/com.vcdemo I/rtc_event_log_impl.cc: (line 219): WebRTC event log successfully stopped.
2021-01-04 21:40:40.813 7078-7142/com.vcdemo I/messagequeue.cc: (line 518): Message took 110ms to dispatch. Posted from: Close@../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/api/peerconnectionproxy.h:135
2021-01-04 21:40:40.814 7078-7142/com.vcdemo I/peerconnection.cc: (line 860): Session: 4459859850551373494 is destroyed.
2021-01-04 21:40:40.821 7078-7139/com.vcdemo D/MediaConnection: Closing the audio manager...
continuing...