The MCClientOptions class gathers options for the client. More...
#import <client.h>
Properties | |
| NSString * | sourceId |
| The ID of the source to publish. This option is related to the multisource feature of the millicast service. For more information refer to The Multisource Broadcasting Guide. This is a publisher only option. | |
| NSString * | pinnedSourceId |
| The receiving source to pin. Refer to The Multiview Guide to learn more about this. This is a subscriber only option. | |
| NSArray * | excludedSourceId |
| Excluded sources that you do not wish to receive. Refer to The Multiview Guide to learn more about this. This is a subscriber only option. | |
| BOOL | dtx |
| Enables discontinuous transmission on the publishing side, so audio data is only sent when a user’s voice is detected. | |
| int | multiplexedAudioTrack |
| The number of multiplxed audio tracks to receive. This is only available for the subscriber. | |
| NSString * | videoCodec |
| The video codec to use for publishing. This is only available for the publisher. | |
| NSString * | audioCodec |
| The audio codec to use for publishing. This is only available for the publisher. | |
| MCDegradationPreferences | degradationPreferences |
| The strategy the use in order to limit the bandwidth usage. Refer to the WebRTC standard | |
| MCBitrateSettings * | bitrateSettings |
| Adjust the bitrate settings. This is only available for publishing. | |
| BOOL | stereo |
| A boolean indicating whether the SDK should enable stereo audio. True enables stereo, false disables it. This is only available for publishing. | |
| int | statsDelayMs |
| The rate at which you want to receive reports with statistics in milliseconds. Defaults to 1 second. | |
| int | jitterMinimumDelayMs |
| The minimum jitter delay that packets of incoming audio/video streams will experience before being played out. This can be tuned to help with networks with higher latency, but be careful using it as it will introduce this delay. For more information, refer to this document to understand more about what this field does. This is only for subscribing. | |
| MCForcePlayoutDelay * | forcePlayoutDelay |
| Sets the minimum/maximum playout delays for the incoming streams. | |
| BOOL | disableAudio |
| Determines whether audio playback should be completely disabled. Disabling unnecessary audio helps reduce audio-to-video synchronization delays. This is only available on the subscriber. | |
| NSNumber * | maximumBitrate |
| Set the maximum bitrate in KBps that can be used by the viewer. | |
| MCScalabilityMode | svcMode |
| Enables Scalable Video Coding selection. Refer to the WebRTC standard to learn which modes are supported by which codecs. This is only available when publishing. | |
| BOOL | simulcast |
Determines whether Simulcast should be enabled (true) or not (false). This is only available for VP8 and H264 codecs, and is false by default. This is only available for publishing. Enabling this will send out 3 simulcast streams (low, medium and high). | |
| NSString * | rtcEventLogOutputPath |
| Enables logging RTC event log into a custom file path. | |
| BOOL | recordStream |
| Indicates whether the SDK should enable stream recording immediately after publishing. Make sure the recording feature is enabled for the publisher token. Recordings can then be viewed on the dashboard. | |
| NSNumber * | priority |
| The priority of redundant streams that indicates the order in which backup streams should be broadcasted in the case of any problems with the primary stream. Refer to the Redundant Ingest Guide to understand more. | |
| BOOL | forceSmooth |
| NSNumber * | bweMonitorDurationUs |
| NSNumber * | bweRateChangePercentage |
| What percentage of the estimated larger bitrate will we increase by when we think the network is good. So if the previous estimate was 1mbps and the new estimate is 10mbps, and bweRateChangePercentage is 0.05, then 0.05*10mbps = 0.5mbps, so the new target at the end of the tick will be 1 + 0.5 = 1.5mbps. This value can be reduced to increase more slowly under good conditions. Expects a double value. | |
| NSNumber * | upwardsLayerWaitTimeMs |
| Duration the Transponder will wait before switching back to a higher layer in msec. Expects an unsigned integer value. | |
Detailed Description
The MCClientOptions class gathers options for the client.
Property Documentation
◆ audioCodec
|
readwritenonatomicretain |
The audio codec to use for publishing. This is only available for the publisher.
◆ bitrateSettings
|
readwritenonatomicretain |
Adjust the bitrate settings. This is only available for publishing.
◆ bweMonitorDurationUs
|
readwritenonatomicretain |
Overrides the duration of time over which bandwidth-estimate-calculations occur in usec. By default this is 150msec, increasing this may smooth out transitions a little at the cost of NOT reacting to changes in the network as fast. Expects an unsigned integer value.
◆ bweRateChangePercentage
|
readwritenonatomicretain |
What percentage of the estimated larger bitrate will we increase by when we think the network is good. So if the previous estimate was 1mbps and the new estimate is 10mbps, and bweRateChangePercentage is 0.05, then 0.05*10mbps = 0.5mbps, so the new target at the end of the tick will be 1 + 0.5 = 1.5mbps. This value can be reduced to increase more slowly under good conditions. Expects a double value.
◆ degradationPreferences
|
readwritenonatomicassign |
The strategy the use in order to limit the bandwidth usage. Refer to the WebRTC standard
◆ disableAudio
|
readwritenonatomicassign |
Determines whether audio playback should be completely disabled. Disabling unnecessary audio helps reduce audio-to-video synchronization delays. This is only available on the subscriber.
◆ dtx
|
readwritenonatomicassign |
Enables discontinuous transmission on the publishing side, so audio data is only sent when a user’s voice is detected.
◆ excludedSourceId
|
readwritenonatomicretain |
Excluded sources that you do not wish to receive. Refer to The Multiview Guide to learn more about this. This is a subscriber only option.
◆ forcePlayoutDelay
|
readwritenonatomicretain |
Sets the minimum/maximum playout delays for the incoming streams.
◆ forceSmooth
|
readwritenonatomicassign |
Forces video to be sent on downlink when switching from higher quality layers
If true, smooth transitions will be enabled when changing from a higher rate to a lower rate, this may cause congested use cases to take longer to recover If false, then when switching from a higher rate to lower rate (due to congestion) we will stop sending video packets until the next I frame arrives at the new lower rate causing a pause but improving the impacts of congestion and recovering quicker.
◆ jitterMinimumDelayMs
|
readwritenonatomicassign |
The minimum jitter delay that packets of incoming audio/video streams will experience before being played out. This can be tuned to help with networks with higher latency, but be careful using it as it will introduce this delay. For more information, refer to this document to understand more about what this field does. This is only for subscribing.
◆ maximumBitrate
|
readwritenonatomicretain |
Set the maximum bitrate in KBps that can be used by the viewer.
◆ multiplexedAudioTrack
|
readwritenonatomicassign |
The number of multiplxed audio tracks to receive. This is only available for the subscriber.
◆ pinnedSourceId
|
readwritenonatomicretain |
The receiving source to pin. Refer to The Multiview Guide to learn more about this. This is a subscriber only option.
◆ priority
|
readwritenonatomicassign |
The priority of redundant streams that indicates the order in which backup streams should be broadcasted in the case of any problems with the primary stream. Refer to the Redundant Ingest Guide to understand more.
◆ recordStream
|
readwritenonatomicassign |
Indicates whether the SDK should enable stream recording immediately after publishing. Make sure the recording feature is enabled for the publisher token. Recordings can then be viewed on the dashboard.
◆ rtcEventLogOutputPath
|
readwritenonatomicretain |
Enables logging RTC event log into a custom file path.
◆ simulcast
|
readwritenonatomicassign |
Determines whether Simulcast should be enabled (true) or not (false). This is only available for VP8 and H264 codecs, and is false by default. This is only available for publishing. Enabling this will send out 3 simulcast streams (low, medium and high).
◆ sourceId
|
readwritenonatomicretain |
The ID of the source to publish. This option is related to the multisource feature of the millicast service. For more information refer to The Multisource Broadcasting Guide. This is a publisher only option.
◆ statsDelayMs
|
readwritenonatomicassign |
The rate at which you want to receive reports with statistics in milliseconds. Defaults to 1 second.
◆ stereo
|
readwritenonatomicassign |
A boolean indicating whether the SDK should enable stereo audio. True enables stereo, false disables it. This is only available for publishing.
◆ svcMode
|
readwritenonatomicassign |
Enables Scalable Video Coding selection. Refer to the WebRTC standard to learn which modes are supported by which codecs. This is only available when publishing.
◆ upwardsLayerWaitTimeMs
|
readwritenonatomicretain |
Duration the Transponder will wait before switching back to a higher layer in msec. Expects an unsigned integer value.
◆ videoCodec
|
readwritenonatomicretain |
The video codec to use for publishing. This is only available for the publisher.
The documentation for this class was generated from the following file:
- objc/MillicastSDK/client.h
