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edit pass: media-quality-sdk
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‎articles/communication-services/concepts/voice-video-calling/includes/media-stats/media-stats-android.md

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@@ -23,7 +23,7 @@ MediaStatsCallFeature mediaStatsCallFeature = call.feature(Features.MEDIA_STATS)
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The `MediaStatsCallFeature` object has the following API structure:
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- The `OnSampleReportedListener` event listens for periodic reports of the media statistics.
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- `setSampleIntervalInSeconds(int value)` sets the interval (in seconds) of the media statistics report generation. If it's not specified, the SDK uses defaults.
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- `setSampleIntervalInSeconds(int value)` sets the interval, in seconds, of the media statistics report generation. If it's not specified, the SDK uses defaults.
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- A `MediaStatsReport` object contains the definition of the outgoing and incoming media statistics, categorized by audio, video, and screen share.
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- `getOutgoingMediaStats()`: The list of media statistics for outgoing media.
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- `getAudio()`: The list of media statistics for outgoing audio.

‎articles/communication-services/concepts/voice-video-calling/includes/media-stats/media-stats-web.md

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@@ -90,7 +90,7 @@ The bandwidth metrics have changes to `availableBitrate` in audio send and video
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| Metric name | Description | Comments |
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| ----------- | ----------- | -------- |
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| `id` | Statistics ID | It's used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
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| `codecName` | Codec name | OPUS, G722|
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| `codecName` | Codec name | OPUS, G722.|
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| `bitrate` | Audio send bit rate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
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| `jitterInMs` | Packet jitter (milliseconds) | Lower is better. |
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| `packetsPerSecond` | Packet rate (packets per second) | |
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| Metric name | Description | Comments |
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| ----------- | ----------- | -------- |
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| `id` | Statistics ID | It's used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
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| `codecName` | Codec name | H264, VP8, VP9 |
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| `codecName` | Codec name | H264, VP8, VP9. |
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| `bitrate` | Video send bitrate (bits per second) | |
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| `jitterInMs` | Packet jitter (milliseconds) | Lower is better. |
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| `packetsPerSecond` | Packet rate (packets per second) | |
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| Metric name | Description | Comments |
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| ----------- | ----------- | -------- |
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| `id` | Statistics ID | It's used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
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| `codecName` | Codec name | H264, VP8, VP9 |
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| `codecName` | Codec name | H264, VP8, VP9. |
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| `bitrate` | Video receive bitrate (bits per second) | |
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| `jitterInMs` | Packet jitter (milliseconds) | Lower is better. |
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| `packetsPerSecond` | Packet rate (packets per second) | |
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### Using media quality statistics on SDK versions earlier than 1.8.0
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If you're using an Azure Communication Services SDK version earlier than 1.8.0, see below for documentation on how to use this functionality.
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If you're using an Azure Communication Services SDK version earlier than 1.8.0, this section describes how to use this functionality.
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As a developer, you can invoke the `startCollector` method with optional `mediaStatsSubscriptionOptions`.
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| `audioSendCodecName` | Sent codec | Audio codec used. | Information only. |
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| `audioSendRtt` | Send round-trip time | Round-trip time between your system and the Azure Communication Services server. Results are in milliseconds. | We recommend a round-trip time of 200 ms or less. |
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| `audioSendPairRtt` | Send pair round-trip time | Round-trip time for the entire transport. Results are in milliseconds. | We recommend a round-trip time of 200 ms or less. |
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| `audioRecvPairRtt` | Receive pair round-trip Time | Round-trip time for the entire transport. Results are in milliseconds. | We recommend a round-trip time of 200 ms or less. |
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| `audioSendAudioInputLevel` | Input level for the microphone | Sent audio playout level. If source data is between 0 and 1, the media stack multiplies it with `0xFFFF`. It depends on the microphone. Used to confirm if the microphone is silent (no incoming energy). | Microphone input level. |
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| `audioRecvPairRtt` | Receive pair round-trip time | Round-trip time for the entire transport. Results are in milliseconds. | We recommend a round-trip time of 200 ms or less. |
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| `audioSendAudioInputLevel` | Input level for the microphone | Sent audio playout level. If source data is between 0 and 1, the media stack multiplies it with `0xFFFF`. It depends on the microphone. Use this metric to confirm if the microphone is silent (no incoming energy). | Microphone input level. |
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| `audioRecvAudioOutputLevel` | Speaker output level | Received audio playout level. If the source data is between 0 and 1, the media stack multiplies it with `0xFFFF`. | Speaker output level. |
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### Video quality metrics
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| `videoSendFrameRateSent` | Sent frame rate | Number of video frames sent. Results are frames per second. | Higher is better. Possible values: 25-30 fps (360p or higher); <br>8-15 fps (270p or lower). |
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| `videoSendFrameWidthSent` | Sent width | Video width resolution sent. | Higher is better. Possible values:<br>1920, 1280, 960, 640, 480, 320. |
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| `videoSendFrameHeightSent` | Sent height | Video height sent. Higher is better. | Higher is better. Possible values:<br>1080, 720, 540, 360, 270, 240. |
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| `videoSendBitrate` | Sent bitrate | Amount of video bitrate being sent. Results are bits per second. | |
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| `videoSendBitrate` | Sent bitrate | Amount of video bitrate sent. Results are bits per second. | |
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| `videoSendPackets` | Sent packets | The number of video packets sent. Results are packets per second (over the last second). | Information only. |
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| `VideoSendCodecName` | Sent codec | Video codec used for encoding video. | Possible values: VP8 (1:1 calls) and H264. |
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| `videoRecvJitterBufferMs` | Received jitter | The jitter buffer is used for smooth playout. This value is how long the packets of the frame stay in the jitter buffer (in milliseconds). | Lower is better. |
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| Metric name | Purpose | Details | Comments |
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| -------------------------------------- | -------------------------------- | ---------------------------------------------------------------- | ------------------------------------- |
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| `screenSharingSendFrameRateSent` | Sent frame rate | Number of video frames sent. Higher is better. | 1-30 FPS (content aware, variable). |
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| `screenSharingSendFrameRateSent` | Sent frame rate | Number of video frames sent. Higher is better. | 1-30 fps (content aware, variable). |
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| `screenSharingSendFrameWidthSent` | Sent width | Video resolution sent. Higher is better. | 1920 pixels (content aware, variable). |
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| `screenSharingSendFrameHeightSent` | Sent height | Video resolution sent. Higher is better. | 1080 pixels (content aware, variable). |
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| `screenSharingSendCodecName` | Sent codec | Codec used for encoding a screen share. | Information only. |
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| `screenSharingRecvFrameRateReceived` | Received frame rate | Number of video frames received. Lower is better. | 1-30 FPS. |
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| `screenSharingRecvFrameRateReceived` | Received frame rate | Number of video frames received. Lower is better. | 1-30 fps. |
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| `screenSharingRecvFrameWidthReceived` | Received width | Video resolution received. Higher is better. | 1920 pixels (content aware, variable). |
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| `screenSharingRecvFrameHeightReceived` | Received height | Video resolution sent. Higher is better. | 1080 pixels (content aware, variable). |
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| `screenSharingRecvCodecName` | Received codec | Codec used for decoding the video stream. | Information only. |

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