Audio Basic Features

If you need to adjust the local playback volume of a specific remote user, it is recommended to call adjustUserPlaybackSignalVolume.

Timing

You can call this method before or after joining a channel.

Parameters

volume

User volume, range [0,400]:

• 0: Mute.
• 100: (Default) Original volume.
• 400: Four times the original volume (amplifies the audio signal by four times).

Return Values

You can call this method to adjust the playback volume of a specified remote user. To adjust the volume for multiple remote users, call this method multiple times, once for each user.

Timing

Call this method after joining a channel.

Parameters

uid

The user ID of the remote user.

volume

User volume, range [0, 400]:

• 0: Mute.
• 100: (Default) Original volume.
• 400: Four times the original volume (amplifies the audio signal by four times).

Return Values

You can call this method to adjust the playback volume of a specified remote user. To adjust the playback volume of multiple remote users, call this method multiple times, once for each user.

Scenario

This method applies to multi-channel scenarios.

Timing

Call this method after calling joinChannelEx.

Parameters

uid

User ID of the remote user.

volume

The volume of the user. The value range is [0, 400]:

• 0: Mute.
• 100: (Default) Original volume.
• 400: Four times the original volume (amplifies the audio signal by four times).

connection

Connection information. See RtcConnection.

Return Values

The audio module is enabled by default. You can call this method to disable it.

Timing

You can call this method before or after joining a channel. It remains effective after leaving the channel.

Return Values

After calling disableAudio to disable the audio module, you can call enableAudio to re-enable it.

Timing

You can call this method before or after joining a channel. It remains effective after leaving the channel.

Return Values

This method enables the SDK to periodically report the volume information of the local sending stream user and up to three remote users with the highest instantaneous volume to the app.

Timing

You can call this method before or after joining a channel.

Related Callbacks

After this method is successfully called, if there are users publishing streams in the channel, the SDK triggers the onAudioVolumeIndication callback based on the configured interval.

Parameters

interval

Sets the interval (ms) between two volume indications:

• ≤ 0: Disable volume indication.
• > 0: Interval (ms) between two volume indications. The value must be greater than 10, otherwise the onAudioVolumeIndication callback will not be received. It is recommended to set it greater than 100.

smooth

Sets the smoothing factor for the sensitivity of volume indication. Range is 0 to 10, recommended value is 3. The higher the value, the more sensitive the indication.

reportVad

• true: Enable voice activity detection (VAD) for the local user. When enabled, the vad parameter in the onAudioVolumeIndication callback reports the local user's voice activity status.
• false: (Default) Disable VAD for the local user. When disabled, the vad parameter in the onAudioVolumeIndication callback does not report the local user's voice activity status unless the engine automatically detects voice activity.

Return Values

This method enables the SDK to periodically report the volume information of the local user and up to three remote users with the highest instantaneous volume.

Scenario

This method applies to multi-channel scenarios.

Timing

Call this method after calling joinChannelEx.

Related Callbacks

If the method call succeeds and users in the channel publish audio streams, the SDK triggers the onAudioVolumeIndication callback based on the set interval.

Parameters

interval

The time interval (ms) between volume indications:

• ≤ 0: Disables the volume indication.
• > 0: Interval between indications. Must be greater than 10 to receive onAudioVolumeIndication callbacks. Recommended value: > 100.

smooth

The smoothing factor for sensitivity. Range: [0, 10]. Recommended: 3. Higher values make indications more sensitive.

reportVad

Whether to enable voice activity detection (VAD) for the local user:

• true: Enables VAD. The vad parameter in the onAudioVolumeIndication callback reports the local user's voice activity.
• false: (Default) Disables VAD. The vad parameter does not report local voice activity unless automatically detected by the engine.

connection

Connection information. See RtcConnection.

Return Values

Deprecated

This method is deprecated. Use setAudioProfile(int profile) to set the audio profile and setAudioScenario(int scenario) to set the audio scenario.

Scenario

This method applies to various audio scenarios. You can choose based on your actual needs. For example, in scenarios with high audio quality requirements (such as music teaching), it is recommended to set profile to AUDIO_PROFILE_MUSIC_HIGH_QUALITY (4) and scenario to AUDIO_SCENARIO_GAME_STREAMING (3). You can call this method before or after joining a channel.

Timing

You can call this method before or after joining a channel.

Parameters

profile

Audio profile, including sampling rate, bitrate, encoding mode, and number of channels. See profile:

• DEFAULT (0): Default.
  • Live Broadcast: 48 kHz, music encoding, mono, max bitrate 64 Kbps.
  • Communication: 32 kHz, speech encoding, mono, max bitrate 18 Kbps.
• SPEECH_STANDARD (1): 32 kHz, speech encoding, mono, max bitrate 18 Kbps.
• MUSIC_STANDARD (2): 48 kHz, music encoding, mono, max bitrate 64 Kbps.
• MUSIC_STANDARD_STEREO (3): 48 kHz, music encoding, stereo, max bitrate 80 Kbps.
• MUSIC_HIGH_QUALITY (4): 48 kHz, music encoding, mono, max bitrate 96 Kbps.
• MUSIC_HIGH_QUALITY_STEREO (5): 48 kHz, music encoding, stereo, max bitrate 128 Kbps.

scenario

Audio scenario. Different scenarios use different volume types. See scenario:

• AUDIO_SCENARIO_DEFAULT (0): Default automatic scenario. The SDK selects suitable audio quality based on user role and audio routing.
• AUDIO_SCENARIO_GAME_STREAMING (3): High-quality scenario, suitable for music playback.
• AUDIO_SCENARIO_CHATROOM (5): Chatroom scenario, suitable for frequent role switching or mute/unmute actions.
• AUDIO_SCENARIO_CHORUS (7): Real-time chorus scenario, suitable for low-latency environments. Call getAudioDeviceInfo to check if the device supports ultra-low latency. To experience ultra-low latency, ensure isLowLatencyAudioSupported = true.
• AUDIO_SCENARIO_MEETING (8): Meeting scenario, voice-focused.
• AUDIO_SCENARIO_AI_CLIENT (10): AI conversation scenario, only for interaction with agents created by Conversational AI Engine.

Return Values

If you need to set the audio scenario, call setAudioScenario, or call create and set mAudioScenario in RtcEngineConfig.

Scenario

You can choose an appropriate audio profile based on your actual needs. For example, in scenarios such as music teaching that require high audio quality, it is recommended to set profile to AUDIO_PROFILE_MUSIC_HIGH_QUALITY (4). If you need to set the audio scenario, you can call setAudioScenario, or call create [2/2] and set audioScenario in RtcEngineConfig.

Timing

You can call this method before or after joining a channel.

Parameters

profile

Audio profile, including sampling rate, bitrate, encoding mode, and number of channels:

• DEFAULT (0): Default profile.
  • Live Broadcast: 48 kHz sampling rate, music encoding, mono, max bitrate 64 Kbps.
  • Communication: 32 kHz sampling rate, speech encoding, mono, max bitrate 18 Kbps.
• SPEECH_STANDARD (1): 32 kHz, speech encoding, mono, max bitrate 18 Kbps.
• MUSIC_STANDARD (2): 48 kHz, music encoding, mono, max bitrate 64 Kbps.
• MUSIC_STANDARD_STEREO (3): 48 kHz, music encoding, stereo, max bitrate 80 Kbps.
• MUSIC_HIGH_QUALITY (4): 48 kHz, music encoding, mono, max bitrate 96 Kbps.
• MUSIC_HIGH_QUALITY_STEREO (5): 48 kHz, music encoding, stereo, max bitrate 128 Kbps.

Return Values

Scenario

This method applies to various audio scenarios. For example, in music teaching scenarios with high audio quality requirements, it is recommended to set scenario to AUDIO_SCENARIO_GAME_STREAMING (3).

Timing

You can call this method before or after joining a channel.

Parameters

scenario

Audio scenario. The device uses different volume types under different audio scenarios:

• AUDIO_SCENARIO_DEFAULT (0): Default automatic scenario. The SDK selects the appropriate audio quality based on the user role and audio route.
• AUDIO_SCENARIO_GAME_STREAMING (3): High-quality scenario, suitable for music playback.
• AUDIO_SCENARIO_CHATROOM (5): Chatroom scenario, suitable for frequent role switching or muting/unmuting the microphone.
• AUDIO_SCENARIO_CHORUS (7): Real-time chorus scenario, suitable for good network conditions and ultra-low latency requirements. Before using this enum value, you need to call getAudioDeviceInfo to check whether the audio device supports ultra-low latency capture and playback. To experience ultra-low latency, ensure the audio device supports ultra-low latency (isLowLatencyAudioSupported = true).
• AUDIO_SCENARIO_MEETING (8): Meeting scenario, mainly for voice.
• AUDIO_SCENARIO_AI_CLIENT (10): AI conversation scenario, only applicable when the user interacts with a conversational agent created by the Conversational AI Engine.

Return Values

Trigger Timing

This callback is triggered when the most active remote speaker is detected.

Parameters

uid

User ID of the most active remote speaker.

Deprecated

This method is deprecated. Use onRemoteAudioStats instead.

This callback is triggered every two seconds to report the audio quality of each remote user sending audio streams.

Parameters

uid

Remote user ID sending the audio stream.

quality

User's audio quality:

• QUALITY_UNKNOWN (0): Unknown quality.
• QUALITY_EXCELLENT (1): Excellent quality.
• QUALITY_GOOD (2): Good network quality but slightly lower bitrate than excellent.
• QUALITY_POOR (3): Slightly impaired call quality.
• QUALITY_BAD (4): Unable to communicate smoothly.
• QUALITY_VBAD (5): Very poor quality, almost unable to communicate.
• QUALITY_DOWN (6): Network disconnected, completely unable to communicate.
• QUALITY_DETECTING (8): Performing Last-mile network probe.

delay

Network delay from sender to receiver (ms), including audio sampling preprocessing, network transmission, and jitter buffer delay.

lost

Packet loss rate (%) of audio data sent from the remote user to the receiver.

By default, this callback is disabled. You can enable it by calling the enableAudioVolumeIndication method. Once enabled, when a user sends an audio stream in the channel, the SDK triggers this callback at the interval set in enableAudioVolumeIndication. The SDK triggers two independent onAudioVolumeIndication callbacks to report the volume information of the local user sending audio and up to three remote users with the highest instantaneous volume.

Parameters

speakers

Volume information of users. If this array is empty, it means there are no remote users in the channel or remote users are not sending audio. See AudioVolumeInfo.

totalVolume

Speaker volume, range [0, 255].

• For local user callback, totalVolume is the volume of the local user sending audio.
• For remote user callback, totalVolume is the sum of volumes of up to three remote users with the highest instantaneous volume.
• If startAudioMixing is called, totalVolume is the mixed volume.

Trigger Timing

This callback is triggered when the first local audio frame is published.

Parameters

elapsed

Time interval (ms) from the local user calling joinChannel to the SDK triggering this callback.

Deprecated

This method is deprecated. Use onRemoteAudioStateChanged instead.

Trigger Timing

Parameters

uid

The user ID of the remote user.

elapsed

Time (ms) from the local user calling joinChannel to the SDK triggering this callback.

Deprecated

This method is deprecated. Use onRemoteAudioStateChanged instead.

Trigger Timing

This callback is triggered when the SDK receives the first audio frame from a remote user.

Parameters

uid

The user ID of the remote user.

elapsed

Time (ms) from the local user calling joinChannel to the SDK triggering this callback.

When the state of the local audio stream changes, the SDK triggers this callback to report the current state, helping you troubleshoot audio issues.

Trigger Timing

This callback is triggered when the state of the local audio stream changes.

Parameters

state

The state of the local audio:

• LOCAL_AUDIO_STREAM_STATE_STOPPED (0): The local audio is in the initial state.
• LOCAL_AUDIO_STREAM_STATE_RECORDING (1): The local audio capturing device is started successfully.
• LOCAL_AUDIO_STREAM_STATE_ENCODING (2): The first frame of audio is encoded successfully.
• LOCAL_AUDIO_STREAM_STATE_FAILED (3): Failed to start the local audio.

reason

The reason for the local audio state change:

• LOCAL_AUDIO_STREAM_REASON_OK (0): The local audio state is normal.
• LOCAL_AUDIO_STREAM_REASON_FAILURE (1): The reason for local audio failure is unclear. Prompt the user to rejoin the channel.
• LOCAL_AUDIO_STREAM_REASON_DEVICE_NO_PERMISSION (2): No permission to use the local audio capturing device. Prompt the user to grant permission.
• LOCAL_AUDIO_STREAM_REASON_DEVICE_BUSY (3): The microphone is occupied. Prompt the user to check if another app is using the microphone. Audio capturing will automatically resume about five seconds after the microphone is free, or you can try rejoining the channel.
• LOCAL_AUDIO_STREAM_REASON_CAPTURE_FAILURE (4): Failed to capture local audio.
• LOCAL_AUDIO_STREAM_REASON_ENCODE_FAILURE (5): Failed to encode local audio.
• LOCAL_AUDIO_STREAM_REASON_INTERRUPTED (8): Local audio capturing is interrupted by a system phone call, smart assistant, or alarm. To continue capturing, prompt the user to end the phone call, smart assistant, or alarm.

This callback is triggered every two seconds.

Trigger Timing

This callback is triggered every two seconds.

Parameters

stats

Local audio statistics. See LocalAudioStats.

This callback is triggered when the audio state of a remote user in a voice/video call channel or a host in a live broadcast channel changes, and is used to report the current state of the remote audio stream.

Trigger Timing

This callback is triggered when the audio state of a remote user changes.

Parameters

uid

The user ID of the remote user whose audio state has changed.

state

The remote audio state:

• REMOTE_AUDIO_STATE_STOPPED (0): The remote audio is in the initial state. The SDK reports this state when REMOTE_AUDIO_REASON_LOCAL_MUTED, REMOTE_AUDIO_REASON_REMOTE_MUTED, or REMOTE_AUDIO_REASON_REMOTE_OFFLINE occurs.
• REMOTE_AUDIO_STATE_STARTING (1): The first packet of remote audio data is received.
• REMOTE_AUDIO_STATE_DECODING (2): The remote audio stream is decoded and playing normally. The SDK reports this state when REMOTE_AUDIO_REASON_NETWORK_RECOVERY, REMOTE_AUDIO_REASON_LOCAL_UNMUTED, or REMOTE_AUDIO_REASON_REMOTE_UNMUTED occurs.
• REMOTE_AUDIO_STATE_FROZEN (3): The remote audio is frozen. The SDK reports this state when REMOTE_AUDIO_REASON_NETWORK_CONGESTION occurs.
• REMOTE_AUDIO_STATE_FAILED (4): The remote audio failed to start. The SDK reports this state when REMOTE_AUDIO_REASON_INTERNAL occurs.

reason

The reason for the remote audio state change:

• REMOTE_AUDIO_REASON_INTERNAL (0): The SDK reports this reason when the audio state changes.
• REMOTE_AUDIO_REASON_NETWORK_CONGESTION (1): Network congestion.
• REMOTE_AUDIO_REASON_NETWORK_RECOVERY (2): Network recovery.
• REMOTE_AUDIO_REASON_LOCAL_MUTED (3): The local user stops receiving the remote audio stream or disables the audio module.
• REMOTE_AUDIO_REASON_LOCAL_UNMUTED (4): The local user resumes receiving the remote audio stream or enables the audio module.
• REMOTE_AUDIO_REASON_REMOTE_MUTED (5): The remote user stops sending the audio stream or disables the audio module.
• REMOTE_AUDIO_REASON_REMOTE_UNMUTED (6): The remote user resumes sending the audio stream or enables the audio module.
• REMOTE_AUDIO_REASON_REMOTE_OFFLINE (7): The remote user leaves the channel.

elapsed

The time (ms) from the local user calling joinChannel until the SDK triggers this callback.

This callback is triggered every two seconds. If there are multiple remote users in the channel, the SDK triggers this callback for each user.

Parameters

stats

Statistics of the remote audio stream. See RemoteAudioStats.

Deprecated

This callback is deprecated. Use onRemoteAudioStats instead.

This callback reports transport-layer statistics of remote audio streams, such as packet loss rate and network delay after the local user receives audio packets from a remote user.

Trigger Timing

This callback is triggered every 2 seconds during the call when the local user receives audio packets from a remote user.

Parameters

uid

Remote user ID, i.e., the user sending the audio stream.

delay

Network delay from the remote user to the receiver, in milliseconds.

lost

Packet loss rate of audio from the remote user to the receiver, in percentage (%).

rxKBitRate

Received audio bitrate in Kbps.

This callback is triggered when a remote user calls muteLocalAudioStream.

Trigger Timing

This callback is triggered when a remote user calls muteLocalAudioStream to stop or resume sending audio streams.

Parameters

uid

Remote user ID.

muted

Whether the remote user's audio stream is muted:

• true: Audio stream is muted.
• false: Audio stream is not muted.