2.8.0

General

• Add PitchShiftEffect audio filter
• Add lock exposure support to Camera2 (lockExposure/unlockExposure)
• Add PCM resample method and allow G711 with any audio config
• Fix readUntil entering an infinite busy-loop when the stream reaches EOF
• Fix partial reads on readUInt16/24/32
• Fix isKeyframe comparing flags with == instead of a bitmask
• Fix getStartCodeSize IndexOutOfBounds on buffers shorter than 4 bytes
• Fix setDelay crash with values under 5ms
• Fix BaseSender resizeCache guard and getCacheSize
• Refactor BitBuffer class, fix readUVLC and bit-remaining off-by-one
• Fix UrlParser to support passwords containing ':'
• Fix sockets: isConnected after close, recreate ktor selector on reconnect, UDP receive buffer size
• Fix audio PTS frozen in TimestampMode.BUFFER
• Fix BaseDecoder busy-wait while paused
• Fix AudioDecoder muted buffer size, output size and PCMUtil stereo conversion
• Refactor MicrophoneManager and fix races and shared buffer
• Fix BaseEncoder start order and processInput offset
• Fix Camera2ApiManager getSupportedFps formula
• Fix G711 decode output size
• Fix TextStreamObject crash with empty text
• Fix ViewSurfaceFilterRender always crashing
• Move AndroidViewFilterRender to Kotlin, fix CPU spin and add targetFps
• Fix CropFilterRender divide by zero when cropping a full axis
• Fix EGL lifecycle on stop (OpenGlView / GlStreamInterface)
• Fix GlUtil createTextures offset and disableResources bound
• Fix Sprite/AndroidViewSprite center translation with non-uniform scale
• Fix SpriteGestureController mutating the internal scale point
• Fix CameraRender placeholder leak and TextureLoader recycling the user bitmap
• Fix GifStreamObject guards for empty/invalid frames
• Fix Media3 audio/video sources not surviving stop/start
• Fix keyframe detection in record controllers

RTMP

• Refactor ASC to share in RTMP and RTSP
• Refactor RtmpConfig to avoid share between RtmpClients
• Fix Acknowledgement sequence value
• Fix write extended timestamp
• Fix parse chunk stream id and read message chunk
• Fix RtmpHeader TYPE_1/TYPE_2 timestamp delta
• Avoid send auth command when no user or password
• Fix AuthUtil description value parsing
• Fix CommandSessionHistory leak
• Stop connection if AMF packet is unknown and disable AMF3 instead of crashing
• Fix crash on read Aggregate or SharedObject
• Avoid crash on malformed command
• Fix AmfEcmaArray length, AmfObject read, UserControl length and Command length
• Fix AmfString equals/hashCode to avoid duplicated AMF keys
• Fix AAC/Opus FLV dropping the first audio frame
• Fix opus and aac audio configs
• Fix VideoSpecificConfigAV1 sRGB condition and chroma_sample_position
• Fix first PONG rtt

RTSP

• Refactor RtpTracks to avoid share between RtspClients
• Fix AAC, G711 and Opus fragmented packets case
• Fix AV1 packetizer on long size obus
• Fix digest auth
• Fix NTP timestamp to be unique per session
• Fix parse method command
• Fix loadServerPorts to use the expected track instead of guessing from the response
• Consume the response body when Content-Length is present

SRT

• Fix overflow message number
• Fix PCR/PTS overflow after ~4 days of device uptime
• Fix resend lost packets and discard packets in queue
• Fix packetHandlingQueue growing without bound when ACKs stop (TLPKTDROP by latency)
• Fix send ACK2 only on full ACK packet and support light/small ACKs
• Fix audio-only AAC never emitting PCR
• Fix Nak crash on a truncated/malformed NAK
• Fix ControlPacket control type to 15 bits
• Fix AdaptationField, Pes flags and pes small stuffing
• Fix Psi continuity counter by pid and Psi version to 5 bits
• Fix initial PSI order to [PAT, PMT, SDT]
• Fix Opus packetizer
• Fix read ipaddress on handshake and support handshake with no path
• Fix streamid carrying the query
• Refactor Pid to avoid share between SrtClient instances
• Move latency to millis in CommandsManager as the single source

UDP

• Fix UdpClient leaving isStreaming true when the port is missing
• Fix initial PSI order to [PAT, PMT, SDT]

WHIP

• Add protocol support (beta)