Documentation
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Overview ¶
Package opus wraps the Opus codec for jargo's audio path: it decodes Opus packets to PCM on the way in and encodes PCM to Opus packets on the way out.
The codec runs at 48 kHz — the rate WebRTC negotiates for Opus — so audio crosses the transport without resampling. Encoding produces 20 ms frames, the standard WebRTC packetization.
The decoder is pure Go. The Encoder has two builds selected by the `libopus` build tag (see encoder_pion.go / encoder_libopus.go): the default is the pure-Go encoder; `-tags libopus` links the C library for higher speech quality. Both expose the same NewEncoder/Encode API.
Index ¶
Constants ¶
const ( // SampleRate is the codec sample rate in Hz. Opus runs at 48 kHz. SampleRate = 48000 // FrameDuration is the duration of one encoded packet. FrameDuration = 20 * time.Millisecond // FrameSamples is the number of samples per channel in one 20 ms frame. FrameSamples = SampleRate / 1000 * 20 // 960 )
Variables ¶
This section is empty.
Functions ¶
func FrameBytes ¶
FrameBytes is the size in bytes of one 20 ms S16LE frame for channels.
Types ¶
type Decoder ¶
type Decoder struct {
// contains filtered or unexported fields
}
Decoder decodes Opus packets into signed 16-bit little-endian PCM.
func NewDecoder ¶
NewDecoder builds a Decoder that outputs channels-channel 48 kHz PCM. It decodes all Opus modes (SILK, CELT and hybrid).
type Encoder ¶
type Encoder struct {
// contains filtered or unexported fields
}
Encoder encodes 48 kHz S16LE PCM into Opus packets using the pure-Go pion SILK encoder. This is the default build — SILK is tuned for speech and needs no cgo. The 48 kHz input is downmixed to mono and resampled to 16 kHz with the pure-Go resampler, then encoded as wideband SILK. Build with `-tags libopus` for the C library instead; both expose the same NewEncoder/Encode API.
func NewEncoder ¶
func NewEncoder(cfg EncoderConfig) (*Encoder, error)
NewEncoder builds a SILK Encoder for 48 kHz audio from cfg. EncoderConfig.InbandFEC is ignored: this encoder emits no FEC redundancy.
type EncoderConfig ¶ added in v0.1.0
type EncoderConfig struct {
// Channels is the number of channels in the PCM handed to Encode.
Channels int
// Bitrate is the target bitrate in bits per second; 0 uses the codec
// default. Speech is transparent well below the codec's fullband range, so
// raising this past ~32 kbps for a single voice mostly costs bandwidth.
Bitrate int
// InbandFEC embeds a low-rate copy of the previous frame in each packet so
// the receiver can reconstruct a dropped one instead of concealing it. It
// trades a little of the bitrate budget for resilience, which is worth it
// on lossy links (mobile radio, relayed media) and inert on clean ones,
// where the redundancy is simply never decoded. Honored by the libopus
// build; the pure-Go SILK encoder ignores it.
InbandFEC bool
// ExpectedPacketLoss is the loss percentage (0-100) the encoder sizes its
// FEC redundancy for. Ignored unless InbandFEC is set.
ExpectedPacketLoss int
}
EncoderConfig configures an Encoder.
Source Files