mirror of https://bitbucket.org/ausocean/av.git
579 lines
16 KiB
Go
579 lines
16 KiB
Go
/*
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NAME
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alsa.go
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AUTHOR
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Alan Noble <alan@ausocean.org>
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Trek Hopton <trek@ausocean.org>
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LICENSE
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This file is Copyright (C) 2019 the Australian Ocean Lab (AusOcean)
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It is free software: you can redistribute it and/or modify them
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under the terms of the GNU General Public License as published by the
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Free Software Foundation, either version 3 of the License, or (at your
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option) any later version.
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It is distributed in the hope that it will be useful, but WITHOUT
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ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
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for more details.
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You should have received a copy of the GNU General Public License in gpl.txt.
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If not, see http://www.gnu.org/licenses.
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*/
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// Package alsa provides access to input from ALSA audio devices.
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package alsa
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import (
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"bytes"
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"errors"
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"fmt"
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"io"
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"os"
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"sync"
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"time"
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yalsa "github.com/yobert/alsa"
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"bitbucket.org/ausocean/av/codec/adpcm"
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"bitbucket.org/ausocean/av/codec/codecutil"
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"bitbucket.org/ausocean/av/codec/pcm"
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"bitbucket.org/ausocean/av/device"
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"bitbucket.org/ausocean/av/revid/config"
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"bitbucket.org/ausocean/utils/logging"
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"bitbucket.org/ausocean/utils/pool"
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)
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const (
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pkg = "alsa: "
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rbTimeout = 100 * time.Millisecond
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rbNextTimeout = 2000 * time.Millisecond
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rbLen = 200
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)
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// "running" means the input goroutine is reading from the ALSA device and writing to the ringbuffer.
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// "paused" means the input routine is sleeping until unpaused or stopped.
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// "stopped" means the input routine is stopped and the ALSA device is closed.
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const (
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running = iota + 1
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paused
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stopped
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)
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const (
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defaultSampleRate = 48000
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defaultBitDepth = 16
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defaultChannels = 1
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defaultRecPeriod = 1.0
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defaultCodec = codecutil.PCM
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)
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// Configuration field errors.
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var (
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errInvalidSampleRate = errors.New("invalid sample rate, defaulting")
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errInvalidChannels = errors.New("invalid number of channels, defaulting")
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errInvalidBitDepth = errors.New("invalid bitdepth, defaulting")
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errInvalidRecPeriod = errors.New("invalid record period, defaulting")
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errInvalidCodec = errors.New("invalid audio codec, defaulting")
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)
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// An ALSA device holds everything we need to know about the audio input stream and implements io.Reader and device.AVDevice.
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type ALSA struct {
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l logging.Logger // Logger for device's routines to log to.
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mode uint8 // Operating mode, either running, paused, or stopped.
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mu sync.Mutex // Provides synchronisation when changing modes concurrently.
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title string // Name of audio title, or empty for the default title.
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dev *yalsa.Device // ALSA device's Audio input device.
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pb pcm.Buffer // Buffer to contain the direct audio from ALSA.
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buf *pool.Buffer // Ring buffer to contain processed audio ready to be read.
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Config // Configuration parameters for this device.
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}
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// Config provides parameters used by the ALSA device.
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type Config struct {
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SampleRate uint
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Channels uint
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BitDepth uint
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RecPeriod float64
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Codec string
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}
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// New initializes and returns an ALSA device which has its logger set as the given logger.
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func New(l logging.Logger) *ALSA { return &ALSA{l: l} }
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// Name returns the name of the device.
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func (d *ALSA) Name() string {
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return "ALSA"
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}
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// Setup will take a Config struct, check the validity of the relevant fields
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// and then perform any configuration necessary. If fields are not valid,
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// an error is added to the multiError and a default value is used.
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// It then initialises the ALSA device which can then be started, read from, and stopped.
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func (d *ALSA) Setup(c config.Config) error {
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var errs device.MultiError
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if c.SampleRate <= 0 {
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errs = append(errs, errInvalidSampleRate)
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c.SampleRate = defaultSampleRate
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}
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if c.Channels <= 0 {
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errs = append(errs, errInvalidChannels)
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c.Channels = defaultChannels
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}
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if c.BitDepth <= 0 {
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errs = append(errs, errInvalidBitDepth)
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c.BitDepth = defaultBitDepth
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}
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if c.RecPeriod <= 0 {
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errs = append(errs, errInvalidRecPeriod)
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c.RecPeriod = defaultRecPeriod
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}
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if c.InputCodec != codecutil.ADPCM && c.InputCodec != codecutil.PCM {
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errs = append(errs, errInvalidCodec)
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c.InputCodec = defaultCodec
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}
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d.Config = Config{
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SampleRate: c.SampleRate,
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Channels: c.Channels,
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BitDepth: c.BitDepth,
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RecPeriod: c.RecPeriod,
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Codec: c.InputCodec,
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}
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// Open the requested audio device.
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err := d.open()
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if err != nil {
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return fmt.Errorf("failed to open device: %w", err)
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}
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// Create a buffer of 1 minute for continuous recordings.
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ab := d.dev.NewBufferDuration(time.Minute)
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sf, err := pcm.SFFromString(ab.Format.SampleFormat.String())
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if err != nil {
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return fmt.Errorf("unable to get sample format from string: %w", err)
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}
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cf := pcm.BufferFormat{
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SFormat: sf,
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Channels: uint(ab.Format.Channels),
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Rate: uint(ab.Format.Rate),
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}
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d.pb = pcm.Buffer{
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Format: cf,
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Data: ab.Data,
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}
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// Create pool buffer with appropriate chunk size.
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cs := d.DataSize()
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d.buf = pool.NewBuffer(rbLen, cs, rbTimeout)
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// Start device in paused mode.
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d.mode = paused
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go d.input()
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if len(errs) != 0 {
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return errs
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}
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return nil
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}
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// Set exists to satisfy the implementation of the Device interface that revid uses.
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// Everything that would usually be in Set is in the Setup function.
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// This is because an ALSA device is different to other devices in that it
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// outputs binary non-packetised data and it requires a different configuration procedure.
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func (d *ALSA) Set(c config.Config) error {
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return nil
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}
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// Start will start recording audio and writing to the ringbuffer.
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// Once an ALSA device has been stopped it cannot be started again. This is likely to change in future.
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func (d *ALSA) Start() error {
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d.mu.Lock()
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mode := d.mode
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d.mu.Unlock()
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switch mode {
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case paused:
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d.mu.Lock()
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d.mode = running
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d.mu.Unlock()
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return nil
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case stopped:
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// TODO(Trek): Make this reopen device and start recording.
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return errors.New("device is stopped")
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case running:
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return nil
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default:
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return fmt.Errorf("invalid mode: %d", mode)
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}
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}
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// Stop will stop recording audio and close the device.
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// Once an ALSA device has been stopped it cannot be started again. This is likely to change in future.
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func (d *ALSA) Stop() error {
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d.mu.Lock()
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d.mode = stopped
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d.mu.Unlock()
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return nil
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}
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// open the recording device with the given name and prepare it to record.
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// If name is empty, the first recording device is used.
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func (d *ALSA) open() error {
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// Close any existing device.
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if d.dev != nil {
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d.l.Debug("closing device", "title", d.title)
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d.dev.Close()
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d.dev = nil
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}
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// Open sound card and open recording device.
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d.l.Debug("opening sound card")
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cards, err := yalsa.OpenCards()
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if err != nil {
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return err
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}
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defer yalsa.CloseCards(cards)
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d.l.Debug("finding audio device")
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for _, card := range cards {
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devices, err := card.Devices()
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if err != nil {
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continue
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}
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for _, dev := range devices {
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if dev.Type != yalsa.PCM || !dev.Record {
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continue
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}
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if dev.Title == d.title || d.title == "" {
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d.dev = dev
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break
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}
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}
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}
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if d.dev == nil {
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return errors.New("no ALSA device found")
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}
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d.l.Debug("opening ALSA device", "title", d.dev.Title)
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err = d.dev.Open()
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if err != nil {
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return err
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}
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// Try to configure device with chosen channels.
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channels, err := d.dev.NegotiateChannels(int(d.Channels))
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if err != nil && d.Channels == 1 {
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d.l.Info("device is unable to record in mono, trying stereo", "error", err)
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channels, err = d.dev.NegotiateChannels(2)
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}
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if err != nil {
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return fmt.Errorf("device is unable to record with requested number of channels: %w", err)
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}
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d.l.Debug("alsa device channels set", "channels", channels)
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// Try to negotiate a rate to record in that is divisible by the wanted rate
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// so that it can be easily downsampled to the wanted rate.
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// rates is a slice of common sample rates including the standard for CD (44100Hz) and standard for professional audio recording (48000Hz).
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// Note: if a card thinks it can record at a rate but can't actually, this can cause a failure.
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// Eg. the audioinjector sound card is supposed to record at 8000Hz and 16000Hz but it can't due to a firmware issue,
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// a fix for this is to remove 8000 and 16000 from the rates slice.
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var rates = [8]int{8000, 16000, 32000, 44100, 48000, 88200, 96000, 192000}
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var rate int
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foundRate := false
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for _, r := range rates {
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if r < int(d.SampleRate) {
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continue
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}
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if r%int(d.SampleRate) == 0 {
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rate, err = d.dev.NegotiateRate(r)
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if err == nil {
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foundRate = true
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d.l.Debug("alsa device sample rate set", "rate", rate)
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break
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}
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}
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}
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// If no easily divisible rate is found, then use the default rate.
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if !foundRate {
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d.l.Warning("unable to sample at requested rate, default used.", "rateRequested", d.SampleRate)
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rate, err = d.dev.NegotiateRate(defaultSampleRate)
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if err != nil {
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return err
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}
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d.l.Debug("alsa device sample rate set", "rate", rate)
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}
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var aFmt yalsa.FormatType
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switch d.BitDepth {
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case 16:
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aFmt = yalsa.S16_LE
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case 32:
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aFmt = yalsa.S32_LE
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default:
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return fmt.Errorf("unsupported sample bits %v", d.BitDepth)
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}
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devFmt, err := d.dev.NegotiateFormat(aFmt)
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if err != nil {
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return err
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}
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var bitdepth int
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switch devFmt {
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case yalsa.S16_LE:
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bitdepth = 16
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case yalsa.S32_LE:
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bitdepth = 32
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default:
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return fmt.Errorf("unsupported sample bits %v", d.BitDepth)
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}
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d.l.Debug("alsa device bit depth set", "bitdepth", bitdepth)
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// A 50ms period is a sensible value for low-ish latency. (this could be made configurable if needed)
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// Some devices only accept even period sizes while others want powers of 2.
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// So we will find the closest power of 2 to the desired period size.
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const wantPeriod = 0.05 //seconds
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bytesPerSecond := rate * channels * (bitdepth / 8)
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wantPeriodSize := int(float64(bytesPerSecond) * wantPeriod)
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nearWantPeriodSize := nearestPowerOfTwo(wantPeriodSize)
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periodSize, err := d.dev.NegotiatePeriodSize(nearWantPeriodSize)
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if err != nil {
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return err
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}
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d.l.Debug("alsa device period size set", "periodsize", periodSize)
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// At least two period sizes should fit within the buffer.
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bufSize, err := d.dev.NegotiateBufferSize(periodSize * 4)
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if err != nil {
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return err
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}
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d.l.Debug("alsa device buffer size set", "buffersize", bufSize)
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if err = d.dev.Prepare(); err != nil {
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return err
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}
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d.l.Debug("successfully negotiated device params")
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return nil
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}
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// input continously records audio and writes it to the ringbuffer.
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// Re-opens the device and tries again if the ASLA device returns an error.
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func (d *ALSA) input() {
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// Make a channel to communicate betwen continuous recording and processing.
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// The channel has a capacity of 5 minutes of audio, which it should never reach.
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ch := make(chan []byte, int(5*60/d.RecPeriod))
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// Read audio in 1 minute sections.
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go chunkingRead(d, ch)
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// Sleep for 1 minute before the first attempt to read.
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d.l.Debug("Sleeping before read", "sleep time", "1 minute")
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time.Sleep(time.Minute)
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for {
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// Check mode.
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d.mu.Lock()
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mode := d.mode
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d.mu.Unlock()
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switch mode {
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case paused:
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time.Sleep(time.Duration(d.RecPeriod) * time.Second)
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continue
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case stopped:
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if d.dev != nil {
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d.l.Debug("closing ALSA device", "title", d.title)
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d.dev.Close()
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d.dev = nil
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}
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err := d.buf.Close()
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if err != nil {
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d.l.Error("unable to close pool buffer", "error", err)
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}
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return
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}
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// Read audio chunk from channel.
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d.l.Debug("recording audio for period", "seconds", d.RecPeriod)
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d.pb.Data = <-ch
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d.l.Debug("read audio from channel", "length read", len(d.pb.Data))
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d.l.Debug("first bytes of audio", "bytes [0:32]", fmt.Sprintf("%x", d.pb.Data[0:4]))
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// Process audio.
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d.l.Debug("processing audio")
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toWrite := d.formatBuffer()
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// Write audio to ringbuffer.
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n, err := d.buf.Write(toWrite.Data)
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switch err {
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case nil:
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d.l.Debug("wrote audio to ringbuffer", "length", n)
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case pool.ErrDropped:
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d.l.Warning("old audio data overwritten")
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default:
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d.l.Error("unexpected ringbuffer error", "error", err.Error())
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}
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}
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}
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// chunkingRead reads continuously from the ALSA buffer in 1 minute sections. The
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// audio is then chunked into the recording period set by d.RecPeriod and sent over
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// the channel.
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func chunkingRead(d *ALSA, ch chan []byte) {
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d.l.Debug("Datasize of recperiod", "datasize", d.DataSize())
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for {
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buf := d.dev.NewBufferDuration(time.Minute)
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// Read audio in 1 minute sections.
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d.l.Debug("Reading audio for 1 minute")
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err := d.dev.Read(buf.Data)
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if err != nil {
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d.l.Debug("read failed", "error", err.Error())
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err = d.open() // re-open
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if err != nil {
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d.l.Fatal("reopening device failed", "error", err.Error())
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return
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}
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continue
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}
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file, err := os.Create("formatted.pcm")
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if err != nil {
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d.l.Error("unable to create output file", "error", err)
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continue
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}
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_, err = file.Write(buf.Data)
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if err != nil {
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d.l.Error("unable to write formatted audio to file", "error", err)
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continue
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}
|
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|
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// Chunk the audio into length of RecPeriod.
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for i := 0; i < len(buf.Data); i += d.DataSize() {
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ch <- buf.Data[i:(i + d.DataSize())]
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}
|
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}
|
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}
|
|
|
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// Read reads from the ringbuffer, returning the number of bytes read upon success.
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func (d *ALSA) Read(p []byte) (int, error) {
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// Ready ringbuffer for read.
|
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d.l.Debug(pkg + "getting next chunk ready")
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_, err := d.buf.Next(rbNextTimeout)
|
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if err != nil {
|
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switch err {
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case io.EOF:
|
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d.l.Debug(pkg + "EOF from Next")
|
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return 0, err
|
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case pool.ErrTimeout:
|
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d.l.Debug(pkg + "pool buffer timeout")
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return 0, err
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default:
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d.l.Error(pkg+"unexpected error from Next", "error", err.Error())
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return 0, err
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}
|
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}
|
|
|
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// Read from pool buffer.
|
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d.l.Debug(pkg + "reading from buffer")
|
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n, err := d.buf.Read(p)
|
|
if err != nil {
|
|
switch err {
|
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case io.EOF:
|
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d.l.Debug(pkg + "EOF from Read")
|
|
return n, err
|
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default:
|
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d.l.Error(pkg+"unexpected error from Read", "error", err.Error())
|
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return n, err
|
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}
|
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}
|
|
d.l.Debug(fmt.Sprintf("%v read %v bytes", pkg, n))
|
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return n, nil
|
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}
|
|
|
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// formatBuffer returns audio that has been converted to the desired format.
|
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func (d *ALSA) formatBuffer() pcm.Buffer {
|
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var err error
|
|
|
|
// If nothing needs to be changed, return the original.
|
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if d.pb.Format.Channels == d.Channels && d.pb.Format.Rate == d.SampleRate {
|
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d.l.Debug("doing nothing in formatBuffer")
|
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return d.pb
|
|
}
|
|
var formatted pcm.Buffer
|
|
if d.pb.Format.Channels != d.Channels {
|
|
// Convert channels.
|
|
// TODO(Trek): Make this work for conversions other than stereo to mono.
|
|
if d.pb.Format.Channels == 2 && d.Channels == 1 {
|
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formatted, err = pcm.StereoToMono(d.pb)
|
|
if err != nil {
|
|
d.l.Fatal("channel conversion failed", "error", err.Error())
|
|
}
|
|
}
|
|
}
|
|
|
|
if d.pb.Format.Rate != d.SampleRate {
|
|
// Convert rate.
|
|
formatted, err = pcm.Resample(formatted, d.SampleRate)
|
|
if err != nil {
|
|
d.l.Fatal("rate conversion failed", "error", err.Error())
|
|
}
|
|
}
|
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|
|
switch d.Codec {
|
|
case codecutil.PCM:
|
|
case codecutil.ADPCM:
|
|
b := bytes.NewBuffer(make([]byte, 0, adpcm.EncBytes(len(formatted.Data))))
|
|
enc := adpcm.NewEncoder(b)
|
|
_, err = enc.Write(formatted.Data)
|
|
if err != nil {
|
|
d.l.Fatal("unable to encode", "error", err.Error())
|
|
}
|
|
formatted.Data = b.Bytes()
|
|
default:
|
|
d.l.Error("unhandled audio codec")
|
|
}
|
|
|
|
return formatted
|
|
}
|
|
|
|
// DataSize returns the size in bytes of the data ALSA device d will
|
|
// output in the duration of a single recording period.
|
|
func (d *ALSA) DataSize() int {
|
|
s := pcm.DataSize(d.SampleRate, d.Channels, d.BitDepth, d.RecPeriod)
|
|
if d.Codec == codecutil.ADPCM {
|
|
s = adpcm.EncBytes(s)
|
|
}
|
|
return s
|
|
}
|
|
|
|
// nearestPowerOfTwo finds and returns the nearest power of two to the given integer.
|
|
// If the lower and higher power of two are the same distance, it returns the higher power.
|
|
// For negative values, 1 is returned.
|
|
// Source: https://stackoverflow.com/a/45859570
|
|
func nearestPowerOfTwo(n int) int {
|
|
if n <= 0 {
|
|
return 1
|
|
}
|
|
if n == 1 {
|
|
return 2
|
|
}
|
|
v := n
|
|
v--
|
|
v |= v >> 1
|
|
v |= v >> 2
|
|
v |= v >> 4
|
|
v |= v >> 8
|
|
v |= v >> 16
|
|
v++ // higher power of 2
|
|
x := v >> 1 // lower power of 2
|
|
if (v - n) > (n - x) {
|
|
return x
|
|
}
|
|
return v
|
|
}
|
|
|
|
// IsRunning is used to determine if the ALSA device is running.
|
|
func (d *ALSA) IsRunning() bool {
|
|
return d.mode == running
|
|
}
|