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Qemu audio support v2 #1952
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Qemu audio support v2 #1952
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,173 @@ | ||
| // The RFB protocol (VNC) is designed for real-time user interactions | ||
| // and allows transferring audio messages together with screen content. | ||
| // It is not possible to use any kind of buffering, because that would | ||
| // introduce large delays between user interaction and content display. | ||
| // | ||
| // This is not really a problem with screen content, because the human | ||
| // brain is quite tolerate about slight speed changes in video content, | ||
| // and we mostly transfer non-video data anyways. | ||
| // | ||
| // With audio, the situation is quite different, as it must be played | ||
| // at a constant speed. Any delay leads to audio distortion, which is | ||
| // unpleasant for humans. | ||
| // | ||
| // Without buffering, it is always possible for audio frames to arrive | ||
| // too late or too early due to changing network speeds. | ||
| // | ||
| // We use the following algorithm: | ||
| // | ||
| // - small Jitter buffer to tolerate small speed changes (20ms) | ||
| // - simply discard late audio frame | ||
| // - Queue early frames with slight speedup (pitch scale) to re-sync audio | ||
| // - if we get to many early frames, skip frames for fast re-sync | ||
| // | ||
| // ## Audio format | ||
| // | ||
| // We use/expect U16, little endian, raw audio data, | ||
| // interleaved channel data: [L0, R0, L1, R1, ...] | ||
|
|
||
| import * as Log from './util/logging.js'; | ||
|
|
||
| export default class Audio { | ||
| constructor(sample_rate, nchannels) { | ||
| this._next_start = 0; | ||
| this._context = null; | ||
| this._jitter = 0.02; | ||
| this._resample_trigger = 5*this._jitter; | ||
| this._stable_time = 1.0; | ||
|
|
||
| // ===== PROPERTIES ===== | ||
| this._sample_rate = sample_rate; | ||
| this._nchannels = nchannels; | ||
| this._little_endian = true; | ||
| } | ||
|
|
||
| // ===== PROPERTIES ===== | ||
| get sample_rate() { return this._sample_rate; } | ||
| get nchannels() { return this._nchannels; } | ||
|
|
||
| // ===== PUBLIC METHODS ===== | ||
|
|
||
| // Stop audio playback | ||
| // | ||
| // Further audio frames are simply dropped. | ||
| stop() { | ||
| this._context = null; | ||
| this._next_start = 0; | ||
| } | ||
|
|
||
| start() { | ||
| this._context = new AudioContext({ | ||
| latencyHint: "interactive", | ||
| sampleRate: this._sample_rate, | ||
| }); | ||
| this._next_start = 0; | ||
| } | ||
|
|
||
| play(payload) { | ||
| if (this._context === null) { | ||
| return true; | ||
| } | ||
|
|
||
| let ctime = this._context.currentTime; | ||
|
|
||
| let time_offset = this._next_start - ctime; | ||
|
|
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| let sample_bytes = 2*this._nchannels; | ||
|
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| if ((time_offset < this._jitter) && (this._resample_trigger !== 5*this._jitter)) { | ||
| Log.Debug("Stop resampling because audio is in sync (delay = " + time_offset + " sec)"); | ||
| this._resample_trigger = 5*this._jitter; | ||
| } | ||
|
|
||
| let buffer = null; | ||
| if (time_offset > this._resample_trigger && (payload.length > (100*sample_bytes))) { | ||
| if (this._resample_trigger !== this._jitter) { | ||
| Log.Debug("Start resampling to re-sync audio (delay = " + time_offset + " sec)"); | ||
| this._resample_trigger = this._jitter; | ||
| } | ||
| buffer = this._pitchScale(payload, 1.01); // increase pitch by 1% | ||
| } else { | ||
| buffer = this._createBuffer(payload); | ||
| } | ||
|
|
||
| if (this._next_start > 0) { | ||
| if (time_offset < -buffer.duration) { | ||
| Log.Warn("Skip delayed audio frame (delay = " + (-time_offset) + " sec)"); | ||
| this._next_start = ctime + this._jitter; | ||
| return true; // do not play delayed frame - skip it! | ||
| } | ||
| if (time_offset > 0.5) { | ||
| Log.Warn("Move fast audio frame (offset = " + time_offset + " sec)"); | ||
| this._stable_time = 0; | ||
| return true; // skip frame. | ||
| } | ||
| } | ||
|
|
||
| this._stable_time += buffer.duration; | ||
|
|
||
| if (this._next_start === 0) { | ||
| this._next_start = ctime + this._jitter; | ||
| } | ||
|
|
||
| let start_time = this._next_start; | ||
| this._next_start += buffer.duration; | ||
|
|
||
| if (this._stable_time >= 1.0) { | ||
| let source = this._context.createBufferSource(); | ||
| source.buffer = buffer; | ||
| source.connect(this._context.destination); | ||
| source.start(start_time); | ||
| } | ||
|
|
||
| return true; | ||
| } | ||
|
|
||
| // ===== PRIVATE METHODS ===== | ||
|
|
||
| // see: https://en.wikipedia.org/wiki/Audio_time_stretching_and_pitch_scaling | ||
| _pitchScale(payload, factor) { | ||
| let sample_bytes = 2*this._nchannels; | ||
| let new_length = Math.ceil(payload.length/(factor*sample_bytes)); | ||
| const payload_view = new DataView(payload); | ||
|
|
||
| let buffer = this._context.createBuffer(this._nchannels, new_length, this._sample_rate); | ||
| for (let ch = 0; ch < this._nchannels; ch++) { | ||
| const channel = buffer.getChannelData(ch); | ||
| let channel_offset = ch*2; | ||
| for (let i = 0; i < buffer.length; i++) { | ||
| let pos_float = i*factor; | ||
| let j = Math.trunc(pos_float); | ||
| let second_weight = pos_float % 1; | ||
| let first_weight = 1 - second_weight; | ||
| let p = j*sample_bytes + channel_offset; | ||
| let value0 = payload_view.getUint16(p, this._little_endian); | ||
| p += sample_bytes; | ||
| let value1 = value0; | ||
| if (p < payload.length) { | ||
| value1 = payload_view.getUint16(p, this._little_endian); | ||
| } | ||
| let value = (value0*first_weight + value1*second_weight); | ||
| channel[i] = (value - 32768) / 32768.0; | ||
| } | ||
| } | ||
| return buffer; | ||
| } | ||
|
|
||
| _createBuffer(payload) { | ||
| let sample_bytes = 2*this._nchannels; | ||
| let buffer = this._context.createBuffer( | ||
| this._nchannels, payload.length/sample_bytes, this._sample_rate); | ||
|
|
||
| for (let ch = 0; ch < this._nchannels; ch++) { | ||
| const channel = buffer.getChannelData(ch); | ||
| let channel_offset = ch*2; | ||
| for (let i = 0; i < buffer.length; i++) { | ||
| let p = i*sample_bytes + channel_offset; | ||
| let value = payload[p] + payload[p+1]*256; | ||
| channel[i] = (value / 32768.0) - 1.0; | ||
| } | ||
| } | ||
| return buffer; | ||
| } | ||
| } | ||
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