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Collections - Formulaire de création rapide
19 février 2013, par
Mis à jour : Février 2013
Langue : français
Type : Image
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Node.js readable maximize throughput/performance for compute intense readable - Writable doesn't pull data fast enough
31 décembre 2022, par flohallGeneral setup


I developed an application using AWS Lambda node.js 14.
I use a custom
ReadableimplementationFrameCreationStreamthat uses node-canvas to draw images, svgs and more on a canvas. This result is then extracted as a raw image buffer in BGRA. A single image buffer contains 1920 * 1080 * 4 Bytes = 8294400 Bytes 8 MB.
This is then piped tostdinof achild_processrunningffmpeg.
ThehighWaterMarkof myReadableinobjectMode:trueis set to 25 so that the internal buffer can use up to 8 MB * 25 = 200 MB.

All this works fine and also doesn't contain too much RAM. But I noticed after some time, that the performance is not ideally.


Performance not optimal


I have an example input that generates a video of 315 frames. If I set
highWaterMarkto a value above 25 the performance increases to the point, when I set to a value of 315 or above.

For some reason
ffmpegdoesn't start to pull any data untilhighWaterMarkis reached. Obviously thats not what I want.ffmpegshould always consume data if minimum 1 frame is cached in theReadableand if it has finished processing the frame before. And theReadableshould produce more frames as longhighWaterMarkisn't reached or the last frame has been reached. So ideally theReadableand theWriteableare busy all the time.

I found another way to improve the speed. If I add a timeout in the
_read()method of theReadableafter let's say every tenth frame for 100 ms. Then theffmpeg-Writablewill use this timeout to write some frames toffmpeg.

It seems like frames aren't passed to
ffmpegduring frame creation because some node.js main thread is busy ?

The fastest result I have if I increase
highWaterMarkabove the amount of frames - which doesn't work for longer videos as this would make the AWS Lambda RAM explode. And this makes the whole streaming idea useless. Using timeouts always gives me stomach pain. Also depending on the execution on different environments a good fitting timeout might differ. Any ideas ?

FrameCreationStream


import canvas from 'canvas';
import {Readable} from 'stream';
import {IMAGE_STREAM_BUFFER_SIZE, PerformanceUtil, RenderingLibraryError, VideoRendererInput} from 'vm-rendering-backend-commons';
import {AnimationAssets, BufferType, DrawingService, FullAnimationData} from 'vm-rendering-library';

/**
 * This is a proper back pressure compatible implementation of readable for a having a stream to read single frames from.
 * Whenever read() is called a new frame is created and added to the stream.
 * read() will be called internally until options.highWaterMark has been reached.
 * then calling read will be paused until one frame is read from the stream.
 */
export class FrameCreationStream extends Readable {

 drawingService: DrawingService;
 endFrameIndex: number;
 currentFrameIndex: number = 0;
 startFrameIndex: number;
 frameTimer: [number, number];
 readTimer: [number, number];
 fullAnimationData: FullAnimationData;

 constructor(animationAssets: AnimationAssets, fullAnimationData: FullAnimationData, videoRenderingInput: VideoRendererInput, frameTimer: [number, number]) {
 super({highWaterMark: IMAGE_STREAM_BUFFER_SIZE, objectMode: true});

 this.frameTimer = frameTimer;
 this.readTimer = PerformanceUtil.startTimer();

 this.fullAnimationData = fullAnimationData;

 this.startFrameIndex = Math.floor(videoRenderingInput.startFrameId);
 this.currentFrameIndex = this.startFrameIndex;
 this.endFrameIndex = Math.floor(videoRenderingInput.endFrameId);

 this.drawingService = new DrawingService(animationAssets, fullAnimationData, videoRenderingInput, canvas);
 console.time("read");
 }

 /**
 * this method is only overwritten for debugging
 * @param size
 */
 read(size?: number): string | Buffer {

 console.log("read("+size+")");
 const buffer = super.read(size);
 console.log(buffer);
 console.log(buffer?.length);
 if(buffer) {
 console.timeLog("read");
 }
 return buffer;
 }

 // _read() will be called when the stream wants to pull more data in.
 // _read() will be called again after each call to this.push(dataChunk) once the stream is ready to accept more data. https://nodejs.org/api/stream.html#readable_readsize
 // this way it is ensured, that even though this.createImageBuffer() is async, only one frame is created at a time and the order is kept
 _read(): void {
 // as frame numbers are consecutive and unique, we have to draw each frame number (also the first and the last one)
 if (this.currentFrameIndex <= this.endFrameIndex) {
 PerformanceUtil.logTimer(this.readTimer, 'WAIT -> READ\t');
 this.createImageBuffer()
 .then(buffer => this.optionalTimeout(buffer))
 // push means adding a buffered raw frame to the stream
 .then((buffer: Buffer) => {
 this.readTimer = PerformanceUtil.startTimer();
 // the following two frame numbers start with 1 as first value
 const processedFrameNumberOfScene = 1 + this.currentFrameIndex - this.startFrameIndex;
 const totalFrameNumberOfScene = 1 + this.endFrameIndex - this.startFrameIndex;
 // the overall frameId or frameIndex starts with frameId 0
 const processedFrameIndex = this.currentFrameIndex;
 this.currentFrameIndex++;
 this.push(buffer); // nothing besides logging should happen after calling this.push(buffer)
 console.log(processedFrameNumberOfScene + ' of ' + totalFrameNumberOfScene + ' processed - full video frameId: ' + processedFrameIndex + ' - buffered frames: ' + this.readableLength);
 })
 .catch(err => {
 // errors will be finally handled, when subscribing to frameCreation stream in ffmpeg service
 // this log is just generated for tracing errors and if for some reason the handling in ffmpeg service doesn't work
 console.log("createImageBuffer: ", err);
 this.emit("error", err);
 });
 } else {
 // push(null) makes clear that this stream has ended
 this.push(null);
 PerformanceUtil.logTimer(this.frameTimer, 'FRAME_STREAM');
 }
 }

 private optionalTimeout(buffer: Buffer): Promise<buffer> {
 if(this.currentFrameIndex % 10 === 0) {
 return new Promise(resolve => setTimeout(() => resolve(buffer), 140));
 }
 return Promise.resolve(buffer);
 }

 // prevent memory leaks - without this lambda memory will increase with every call
 _destroy(): void {
 this.drawingService.destroyStage();
 }

 /**
 * This creates a raw pixel buffer that contains a single frame of the video drawn by the rendering library
 *
 */
 public async createImageBuffer(): Promise<buffer> {

 const drawTimer = PerformanceUtil.startTimer();
 try {
 await this.drawingService.drawForFrame(this.currentFrameIndex);
 } catch (err: any) {
 throw new RenderingLibraryError(err);
 }

 PerformanceUtil.logTimer(drawTimer, 'DRAW -> FRAME\t');

 const bufferTimer = PerformanceUtil.startTimer();
 // Creates a raw pixel buffer, containing simple binary data
 // the exact same information (BGRA/screen ratio) has to be provided to ffmpeg, because ffmpeg cannot detect format for raw input
 const buffer = await this.drawingService.toBuffer(BufferType.RAW);
 PerformanceUtil.logTimer(bufferTimer, 'CANVAS -> BUFFER');

 return buffer;
 }
}
</buffer></buffer>

FfmpegService


import {ChildProcess, execFile} from 'child_process';
import {Readable} from 'stream';
import {FPS, StageSize} from 'vm-rendering-library';
import {
 FfmpegError,
 LOCAL_MERGE_VIDEOS_TEXT_FILE, LOCAL_SOUND_FILE_PATH,
 LOCAL_VIDEO_FILE_PATH,
 LOCAL_VIDEO_SOUNDLESS_MERGE_FILE_PATH
} from "vm-rendering-backend-commons";

/**
 * This class bundles all ffmpeg usages for rendering one scene.
 * FFmpeg is a console program which can transcode nearly all types of sounds, images and videos from one to another.
 */
export class FfmpegService {

 ffmpegPath: string = null;


 constructor(ffmpegPath: string) {
 this.ffmpegPath = ffmpegPath;
 }

 /**
 * Convert a stream of raw images into an .mp4 video using the command line program ffmpeg.
 *
 * @param inputStream an input stream containing images in raw format BGRA
 * @param stageSize the size of a single frame in pixels (minimum is 2*2)
 * @param outputPath the filepath to write the resulting video to
 */
 public imageToVideo(inputStream: Readable, stageSize: StageSize, outputPath: string): Promise<void> {
 const args: string[] = [
 '-f',
 'rawvideo',
 '-r',
 `${FPS}`,
 '-pix_fmt',
 'bgra',
 '-s',
 `${stageSize.width}x${stageSize.height}`,
 '-i',
 // input "-" means input will be passed via pipe (streamed)
 '-',
 // codec that also QuickTime player can understand
 '-vcodec',
 'libx264',
 '-pix_fmt',
 'yuv420p',
 /*
 * "-movflags faststart":
 * metadata at beginning of file
 * needs more RAM
 * file will be broken, if not finished properly
 * higher application compatibility
 * better for browser streaming
 */
 '-movflags',
 'faststart',
 // "-preset ultrafast", //use this to speed up compression, but quality/compression ratio gets worse
 // don't overwrite an existing file here,
 // but delete file in the beginning of execution index.ts
 // (this is better for local testing believe me)
 outputPath
 ];

 return this.execFfmpegPromise(args, inputStream);
 }

 private execFfmpegPromise(args: string[], inputStream?: Readable): Promise<void> {
 const ffmpegServiceSelf = this;
 return new Promise(function (resolve, reject) {
 const executionProcess: ChildProcess = execFile(ffmpegServiceSelf.ffmpegPath, args, (err) => {
 if (err) {
 reject(new FfmpegError(err));
 } else {
 console.log('ffmpeg finished');
 resolve();
 }
 });
 if (inputStream) {
 // it's important to listen on errors of input stream before piping it into the write stream
 // if we don't do this here, we get an unhandled promise exception for every issue in the input stream
 inputStream.on("error", err => {
 reject(err);
 });
 // don't reject promise here as the error will also be thrown inside execFile and will contain more debugging info
 // this log is just generated for tracing errors and if for some reason the handling in execFile doesn't work
 inputStream.pipe(executionProcess.stdin).on("error", err => console.log("pipe stream: " , err));
 }
 });
 }
}
</void></void>

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aacenc : use the decoder’s lcg PRNG
8 octobre 2016, par Rostislav Pehlivanovaacenc : use the decoder’s lcg PRNG
Using lfg was an overkill in this case where the random numbers
were only used for encoder descisions. Should increase result
uniformity between different FPUs and gives a slight speedup.Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
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Using ffmpeg libraries to decode wav audio as PCM samples and display waveform
26 septembre 2016, par Lorenzo MonniI’m using the ffmpeg libraries to process audio files.
I need to decode .wav audio files to make some operations having their samples in an understandable format, i.e. decimal numbers comprised between [-1,1] as a normal audio waveform.
I have written the code for decoding and it’s apparently working well, but when I see the decoded samples it seems something in the sample numbers translation went bad. I paste here only the part of code where I translate the samples from the audio frame in PCM 16 bits :
while(av_read_frame(pFormatCtx, &apkt)>=0) {
if(apkt.stream_index==audioStream->index)
{
// Try to decode the packet into a frame
int frameFinished = 0;
avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &apkt);
int data_size = av_samples_get_buffer_size(&plane_size, aCodecCtx->channels,
aFrame->nb_samples,
aCodecCtx->sample_fmt, 1);
// Some frames rely on multiple packets, so we have to make sure the frame is finished before
// we can use it
if (frameFinished)
{
for(int a=0;a < plane_size/sizeof(int16_t);a++)
{
fprintf(fd,"%d\n",(int16_t*)aFrame->data[a]);
}
}
}
av_free_packet(&apkt);
// Free the packet that was allocated by av_read_frame
}Additional information and issues :
-
the sample_fmt in my allocated AVCodecContext is "AV_SAMPLE_FMT_S16" so the samples numbers should be 16bit signed binaries, I guess if translated in decimal format numbers comprised between -32768 and 32767 (I don’t remember how the problem of disparity between positives and negatives is solved). However when I decode them in int16_t I see much higher numbers that seem to fall in the 32bit signed format (but the file is in 16bit anyway). E.g., the max of my decoded audio (after int16_t translation) is 2044951012 ;
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My .wav file has two channels, but I can’t access both two, if I use extended_data of the audio frame struct pointing to the second channel (index 1) the execution returns a segmentation fault. The same happens with the data pointer. I’m able to recover only one channel, from data[0].
Here is how my audio file decoded with the aforementioned code and saved in a txt looks like :
Here is how the trend of the signal should look like :
If I play my decoded signal the sound shows some similarities with the original audio file, but with a lot of destructing artifacts in it.
Final remarks : ffmpeg documentation and past questions of Stackoverflow are not working well to solve this problem.
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