Доброе время суток.
Сталкнулся с небольшой проблемой при конвертации byteArray, полученного с микрофона, в WAV: После конвертации фаил получается обрезанным.
Код AS3:
/*
Copyright (c) 2009, Adobe Systems Incorporated
All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
* Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
* Neither the name of Adobe Systems Incorporated nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS
IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
package com.adobe.audio.format
{
import flash.utils.ByteArray;
import flash.utils.Endian;
import flash.utils.IDataOutput;
/**
* Helper class to write WAV formated audio files. The class
* expects audio input data in a byte array with samples represented
* as floats.
*
* <p>The default compressed code is set to PCM. The class
* resamples and formats the audio samples according to the
* class properties. The resampling geared for performance and
* not quality, for best quality use sampling rates that divide/multiple
* into the desired output <code>samplingRate</code>.
*
* For more information about the WAVE file format see:
* http://ccrma.stanford.edu/courses/422/projects/WaveFormat/</p>
*
* TODO Get WAV's for testing
*
* @langversion ActionScript 3.0
* @playerversion Flash 10.0
*/
public class WAVWriter
{
//-------------------------------------------------------------------
// Variables
//-------------------------------------------------------------------
/**
* @private
* Used for resampling channels where input channels > output channels
*/
private var tempValueSum:Number = 0;
/**
* @private
* Used for resampling channels where input channels > output channels
*/
private var tempValueCount:int = 0;
//-------------------------------------------------------------------
// Properties
//-------------------------------------------------------------------
/**
* The sampling rate, in Hz, for the data in the WAV file.
*
* @default 44100
*/
public var samplingRate:Number = 8000;
/**
* The audio sample bit rate. Has to be set to 8, 16, 24, or 32.
*
* @default 16
*/
public var sampleBitRate:int = 16; // ie: 16 bit wav file
/**
* The number of audio channels in the WAV file.
*
* @default 2
*/
public var numOfChannels:int = 2;
/**
* The WAV header compression code value. The default is the PCM
* code.
*
* @default 1
*/
public var compressionCode:int = 1;
//-------------------------------------------------------------------
// Methods
//-------------------------------------------------------------------
/**
* @private
* Create WAV header bytes
*/
private function header(ba:ByteArray, fileSize:Number):void
{
ba.writeUTFBytes("RIFF");
ba.writeUnsignedInt(fileSize); // Size of whole file
ba.writeUTFBytes("WAVE");
// WAVE Chunk
ba.writeUTFBytes("fmt "); // Chunk ID
ba.writeUnsignedInt(16); // Header Chunk Data Size
ba.writeShort(compressionCode); // Compression code - 1 = PCM
ba.writeShort(numOfChannels); // Number of channels
ba.writeUnsignedInt(samplingRate); // Sample rate
ba.writeUnsignedInt(samplingRate * numOfChannels * sampleBitRate / 8); // Byte Rate == SampleRate * NumChannels * BitsPerSample/8
ba.writeShort(numOfChannels * sampleBitRate / 8); // Block align == NumChannels * BitsPerSample/8
ba.writeShort(sampleBitRate); // Bits Per Sample
}
/**
* Resample the <code>dataInput</code> audio data into the WAV format.
* Writing the output to the <code>dataOutput</code> object.
*
* <p>The <code>dataOutput.endian</code> will be set to <code>Endian.LITTLE_ENDIAN</code>
* with the header and data written out on the data stream. The <code>dataInput</code>
* will set the position = 0 and read all bytes in the <code>ByteArray</code> as samples.
*
*
* </p>
*
* @param dataOutput The IDataOutput object that you want the WAV formated bytes to be written to.
* @param dataInput The audio sample data in float format.
* @param inputSamplingRate The sampling rate of the <code>dataInput</code> data.
* @param inputNumChannels The number of audio changes in <code>dataInput</code> data.
*
*/
public function processSamples(dataInput:ByteArray, inputSamplingRate:int, inputNumChannels:int = 1):ByteArray
{
if (!dataInput || dataInput.bytesAvailable <= 0) // Return if null
throw new Error("No audio data");
var ba:ByteArray = new ByteArray();
// 16 bit values are between -32768 to 32767.
var bitResolution:Number = (Math.pow(2, sampleBitRate)/2)-1;
var soundRate:Number = samplingRate / inputSamplingRate;
var dataByteLength:int = ((dataInput.length/4) * soundRate * sampleBitRate/8);
// data.length is in 4 bytes per float, where we want samples * sampleBitRate/8 for bytes
var fileSize:int = 32 + 8 + dataByteLength;
// WAV format requires little-endian
ba.endian = Endian.LITTLE_ENDIAN;
// RIFF WAVE Header Information
header(ba, fileSize);
// Data Chunk Header
ba.writeUTFBytes("data");
ba.writeUnsignedInt(dataByteLength); // Size of whole file
//ba.writeBytes(dataInput);
//return ba;
// Write data to file
dataInput.position = 0;
var tempData:ByteArray = new ByteArray();
tempData.endian = Endian.LITTLE_ENDIAN;
// Write to file in chunks of converted data.
while (dataInput.bytesAvailable > 0)
{
tempData.clear();
// Resampling logic variables
var minSamples:int = Math.min(dataInput.bytesAvailable/4, 8192);
var readSampleLength:int = minSamples;//Math.floor(minSamples/soundRate);
var resampleFrequency:int = 100; // Every X frames drop or add frames
var resampleFrequencyCheck:int = (soundRate-Math.floor(soundRate))*resampleFrequency;
var soundRateCeil:int = Math.ceil(soundRate);
var soundRateFloor:int = Math.floor(soundRate);
var jlen:int = 0;
var channelCount:int = (numOfChannels-inputNumChannels);
/*
trace("resampleFrequency: " + resampleFrequency + " resampleFrequencyCheck: " + resampleFrequencyCheck
+ " soundRateCeil: " + soundRateCeil + " soundRateFloor: " + soundRateFloor);
*/
var value:Number = 0;
// Assumes data is in samples of float value
for (var i:int = 0;i < readSampleLength;i+=4)
{
value = dataInput.readFloat();
// Check for sanity of float value
if (value > 1 || value < -1)
throw new Error("Audio samples not in float format");
// Special case with 8bit WAV files
if (sampleBitRate == 8)
value = (bitResolution * value) + bitResolution;
else
value = bitResolution * value;
// Resampling Logic for non-integer sampling rate conversions
jlen = (resampleFrequencyCheck > 0 && i % resampleFrequency < resampleFrequencyCheck) ? soundRateCeil : soundRateFloor;
for (var j:int = 0; j < jlen; j++)
{
writeCorrectBits(tempData, value, channelCount);
}
}
ba.writeBytes(tempData);
}
return ba;
}
/**
* @private
* Change the audio sample to the write resolution
*/
private function writeCorrectBits(outputData:ByteArray, value:Number, channels:int):void
{
// Handle case where input channels > output channels. Sum values and divide values
if (channels < 0)
{
if (tempValueCount+channels == 1)
{
value = int(tempValueSum/tempValueCount);
tempValueSum = 0;
tempValueCount = 0;
channels = 1;
}
else
{
tempValueSum += value;
tempValueCount++;
return;
}
}
else
{
channels++;
}
// Now write data according to channels
for (var i:int = 0;i < channels; i++)
{
if (sampleBitRate == 8)
outputData.writeByte(value);
else if (sampleBitRate == 16)
outputData.writeShort(value);
else if (sampleBitRate == 24)
{
outputData.writeByte(value & 0xFF);
outputData.writeByte(value >>> 8 & 0xFF);
outputData.writeByte(value >>> 16 & 0xFF);
}
else if (sampleBitRate == 32)
outputData.writeInt(value);
else
throw new Error("Sample bit rate not supported");
}
}
}
}
Заранее спасибо.
18.03.2010, 22:31
Конвертация ByteArray в WAV
Древовидный вид