Actionscript 中的录音文件损坏问题
Audio recorded file corrupted issue in Actionscript
我使用 Adobe Flash Builder 4.6 / AIR 从我的麦克风录制了语音样本，语音录制成功。我首先在 actionscript 中将语音数据（字节数组）转换为 base64 格式，然后使用我的 PHP 代码将该 base64 数据转换为 WAV 文件。但是那个 WAV 文件在 RiffPad 中抛出了文件损坏的问题。
RIFFPad 是 RIFF 格式文件（如 WAV、AVI）的查看器。
预期的 wav 文件规格：
采样率：22KHZ
// -- saves the current audio data as a .wav file
protected function onSubmit( event:Event ):void {
alertBox.show("Processing ... please wait.");
stopPlayback();
stopRecording();
playBtn.enabled = recordBtn.enabled = submitBtn.enabled = false;
var position:int = capture.buffer.position;
var wavWriter:WAVWriter = new WAVWriter()
var wavWriter1:WaveEncoder = new WaveEncoder()
wavWriter.numOfChannels = 1;
wavWriter.samplingRate = 22050;
wavWriter.sampleBitRate = 16;
var wavBytes:ByteArray = new ByteArray;
capture.buffer.position = 0;
wavWriter.processSamples(wavBytes, capture.buffer, capture.microphone.rate * 1000, 1);
Settings.alertBox3.show("RATE :"+capture.microphone.rate); //Here show RATE: 8
//wavWriter.processSamples(wavBytes, capture.buffer, 22050, 1);
//wavBytes = wavWriter1.encode( capture.buffer, 1, 16, 22050);
capture.buffer.position = position;
wavBytes.position=0;
submitVoiceSample(Base64_new.encodeByteArray(wavBytes));
}
WAV Writer 头函数:
public var samplingRate = 22050;
public var sampleBitRate:int = 8;
public var numOfChannels:int = 2;
private var compressionCode:int = 1;
private function header(dataOutput:IDataOutput, fileSize:Number):void
{
dataOutput.writeUTFBytes("RIFF");
dataOutput.writeUnsignedInt(fileSize); // Size of whole file
dataOutput.writeUTFBytes("WAVE");
// WAVE Chunk
dataOutput.writeUTFBytes("fmt "); // Chunk ID
dataOutput.writeUnsignedInt(16); // Header Chunk Data Size
dataOutput.writeShort(compressionCode); // Compression code - 1 = PCM
dataOutput.writeShort(numOfChannels); // Number of channels
dataOutput.writeUnsignedInt(samplingRate); // Sample rate
dataOutput.writeUnsignedInt(samplingRate * numOfChannels * sampleBitRate / 8); // Byte Rate == SampleRate * NumChannels * BitsPerSample/8
dataOutput.writeShort(numOfChannels * sampleBitRate / 8); // Block align == NumChannels * BitsPerSample/8
dataOutput.writeShort(sampleBitRate); // Bits Per Sample
}
WAV 文件写入函数:
public function processSamples(dataOutput:IDataOutput, dataInput:ByteArray, inputSamplingRate:int, inputNumChannels:int = 1):void
{
if (!dataInput || dataInput.bytesAvailable <= 0) // Return if null
throw new Error("No audio data");
// 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;
var fileSize:int = 32 + 4 + dataByteLength;
// WAV format requires little-endian
dataOutput.endian = Endian.LITTLE_ENDIAN;
// RIFF WAVE Header Information
header(dataOutput, fileSize);
// Data Chunk Header
dataOutput.writeUTFBytes("data");
dataOutput.writeUnsignedInt(dataByteLength); // Size of whole file
// 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);
}
}
dataOutput.writeBytes(tempData);
}
}
我将该 base64 数据发送到我的服务请求
php 我得到“$this->request->voiceSample”参数并将 base64 解码为 .wav 文件
file_put_contents('name.wav', base64_decode($this->request->voiceSample));
在 Riffpad 中加载 "name.wav" 文件后
我有问题
There is extra junk at the end of the file.
任何人请给我解决这个问题的建议...
此行存在固有错误：
wavWriter.processSamples(wavBytes, capture.buffer, capture.microphone.rate * 1000, 1);
Microphone.rate 手册指出实际采样频率不同于此代码预期的 microphone.rate*1000。实际的table如下：
rate Actual frequency
44 44,100 Hz
22 22,050 Hz
11 11,025 Hz
8 8,000 Hz
5 5,512 Hz
因此，虽然您的代码注释指出 rate 被报告为 8，但在客户端通常情况可能并非如此，因此在将推导的采样率传递给 [= 之前执行查找16=].
接下来，您通过浮点计算来预计算dataByteLength，这可能最终会不准确，因为您随后会逐字节采样数据，因此最好先重新采样，然后收集数据长度，然后再收集将所有数据写入dataOutput，像这样：
public function processSamples(dataOutput:IDataOutput, dataInput:ByteArray, inputSamplingRate:int, inputNumChannels:int = 1):void
{
if (!dataInput || dataInput.bytesAvailable <= 0) // Return if null
throw new Error("No audio data");
// 16 bit values are between -32768 to 32767.
var bitResolution:Number = (Math.pow(2, sampleBitRate)/2)-1;
// var soundRate:Number = samplingRate / inputSamplingRate;
// var fileSize:int = 32 + 4 + dataByteLength; kept for reference
// fmt tag is 4+4+16, data header is 8 bytes in size, and 4 bytes for WAVE
// but the data length is not yet determined
// WAV format requires little-endian
dataOutput.endian = Endian.LITTLE_ENDIAN;
// Prepare data for data to file
dataInput.position = 0;
var tempData:ByteArray = new ByteArray();
tempData.endian = Endian.LITTLE_ENDIAN;
// Writing in chunks is no longer possible, because we don't have the header ready
// Let's precalculate the data needed in the loop
var step:Number=inputSamplingRate / samplingRate; // how far we should step into the input data to get next sample
var totalOffset:Number=1.0-1e-8; // accumulator for step
var oldChannels:Array=[];
var i:int;
for (i=0;i<numOfChannels;i++) oldChannels.push(0.0);
// previous channels' sample holder
var newChannels:Array=oldChannels.slice(); // same for new channels that are to be read from byte array
// reading first sample set from input byte array
if (dataInput.bytesAvailable>=inputNumChannels*4) {
for (i=0;i<inputNumChannels;i++) {
var buf:Number=dataInput.readFloat();
if (buf > 1) buf=1; if (buf < -1) buf=-1;
newChannels[i]=buf;
}
// if there's one channel, copy data to other channels
if ((inputNumChannels==1) && (numOfChannels>1)) {
for (i=1;i<numOfChannels;i++) newChannels[i]=newChannels[0];
}
}
while ((dataInput.bytesAvailable>=inputNumChannels*4) || (totalOffset<1.0))
{
// sample next value for output wave file
var value:Number;
for (i=0;i<numOfChannels;i++) {
value = (totalOffset*newChannels[i])+(1.0-totalOffset)*oldChannels[i];
// linear interpolation between old sample and new sample
// Special case with 8bit WAV files
if (sampleBitRate == 8)
value = (bitResolution * value) + bitResolution;
else
value = bitResolution * value;
// writing one channel into tempData
writeCorrectBits(tempData, value, 0);
}
totalOffset+=step; // advance per output sample
while ((totalOffset>1) && (dataInput.bytesAvailable>=inputNumChannels*4)) {
// we need a new sample, and have a sample to process in input
totalOffset-=1;
for (i=0;i<numOfChannels;i++) oldChannels[i]=newChannels[i]; // store old sample
// get another sample, copypasted from above
for (i=0;i<inputNumChannels;i++) {
value=dataInput.readFloat();
if (value > 1) value=1; if (value < -1) value=-1; // sanity check
// I made it clip instead of throwing exception, replace if necessary
// if (value > 1 || value < -1) throw new Error("Audio samples not in float format");
newChannels[i]=value;
}
if ((inputNumChannels==1) && (numOfChannels>1)) {
for (i=1;i<numOfChannels;i++) newChannels[i]=newChannels[0];
}
} // end advance by totalOffset
} // end main loop
var dataBytesLength:uint=tempData.length; // now the length will be correct by definition
header(dataOutput, 32+4+dataBytesLength);
dataOutput.writeUTFBytes("data");
dataOutput.writeUnsignedInt(dataBytesLength);
dataOutput.writeBytes(tempData);
}
我重写了重采样例程以使用滑动 window 算法（如果新采样率高于旧采样率则效果最佳，但接受任何比率）。该算法在样本之间使用线性插值，而不是在插值序列的长度上使用 re-using 旧样本。随意替换为您自己的循环。应该保留的原则是你首先编译 full tempData 然后才写 header 现在正确定义的数据长度。
如有任何问题，请报告。
我使用 Adobe Flash Builder 4.6 / AIR 从我的麦克风录制了语音样本，语音录制成功。我首先在 actionscript 中将语音数据（字节数组）转换为 base64 格式，然后使用我的 PHP 代码将该 base64 数据转换为 WAV 文件。但是那个 WAV 文件在 RiffPad 中抛出了文件损坏的问题。
RIFFPad 是 RIFF 格式文件（如 WAV、AVI）的查看器。
预期的 wav 文件规格：
采样率：22KHZ
// -- saves the current audio data as a .wav file
protected function onSubmit( event:Event ):void {
alertBox.show("Processing ... please wait.");
stopPlayback();
stopRecording();
playBtn.enabled = recordBtn.enabled = submitBtn.enabled = false;
var position:int = capture.buffer.position;
var wavWriter:WAVWriter = new WAVWriter()
var wavWriter1:WaveEncoder = new WaveEncoder()
wavWriter.numOfChannels = 1;
wavWriter.samplingRate = 22050;
wavWriter.sampleBitRate = 16;
var wavBytes:ByteArray = new ByteArray;
capture.buffer.position = 0;
wavWriter.processSamples(wavBytes, capture.buffer, capture.microphone.rate * 1000, 1);
Settings.alertBox3.show("RATE :"+capture.microphone.rate); //Here show RATE: 8
//wavWriter.processSamples(wavBytes, capture.buffer, 22050, 1);
//wavBytes = wavWriter1.encode( capture.buffer, 1, 16, 22050);
capture.buffer.position = position;
wavBytes.position=0;
submitVoiceSample(Base64_new.encodeByteArray(wavBytes));
}
WAV Writer 头函数:
public var samplingRate = 22050;
public var sampleBitRate:int = 8;
public var numOfChannels:int = 2;
private var compressionCode:int = 1;
private function header(dataOutput:IDataOutput, fileSize:Number):void
{
dataOutput.writeUTFBytes("RIFF");
dataOutput.writeUnsignedInt(fileSize); // Size of whole file
dataOutput.writeUTFBytes("WAVE");
// WAVE Chunk
dataOutput.writeUTFBytes("fmt "); // Chunk ID
dataOutput.writeUnsignedInt(16); // Header Chunk Data Size
dataOutput.writeShort(compressionCode); // Compression code - 1 = PCM
dataOutput.writeShort(numOfChannels); // Number of channels
dataOutput.writeUnsignedInt(samplingRate); // Sample rate
dataOutput.writeUnsignedInt(samplingRate * numOfChannels * sampleBitRate / 8); // Byte Rate == SampleRate * NumChannels * BitsPerSample/8
dataOutput.writeShort(numOfChannels * sampleBitRate / 8); // Block align == NumChannels * BitsPerSample/8
dataOutput.writeShort(sampleBitRate); // Bits Per Sample
}
WAV 文件写入函数:
public function processSamples(dataOutput:IDataOutput, dataInput:ByteArray, inputSamplingRate:int, inputNumChannels:int = 1):void
{
if (!dataInput || dataInput.bytesAvailable <= 0) // Return if null
throw new Error("No audio data");
// 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;
var fileSize:int = 32 + 4 + dataByteLength;
// WAV format requires little-endian
dataOutput.endian = Endian.LITTLE_ENDIAN;
// RIFF WAVE Header Information
header(dataOutput, fileSize);
// Data Chunk Header
dataOutput.writeUTFBytes("data");
dataOutput.writeUnsignedInt(dataByteLength); // Size of whole file
// 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);
}
}
dataOutput.writeBytes(tempData);
}
}
我将该 base64 数据发送到我的服务请求 php 我得到“$this->request->voiceSample”参数并将 base64 解码为 .wav 文件
file_put_contents('name.wav', base64_decode($this->request->voiceSample));
在 Riffpad 中加载 "name.wav" 文件后 我有问题
There is extra junk at the end of the file.
任何人请给我解决这个问题的建议...
此行存在固有错误：
wavWriter.processSamples(wavBytes, capture.buffer, capture.microphone.rate * 1000, 1);
Microphone.rate 手册指出实际采样频率不同于此代码预期的 microphone.rate*1000。实际的table如下：
rate Actual frequency
44 44,100 Hz
22 22,050 Hz
11 11,025 Hz
8 8,000 Hz
5 5,512 Hz
因此，虽然您的代码注释指出 rate 被报告为 8，但在客户端通常情况可能并非如此，因此在将推导的采样率传递给 [= 之前执行查找16=].
接下来，您通过浮点计算来预计算dataByteLength，这可能最终会不准确，因为您随后会逐字节采样数据，因此最好先重新采样，然后收集数据长度，然后再收集将所有数据写入dataOutput，像这样：
public function processSamples(dataOutput:IDataOutput, dataInput:ByteArray, inputSamplingRate:int, inputNumChannels:int = 1):void
{
if (!dataInput || dataInput.bytesAvailable <= 0) // Return if null
throw new Error("No audio data");
// 16 bit values are between -32768 to 32767.
var bitResolution:Number = (Math.pow(2, sampleBitRate)/2)-1;
// var soundRate:Number = samplingRate / inputSamplingRate;
// var fileSize:int = 32 + 4 + dataByteLength; kept for reference
// fmt tag is 4+4+16, data header is 8 bytes in size, and 4 bytes for WAVE
// but the data length is not yet determined
// WAV format requires little-endian
dataOutput.endian = Endian.LITTLE_ENDIAN;
// Prepare data for data to file
dataInput.position = 0;
var tempData:ByteArray = new ByteArray();
tempData.endian = Endian.LITTLE_ENDIAN;
// Writing in chunks is no longer possible, because we don't have the header ready
// Let's precalculate the data needed in the loop
var step:Number=inputSamplingRate / samplingRate; // how far we should step into the input data to get next sample
var totalOffset:Number=1.0-1e-8; // accumulator for step
var oldChannels:Array=[];
var i:int;
for (i=0;i<numOfChannels;i++) oldChannels.push(0.0);
// previous channels' sample holder
var newChannels:Array=oldChannels.slice(); // same for new channels that are to be read from byte array
// reading first sample set from input byte array
if (dataInput.bytesAvailable>=inputNumChannels*4) {
for (i=0;i<inputNumChannels;i++) {
var buf:Number=dataInput.readFloat();
if (buf > 1) buf=1; if (buf < -1) buf=-1;
newChannels[i]=buf;
}
// if there's one channel, copy data to other channels
if ((inputNumChannels==1) && (numOfChannels>1)) {
for (i=1;i<numOfChannels;i++) newChannels[i]=newChannels[0];
}
}
while ((dataInput.bytesAvailable>=inputNumChannels*4) || (totalOffset<1.0))
{
// sample next value for output wave file
var value:Number;
for (i=0;i<numOfChannels;i++) {
value = (totalOffset*newChannels[i])+(1.0-totalOffset)*oldChannels[i];
// linear interpolation between old sample and new sample
// Special case with 8bit WAV files
if (sampleBitRate == 8)
value = (bitResolution * value) + bitResolution;
else
value = bitResolution * value;
// writing one channel into tempData
writeCorrectBits(tempData, value, 0);
}
totalOffset+=step; // advance per output sample
while ((totalOffset>1) && (dataInput.bytesAvailable>=inputNumChannels*4)) {
// we need a new sample, and have a sample to process in input
totalOffset-=1;
for (i=0;i<numOfChannels;i++) oldChannels[i]=newChannels[i]; // store old sample
// get another sample, copypasted from above
for (i=0;i<inputNumChannels;i++) {
value=dataInput.readFloat();
if (value > 1) value=1; if (value < -1) value=-1; // sanity check
// I made it clip instead of throwing exception, replace if necessary
// if (value > 1 || value < -1) throw new Error("Audio samples not in float format");
newChannels[i]=value;
}
if ((inputNumChannels==1) && (numOfChannels>1)) {
for (i=1;i<numOfChannels;i++) newChannels[i]=newChannels[0];
}
} // end advance by totalOffset
} // end main loop
var dataBytesLength:uint=tempData.length; // now the length will be correct by definition
header(dataOutput, 32+4+dataBytesLength);
dataOutput.writeUTFBytes("data");
dataOutput.writeUnsignedInt(dataBytesLength);
dataOutput.writeBytes(tempData);
}
我重写了重采样例程以使用滑动 window 算法（如果新采样率高于旧采样率则效果最佳，但接受任何比率）。该算法在样本之间使用线性插值，而不是在插值序列的长度上使用 re-using 旧样本。随意替换为您自己的循环。应该保留的原则是你首先编译 full tempData 然后才写 header 现在正确定义的数据长度。
如有任何问题，请报告。