AVAEInternal.h:76 required condition is false: [AVAudioIONodeImpl.mm:1158:SetOutputFormat: (format.sampleRate...== hwFormat.sampleRate)] 2021-12-15 20:13:30.764037+0800 ***[1659:708411] *** Terminating app due to...uncaught exception ‘com.apple.coreaudio.avfaudio’, reason: ‘required condition is false: format.sampleRate...== hwFormat.sampleRate’ *** First throw call stack: (0x1989799d8 0x1accffb54 0x19888850c 0x1a88e2984...== hwFormat.sampleRate’ terminating with uncaught exception of type NSException *** Terminating app
采样率：sampleRate :，每秒从连续信号中提取并组成离散信号的采样个数，它用赫兹（Hz）来表示。...(1) sampleRate ：可以简单理解为每秒多少个frames。...(2) frames(count) = sampleRate * time (3) bytes = framesize * frames bytes = framesize * sampleRate *
如果要获得很高的性能，官方建议安装libsamplerate和其相应的python模块scikits.samplerate。...源码安装scikits.samplerate 如果要使用pip进行安装的话，系统可能找不到libsamplerate的lib文件，导致安装失败。...下载了scikits.samplerate的源码包之后，需要对site.cfg文件进行配置，制定samplerate库的lib文件名和所在目录才可以正常的安装，不然也会导致失败。...在scikits.samplerate目录下，打开site.cfg文件，修改内容为： [samplerate] library_dirs = c:\libsamplerate-0.1.8 libraries...这里还要注意，安装成功之后，在python的第三方库文件夹中找到scikits.samplerate，在scikits\samplerate文件夹中将libsamplerate编译好的dll文件放进去，
各个平台只接受 16K/8K SampleRate 的音频，JS 录制默认是 44.1K SampleRate，这时需要压缩采样率。...channel count */ view.setUint16(22, numChannels, true); /* sample rate */ view.setUint32(24, sampleRate..., true); /* byte rate (sample rate * block align) */ view.setUint32(28, sampleRate * 4, true)
format.channels = 1; format.sampleRate = (drwav_uint32) sampleRate; format.bitsPerSample = 16...)); for (uint32_t idx = 0; idx < dstSize; idx++) { float index = ((float) idx * sampleRate..., &totalSampleCount); uint32_t out_sampleRate = in_sampleRate * 2; uint32_t out_size = (uint32..._t) (totalSampleCount * ((float) out_sampleRate / in_sampleRate)); int16_t *data_out = (int16_t *...= NULL) { resampleData(data_in, in_sampleRate, (uint32_t) totalSampleCount, data_out, out_sampleRate
= (drwav_uint32) sampleRate; format.bitsPerSample = 16; drwav *pWav = drwav_open_file_write(...)); for (uint32_t idx = 0; idx < dstSize; idx++) { float index = ((float) idx * sampleRate...); uint32_t out_sampleRate = 48000; uint32_t out_size = (uint32_t) (in_size * ((float) out_sampleRate...= NULL) { resampleData(data_in, in_sampleRate, (uint32_t) in_size, data_out, out_sampleRate);...out_size, data_in, in_sampleRate); wavWrite_int16(out_file, data_in, in_sampleRate, (uint32_t
文件转WAV文件 * @param inPcmFilePath 输入PCM文件路径 * @param outWavFilePath 输出WAV文件路径 * @param sampleRate...或16 */ public static void convertPcmToWav(String inPcmFilePath, String outWavFilePath, int sampleRate...totalDataLen = totalAudioLen + 36; writeWaveFileHeader(out, totalAudioLen, totalDataLen, sampleRate...& 0xff); header[25] = (byte) ((sampleRate >> 8) & 0xff); header[26] = (byte) ((sampleRate...>> 16) & 0xff); header[27] = (byte) ((sampleRate >> 24) & 0xff); //音频数据传送速率,采样率*通道数*
如果您希望采样收集错误，比如“只收集30%的错误”，可以将sampleRate属性设为0.3。这样的话，您可以选择更加合适套餐。 1....在HTML中配置标签中配置sampleRate属性 <script src="https://js.fundebug.cn/fundebug.0.4.0.min.js"...apikey="API-KEY" sampleRate=0.3> 2....在JavaScript中配置sampleRate变量 fundebug.sampleRate = 0.3; 注意，是否收集错误是完全随机的，因此理论上这样可能会导致一些错误不会被收集。...因此，您需要自行权衡利弊，选择是否配置sampleRate以及配置多大的sampleRate。 另外，0.4.0的插件还支持过滤特定属性不存在的错误。
format.channels = 1; format.sampleRate = (drwav_uint32) sampleRate; format.bitsPerSample = 16...exit(1); } } } //读取wav文件 int16_t *wavRead_int16(char *filename, uint32_t *sampleRate...unsigned int channels; int16_t *buffer = drwav_open_and_read_file_s16(filename, &channels, sampleRate...= nullptr) { nsProcess(inBuffer, sampleRate, inSampleCount, kVeryHigh); wavWrite_int16...(out_file, inBuffer, sampleRate, inSampleCount); free(inBuffer); } } int main(int argc,
: int, q_factor: float = 1 / sqrt(2) ) -> IIRFilter: w0 = tau * frequency / samplerate _sin =...: int, q_factor: float = 1 / sqrt(2) ) -> IIRFilter: w0 = tau * frequency / samplerate _sin =...: int, q_factor: float = 1 / sqrt(2) ) -> IIRFilter: w0 = tau * frequency / samplerate _sin =...: int ) -> tuple[int | float, int | float]: lowest = min([-20, np.min(fft_results[1: samplerate /...- size) outputs += filler fft_out = np.angle(np.fft.fft(outputs)) plt.xlim(24, samplerate
Lenovo\Desktop\G0001.wav"f = wave.open(wavFile)# 音频头 参数params = f.getparams()Channels = f.getnchannels()SampleRate...= f.getframerate()bit_type = f.getsampwidth() * 8frames = f.getnframes()Duration = frames / float(SampleRate...) # 单位为sprint("音频头参数：", params)print("通道数(Channels)：", Channels)print("采样率(SampleRate)：", SampleRate)...framerate=48000, nframes=171698592, comptype='NONE', compname='not compressed')通道数(Channels)： 1采样率(SampleRate
(A) : (B)) #endif //写wav文件 void wavWrite_int16(char *filename, int16_t *buffer, size_t sampleRate, size_t...format.channels = 1; format.sampleRate = (drwav_uint32) sampleRate; format.bitsPerSample = 16...exit(1); } } } //读取wav文件 int16_t *wavRead_int16(char *filename, uint32_t *sampleRate...unsigned int channels; int16_t *buffer = drwav_open_and_read_file_s16(filename, &channels, sampleRate...(A) : (B)) #endif //读取wav文件 int16_t *wavRead_int16(char *filename, uint32_t *sampleRate, uint64_t *
format.channels = 1; format.sampleRate = sampleRate; format.bitsPerSample = 16; drwav* pWav...unsigned int channels; int16_t* buffer = drwav_open_and_read_file_s16(filename, &channels, sampleRate...= 1) { drwav_free(buffer); buffer = NULL; *sampleRate = 0; *totalSampleCount...<< std::endl; if (argc < 2) return -1; char* in_file = argv[1]; //音频采样率 uint32_t sampleRate...wavBuffer = NULL; double nLoadTime = bench([&] { wavBuffer = wavRead_int16(in_file, &sampleRate
, &totalSampleCount); uint32_t out_sampleRate = in_sampleRate * 2; uint32_t out_size = (uint32..._t) (totalSampleCount * ((float) out_sampleRate / in_sampleRate)); int16_t *data_out = (int16_t *...= NULL) { resampleData(data_in, in_sampleRate, (uint32_t) totalSampleCount, data_out, out_sampleRate...); float speed = 0.88;//增加一个速度权重 uint32_t out_sampleRate = in_sampleRate * speed; uint32_...); //out_sampleRate改为输出一样的采样率in_sampleRate wavWrite_int16(out_file, data_out, in_sampleRate
= 1) { drwav_free(buffer); buffer = NULL; *sampleRate = 0; *totalSampleCount...= 0; } return buffer; } float getGaindB(int16_t *buffer, size_t totalSampleCount, int sampleRate...= 0; uint64_t totalSampleCount = 0; int16_t *wavBuffer = wavRead_int16(in_file, &sampleRate,...= NULL) { size_t analyzeSamples = ref_ms * (sampleRate / 1000); float gain = getGaindB...(wavBuffer, totalSampleCount, sampleRate, analyzeSamples); printf("recommended dB change: %f
* - sampleRate：采样率，仅支持48000。 * - channel：频道数量（如果是立体声，数据是交叉的），单声道：1； 双声道：2。...:(int)sampleRate channels:(int)channels ts:(uint32_t)timestampMs { //拿到PCM数据调用SDK接口去编码和网络传输...TRTCAudioFrame * frame = [[TRTCAudioFrame alloc] init]; frame.data = pcmData; frame.sampleRate...@protocol CustomAudioCapturorDelegate - (void)onAudioCapturePcm:(NSData *)pcmData sampleRate:(int)sampleRate...@protocol CustomAudioFileReaderDelegate - (void)onAudioCapturePcm:(NSData *)pcmData sampleRate:(int)
return (fn(), took + now()) / 1000; } //写wav文件 void wavWrite_int16(char* filename, int16_t* buffer, int sampleRate..., fp); short_number = channels; // 音频通道数 fwrite(&short_number, 2, 1, fp); long_number = sampleRate...; // 采样率 fwrite(&long_number, 4, 1, fp); long_number = sampleRate * nbyte; // 比特率 fwrite(...fclose(fd); return file_buf; } //mp3解码 int16_t* DecodeMp3ToBuffer(char* filename, uint32_t *sampleRate...) *sampleRate = info.hz; if (channels) *channels = info.channels;
=sampleRate) { // 插值去点 buffer = interpolateArray(buffer..., desiredSamplingRate, sampleRate); } buffers.push(buffer);...:sampleRate}); } worker里新增encode函数，处理encode请求，完成后执行回调 self.onmessage = function (e) {...break; } }; encode(cb,buffer,sampleRate...:sampleRate}); } wav上传 增加一个上传函数： exportWAVAndUpload(url, callback) { var _url = url;
, 其 采样波形 与 频谱 如下 : 代码示例 : % wave 文件采样 pcmFileName='music.wav'; % 读取 music.wav 文件 % s1 是读取的信号向量 % sampleRate...是采样率 44100Hz 表示每秒有 44100 个音频样本 [signal,sampleRate] = audioread(pcmFileName); % 上面的 sampleRate 计算出来是...44100 单位是 Hz % 这里需要将 sampleRate 的单位变为 kHz , 除以 1000 sampleRate = sampleRate/1000; % 矩阵转置 % 将 1x132300...))); xf1(1:2048)=xf(2049:4096); xf1(2049:4096)=xf(1:2048); % 绘制实信号的频谱图 figure; plot((-2048:2047) * sampleRate
unsafe.Pointer(uintptr(unsafe.Pointer(p)) + uintptr(8)))}return 0}/* just pick the highest supported samplerate...*/func select_sample_rate(codec *libavcodec.AVCodec) ffcommon.FInt {var p *ffcommon.FIntvar best_samplerate...= 0 {if best_samplerate == 0 || int32(math.Abs(float64(44100-*p))) < int32(math.Abs(float64(44100-best_samplerate...))) {best_samplerate = *p}p = (*int32)(unsafe.Pointer(uintptr(unsafe.Pointer(p)) + uintptr(4)))}return...best_samplerate}/* select layout with the highest channel count */func select_channel_layout(codec *
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