pcm

Android中播放DSD音乐

由 ♀尐吖头ヾ 提交于 2020-01-14 04:49:27
Github上有个简单的Alsa DSD测试程序，可以播放DSD，地址位于：https://github.com/zonque/alsa-dsd-player 细看其代码，发现有ALSA_FORMAT_SND_PCM_FORMAT_DSD_U8的定义，详情如下： #if 1 /* 8-bit DSD */ #define ALSA_FORMAT SND_PCM_FORMAT_DSD_U8 #define SAMPLE_SIZE (sizeof(uint8_t) * 2) #define SAMPLE_RATE_DIV 1 #else /* 16-bit DSD */ #define ALSA_FORMAT SND_PCM_FORMAT_DSD_U16 #define SAMPLE_SIZE (sizeof(uint16_t) * 2) #define SAMPLE_RATE_DIV 2 #endif 那么在Android中能运行这个测试程序吗？ALSA_FORMAT_SND_PCM_FORMAT_DSD_U8似乎是亮点，之前没有看到过。尝试在Android的源码中寻找它的定义，没找到。但是在标准的alsa-lib中找到了： typedef enum _snd_pcm_format { 125 SND_PCM_FORMAT_UNKNOWN = -1, 127 SND_PCM

How to play non-PCM file or convert it to PCM on the fly ?

由 Deadly 提交于 2020-01-11 09:32:24
问题 The following code works with some wav files, but with others I get, "InvalidOperationException was unhandled. Message=Sound API only supports playing PCM wave files." var webClient = new WebClient(); webClient.DownloadFile(url, fileName); var fileSound = new SoundPlayer(fileName); fileSound.PlaySync(); Is there a way to programmatically check if a wav file is "bad" (not a PCM wave file) and then convert it as necessary? What is odd is that the code works in the legacy Delphi app - all of the

How to play non-PCM file or convert it to PCM on the fly ?

由 早过忘川 提交于 2020-01-11 09:32:14
问题 The following code works with some wav files, but with others I get, "InvalidOperationException was unhandled. Message=Sound API only supports playing PCM wave files." var webClient = new WebClient(); webClient.DownloadFile(url, fileName); var fileSound = new SoundPlayer(fileName); fileSound.PlaySync(); Is there a way to programmatically check if a wav file is "bad" (not a PCM wave file) and then convert it as necessary? What is odd is that the code works in the legacy Delphi app - all of the

Unity处理MP3流播放

由 拟墨画扇 提交于 2020-01-10 08:19:46
Unity处理MP3流播放 PCMReaderCallback回调 Unity音频数据是通过AudioClip去处理的，它提供了PCMReaderCallback回调，用于加载流音频数据。它的声明如下： public static AudioClip Create(string name, int lengthSamples, int channels, int frequency, bool stream, AudioClip.PCMReaderCallback pcmreadercallback); public delegate void PCMReaderCallback(float[] data); PCMReaderCallback中的PCM指的是音频数据格式，下面会讲到。在stream模式下，lengthSamples表示播放时，每次循环处理的sample数，这会影响到回调参数data的大小。回调函数参数data就是需要填充的音频数据，它的格式是PCM，data的长度可以由lengthSamples * channels计算出来。 值得注意的是，PCMReaderCallback执行环境不一定，它是由Unity回调的，可能是主线程，也可能是子线程。所以当它卡住的时候，可能会卡掉整个应用。 PCM格式 PCM是Pulse Code Modulation的缩写

audio volume normalization [closed]

由 痞子三分冷 提交于 2020-01-09 11:26:06
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 2 years ago . I am writing a music player and I want to normalize the audio volume across different songs. I could think of some different ways to do this, e.g.: Go through all PCM samples (assume floating point from -1 to 1) and select the m = max(abs(sample)). Then apply the factor 1/m to all the PCM samples. This would

多媒体文件格式（五）：PCM / WAV 格式

由 こ雲淡風輕ζ 提交于 2020-01-06 21:30:00
一、名词解析 PCM(Pulse Code Modulation)也被称为脉码编码调制，PCM中的声音数据没有被压缩，它是由模拟信号经过采样、量化、编码转换成的标准的数字音频数据。采样转换方式参考下图进行了解： 音频采样包含以下几大要素： 1. 采样率 采样率表示音频信号每秒的数字快照数。该速率决定了音频文件的频率范围。采样率越高，数字波形的形状越接近原始模拟波形。低采样率会限制可录制的频率范围，这可导致录音表现原始声音的效果不佳。根据奈奎斯特采样定理，为了重现给定频率，采样率必须至少是该频率的两倍。例如，一般CD唱片的采样率为每秒 44,100 个采样，因此可重现最高为 22,050 Hz 的频率，此频率刚好超过人类的听力极限 20,000 Hz。 图中A是低采样率的音频信号，其效果已经将原始声波进行了扭曲，B则是完全重现原始声波的高采样率的音频信号。 数字音频常用的采样率如下： 2. 位深度 位深度决定动态范围。采样声波时，为每个采样指定最接近原始声波振幅的振幅值。较高的位深度可提供更多可能的振幅值，产生更大的动态范围、更低的噪声基准和更高的保真度。 位深度越高，提供的动态范围越大。 二、PCM 在上面的名词解析中我们应该对PCM有了一定的理解和认识，下面我们将对PCM做更多的讲解。 1. PCM音频数据存储方式 如果是单声道的文件，采样数据按时间的先后顺序依次存入

Compress WAV PCM to Microsoft GSM 6.10 Audio Codec using C#

由 和自甴很熟 提交于 2020-01-06 04:32:47
问题 Does any one know how to convert a uncompressed wave file in PCM format to a wav file compressed in GSM 6.10 Audio Codec? 回答1: Try How to convert to Gsm Wav file static void ToWavWithGsm() { string fileName = @"e:\Down\male.wav"; WaveReader wr = new WaveReader(File.OpenRead(fileName)); IntPtr format = wr.ReadFormat(); IntPtr formatGsm = AudioCompressionManager.GetCompatibleFormat(format, AudioCompressionManager.Gsm610FormatTag); byte[] dataGsm = AudioCompressionManager.ToFormat(wr, formatGsm)

lame,把ios录音转换为mp3格式

由 房东的猫 提交于 2020-01-04 21:29:09
在ios设备中进行录音，录音文件的格式为caf。但这种格式在很多设备中没法播放。为了适应终端的播放功能，特将caf转换为mp3格式文件来使用。 在录制caf文件时，需要使用双通道，否则在转换为MP3格式时，声音不对。caf录制端的设置为： NSMutableDictionary * recordSetting = [ NSMutableDictionary dictionary ]; [recordSetting setValue :[ NSNumber numberWithInt : kAudioFormatLinearPCM ] forKey : AVFormatIDKey ]; // [recordSetting setValue :[ NSNumber numberWithFloat : 8000.0 ] forKey : AVSampleRateKey ];//采样率 [recordSetting setValue :[ NSNumber numberWithInt : 2 ] forKey : AVNumberOfChannelsKey ];//声音通道， 这里必须为双通道 [recordSetting setValue :[ NSNumber numberWithInt : AVAudioQualityLow ] forKey :

How can I get frequency data from PCM using FFT

由 纵饮孤独 提交于 2020-01-01 03:11:38
问题 I have an array of audio data I am passing to a reader: recorder.read(audioData,0,bufferSize); The instantiation is as follows: AudioRecord recorder; short[] audioData; int bufferSize; int samplerate = 8000; //get the buffer size to use with this audio record bufferSize = AudioRecord.getMinBufferSize(samplerate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT)*3; //instantiate the AudioRecorder recorder = new AudioRecord(AudioSource.MIC,samplerate, AudioFormat.CHANNEL

JLayer Mono Mp3 to PCM decoding

由 折月煮酒 提交于 2019-12-30 12:23:48
问题 I am currently working on mp3 decoding with javalayer 1.1. So I want to receive the raw PCM data from my 44100 Hz, 16bit, Mp3s. It is perfectly working fine with stereo mp3s, but i have strange issues with mono mp3s. Here some code. InputStream data = c.getResources().openRawResource(resId); Bitstream bitstream = new Bitstream(data); Decoder decoder = new Decoder(); while(thereIsData) { Head frameHeader = bitstream.readFrame(); SampleBuffer buffer = (SampleBuffer) decoder.decodeFrame
订阅 pcm