bogen pcm

Introduction Audio and video are the main methods for multimedia applications to provide information to users. These audio and video data generally have a high sampling rate, and the compressed raw data has practical value, otherwise, it will not only occupy a large amount of storage space, but also have a low validity rate during playback or network transmission. Therefore, audio and video digital compression encoding is widely used in multimedia applications. This article mainly describes Audi

First, Audio architecture overview(1)ALSA is the acronym for Advanced Linux Sound Architecture, which has become the mainstream audio architecture for Linux and wants to learn moreFor information and knowledge about this open source project at ALSA, please see the following URL: http://www.alsa-project.org/.In the kernel device driver layer, ALSA provides alsa-driver, while at the application layer, ALSA provides us with alsa-lib, as long as the application callsAlsa-lib provides an API that ena

This article comes from: Http://blog.csdn.net/jtlyr/article/details/5321884 Here are some other websites that introduce wav files, which are recorded below: Https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Http://blog.csdn.net/zhihu008/article/details/7854529 There are many methods for compressing PCM audio files into ADPCM files (such as ms acm and SOx). This article mainly introduces the public algorithm (as shown below, if you need to

interacts with memory through DMA. To be continued 1. The ac97 interface or I2S or PCM Interface can connect the CPU and CODEC (wm9714/alc5620/alc5621), and configure the format: the PCM Interface must be configured with the sampling rate, number of sampling digits, number of channels, and transmission format. The I2S interface must be configured with the sampling rate, number of sampling digits, number of

Summarize the Web site audio live program and the problems encountered.Code: (GitHub, to be sorted)Results: With opus Audio encoding, the Web Audio API plays, which can reach up to 100ms latency, high-quality, low-flow audio live.Background: VDI (Virtual Desktop) h264 Web-site pre-research, after the H264 video live solution after the resolution of a delay has a high demand for audio live program (interactive, audio and video synchronization).Premise: The FLEXVDI open source project supports aud

IntroductionThe PDM represents the pulse density modulation. However, a better abbreviation is "1-bit oversampling audio" because it is simply a high-sample-rate, single bitrate digital system. If you are looking for an advantage, it is that the sample rate is several times the audio CD, and a proper way to reduce the word length from 16bit to 1bit, which will be the basis of a PDM system.Most modern digital audio systems use multi-bit PCM (pulse-code

The simplest Audio Player Based on FFMPEG + SDL: Split-decoder and player, ffmpegsdl Two examples of the simplest FFMPEG + SDL-based audio player are described in this document: FFmpeg audio decoder and SDL audio sampling data player. These two parts are two examples split from the audio player. The FFmpeg audio decoder decodes the video data from PCM sampling data, while the SDL audio sampling data player implements the playback of

Although ac97 is currently debugging audio, the idea is the same. Transfer to another person's article record Original article address: Http://chxxxyg.blog.163.com/blog/static/150281193201033105123937/ # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include # Include "s3c24xx-pcm.h" # Define s3c24xx_pcm_debug 0# If s3c24xx_pcm_debug# Define dbg (X...) printk (kern_debug "s3c24xx-

the maximum frequency of the sound signal, in order to restore the voice of the digital signal as the original sound, the audio file sampling rate is generally 40~50khz, such as the most common CD quality sampling rate of 44.1KHZ. The process of sampling and quantifying sound is calledPulse coded modulation(Pulse Code modulation), referred to as PCM. PCM data is the most original audio data completely loss

for the audio file format is 20KHZ. According to the Nyquist theory, only if the sampling frequency is higher than twice times the maximum frequency of the sound signal, the voice of the digital signal can be restored to the original sound, so the audio file sampling rate is generally 40~50khz, such as the most common CD quality sampling rate of 44.1KHZ.The process of sampling and quantifying sound is called Pulse Code modulation (modulation), or short PCM