audio watermarking

I. Digital Audio The audio signal is a continuous analog signal, but the computer can only process and record binary digital signals. The audio signals obtained from natural audio sources must undergo certain transformations, the digital audio signal can be sent to the compu

HTML5 media element detection support for codecs Although media elements can implement audio and video functions, not all browsers support video tags and audio tags. This means that developers must provide many media sources. The JavaScript API can detect whether the browser supports a certain format and codec. Both media elements have a canPlayType () method that receives a format/decoder string and retur

Recent projects need to deal with audio, so the online collection of some basic audio-related knowledge, organized as followsThe sound in nature is very complex, the waveform is extremely complex, usually we use pulse Code code modulation code, namely PCM coding. PCM encoding converts a continuously varying analog signal to a digital signal by sampling, quantization, and encoding three stepsSampling (sample

White Wolf Source: Http://www.manks.top/article/h5_audio This article copyright belongs to the author, welcome reprint, but without the consent of the author must retain this paragraph statement, and in the article page obvious location to the original link, otherwise reserves the right to pursue legal responsibility.How do I play audio on a webpage? The topic is very tall, actually the development work really is very easy, only need to use HTML5 's o

IOS offers a rich set of tools for working with sound in your application.These tools are arranged into frameworks according to the features they provide, as follows: ■ Use themedia player framework to play songs, audio books, or audio podcasts froma user's iPod library.For details, seemedia player framework reference, iPod libraryaccess programmingguide, And the addmusic sample code project. ■ Use the

Two audio recording methods for Android In mobile APP development, every time an APP is designed for multimedia development, many programmers will suffer, and the project development progress will slow down And the APP testing will increase. Multimedia Development in Android includes audio playback, audio recording, video playback, and video recording. Although t

From: http://book.51cto.com/art/201001/179672.htm Chapter 2Android audio system This chapter introduces the content of the android audio system, which is mainly the audio input/output link and does not involve codec content (the audio system has content from the driver, local framework to Java framework ). This chapt

Audio Processing in WINDOWS　　 In WINDOWS, audio processing can be roughly divided into two parts: audio input and output, and ACM compression.In WINDOWS, Apis such as sndPlaySound (MCI) can be called to play a WAV file, but this is obviously not what we need to do. You must be able to directly Process audio data stream

In the production of the publishing side, you need to collect the sound card output signal, in order to mix with the microphone input signal, for audio processing technology, mainly the following: Capture Microphone Input Collect sound card output Send audio data to the sound card for playback Mixing multi-channel audio inputs Here are t

016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchr

Enable recording and playing on Android 4.4Background since the beginning of ICS, the audio system has undergone great changes. First, alsalib has been abandoned, and then AIO has been adopted. All levels of frameworks have their own characteristics, it is a step away from the Linux Audio Application, forming its own unique audio management and

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374668.htmlThe previous blog introduces the local video collection, this article introduces the audio capture process, but also introduces WEBRTC native audio collection, and then introduces chromium source to its customization.1. WebRTC native Audio captureLet's first introduce the interface concept

Android 4.1, abbreviated as JB. In Chinese eyes, the word JB is also related to animals. Google modified android so frequently, and finally launched a version that can be put on the mouth of JB all day long. In the future, my articles can also use JB to express the version number, while JB to express what Chairman Mao often said, "Strategic contempt". Please try to understand my mood for writing the word JB according to the context. Today, we will give a brief introduction to the earth-shaking c

Today, as mobile QQ or MSN enters the mobile phone, mobile users begin to enter the new era of video chat or video conferencing, but due to mobile environments (such as elevators, roads, squares, restaurants, bars, trains, and automobiles) mobile phone users are increasingly eager for HD Audio due to various noisy noises or interference. This is why HD Audio has already ranked first in the mobile phone mark

Transferred from: http://www.cnblogs.com/fangkm/p/4374668.htmlThe previous blog introduces the local video collection, this article introduces the audio capture process, but also introduces WEBRTC native audio collection, and then introduces chromium source to its customization.1. WEBRTC native Audio captureLet's first introduce the interface concepts in WEBRTC t

Web Audio is also a relatively new JavaScript API, which is different from the General Web applications should not fall into these APIs, but some types of web sites such as game engines or online music editors should be used.The Web audio API is tightly wrapped around a conceptual design: audio context, which is like a graph, each node in the way is an

In the FFmpeg ultra-detailed comprehensive tutorial---------------a sample of reading PC camera video data and sending it as a live stream with rtmp protocol is done in the camera live text, but the audio support is not implemented, so the example is further perfected in this article and the video synchronization problem of live stream is analyzed in detail. , a code example is also given. For the live stream, here only the synchronization of the send

To achieve CD quality (16bit/44.1khz stereo) need 1411.2 kbit/s bandwidth, obviously Bluetooth A2DP cannot allow (Bluetooth 2.0/2.1 + EDR The largest carrier rate is about 2.1mbit/s, minus the protocol header, interference, power dissipation factor influence , the actual maximum transfer rate is approximately 1000bit/s), so the audio data needs to be compressed. We know that the same bit rate, different encoding format compression rate is different, t

1. Audio package structure: The key member variables are as follows, and their fields are self-explanatory and need not be further described: Public var lastseq: uint; Public var from: uint; Public var Sid: uint; Public var seq: uint; Public var time_stamp: uint; Public var fec_id: uint; Public var type: uint; Public var voice_type: uint; Public var payload: bytearray; Public var wavbytes: bytearray; Public var isbroadcast: Boolean; Public var playtim

from the theoretical height of analysis, if you are experiencing problems, through the search engine came here, please directly read the article "Summary", this part of the reference post is basically Ubuntu audio problem Daquan.Now let's take a bottom-up analysis of this problem from the bottom up.First of all, the Linux community is facing the biggest and most realistic status quo, is to get too little support. From the chip level, each OEM has the