Before talking about this topic, you need to know the following points:
1. mediarecorder/audiorecord differencesMediarecorder is a common recording class provided by the system, which can be used to generate Audio Recording files. Audiorecord is also a recording class provided by the system, but it can directly capture audio streams and developers can process their contents in real time. Common scenarios such as voice chat, tomcat, and karaoke.2. Riff/WAV/PC
write the sound to the file.Two. Implement automatic conversion of audio format and call speech recognitionThe problem of recording solved, quickly and Baidu voice recognition to join together to use:No matter how clear your recordings are, you find that Baidu will always return to you:{'err_msg'Speech quality error. ' ' Err_no ' ' SN ' ' 6397933501529645284 '} # The sound is not clearActually not not hear clearly, but Baidu supports the audio format PCM
The recent porting of the WinCE.NET platform to the Pocket PC has been a little cumbersome. The original project has a DirectShow package class, used to play the MP3 format of audio, and Pocket PC 2003 does not support DirectShow, in order to be compatible with the old system it seems to change this piece of code. First, the Wmplayer control is excluded, which is too expensive (DirectShow is the same). Libmad is a cross-platform MPEG Audio decoding library, which uses integer arithmetic to simul
this class is the original PCM binary audio data. If there is no file header or end, the generated PCM file cannot be played directly using Mediaplayer and can only be played using AudioTrack. You can use AudioRecord to Process audio recorded and played in real time.
After learning about the features of these two classes, I initially decided to use the MediaRecorder class to solve the recording pause prob
Coding2.1 Digital Sound Signal(1). Frequency of Sound: (2). Sampling--Quantization--code:Sampling frequency: The Nyquist theory states that the sampling frequency should not be less than twice times the maximum frequency of the sound signal, so that the voice can be restored to the original sound as a digital expression, which is called lossless digitization (lossless digitization) Sampling accuracy: Number of bits per sound sample (bps)Amount of data (Bytes/sec) = (sampling frequency (Hz
, USB audio class device, USB video class device, and mass storage devices can be used to develop general device drivers. You only need to modify the vid and PID.(2) In addition to implementing the stream Driver Interface for a USB device, you also need to implement the three functions: USB installdriver, USB deviceattach, and USB uninstalldriver, And the callback function for pulling out the device.
3. Audio Processing(1) PCM coding method: pulse cod
Tutorial
Foreword: Many friends all like the vocal ring, after all, the simple chord bell is still monotonous some. But it is limited to the complexity of the production, and there is no good tutorial to make their own ringtones. Now I put the method of making a simple, I believe that the software to start fast friends after watching it can make their own ringtones.
(although now have pcsync software, can very simple produce the MMF human voice, but because the upper limit is only 200K, and is
channel samples, data is stored in a continuous channel. But in most cases, we just need to use staggered mode. Period (cycle): The time between interrupts in the hardware. It represents the input delay.
The sound card interface has a pointer to indicate the current read/write location in the hardware cache of the sound card. As long as the interface is running, the pointer loops to a location in the cache.Frame size = sizeof (one sample) * NchannelsThe cache (buffer) and period (size) sizes co
different ways. For example, a file suffixed with CAF is a file format that can contain audio data encoded in MP3, linear PCM (lpcm), and many other formats.
Next let's take a deeper look.
Data format (or audio encoding)
Next we will begin to understand audio encoding, not file format, because audio encoding is the most important thing.
The following are the audio data encoding formats supported by the iPhone and the descriptions for each of them:
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above? Or does DLM reconcile the conflict of resources at that level? Is that a a line of records? or a block of data? Or is it a data file? Haha--The answer is: Data block!!!That is, when a process wants to modify a, the request to the DLM is "block A's operation rights".Oracle Cluster development history is divided into two stages, originally Oracle Parallel server (Oracle Parallel server OPS), then to 9i to RAC, two stages of the DLM name is also different, OPS called
strategy
Oracle parallel servers allow multiple instance instances on different nodes to access a database simultaneously to improve system availability, scalability, and performance. Each instance instance in an Oracle parallel server can read the data blocks of a table or index in a shared database into a local buffer, which means that a block of data can exist in the SGA area of multiple instance instances. It is important to maintain the consistency of the data in these buffers. Oracle uses
databasePartition and distribution of system hard diskConsiderations for backup and recovery strategies Second, Oracle Parallel server application partitioning strategyOracle parallel servers allow multiple instance instances on different nodes to access a database simultaneously to improve system availability, scalability, and performance. Each instance instance in an Oracle parallel server can read the data blocks of a table or index in a shared database into a local buffer, which means th
AAC V.s. MP3Http://en.wikipedia.org/wiki/Advanced_Audio_Coding#AAC.27s_improvements_over_MP3AAC Patent Lisense FAQ:Http://www.vialicensing.com/licensing/aac-faq.aspxNeed a license but usually no fees, exept that's when end user products is (/contain?) encoders/decoders.Overview:$ (ndk_root)/docs/opensles/index.html
Decode Audio to PCM
Note:this feature is available at API level and higher.
A standard audio Player plays a audio device, and the
1 PCM InterfaceFor different digital audio subsystems, there are several microprocessor or interfaces used for digital conversion between DSP and audio devices. The simplest audio interface is the PCM (pulse Coding Modulation) interface, which consists of Clock Pulse (bclk), frame synchronous signal (FS), and receive data (DR) and data transmission (dx. On the rising edge of the FS signal, the data is trans
1 PCM InterfaceFor different digital audio subsystem, there are several interfaces between microprocessor or DSP and audio devices for digital conversion. The simplest audio interface is the PCM (Pulse coded modulation) interface, which consists of a clock pulse (BCLK), a Frame sync signal (FS), and a receive data (DR) and send data (DX). At the rising edge of the FS signal, the data transfer begins with th
partitioning strategyOracle parallel servers allow multiple instance instances on different nodes to access a database simultaneously to improve system availability, scalability, and performance. Each instance instance in an Oracle parallel server can read the data blocks of a table or index in a shared database into a local buffer, which means that a block of data can exist in the SGA area of multiple instance instances. It is important to maintain the consistency of the data in these buffers.
port number Socket.bind (address) # receiving packets from the socket Socket.recvfrom (bufsize[, flags]) # Close Socketsocket.close ()The use of the socket module is relatively simple, there are many examples on the internet, which is no longer described in detail.Three, Pyalsaaudio modulePyalsaaudio () is a module for accessing the ALSA API in Python, which makes it easy for users to access ORANGPI zero PCM and mixer devices in the program, with ins
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