bogen pcm

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Ubuntu can identify sound cards but does not provide a sound solution

Configure alsa (If your sound card exclusive, perform this step) sudoapt-getinstallalsa-osssudogedit/etc/asound. conf Add the following content: pcm. card0 {typehwcard0} pcm .! Default {typeplugslave. pcm "dmixer"} pcm. dmixer {typedmixipc_key1025 configure alsa (if you have an exclusive sound card, perform this step)

Spread voice over the Internet

to ActiveX control. ActiveX controls allow programmers to call functions that others can achieve without starting from scratch. Codecs provides similar functions, but it focuses on how to convert media formats. For example, if you want to write a CD-to-MP3 application, you only need to do the following: L read data from the CD audio track. L generate a valid MP3 file header. L call the corresponding CODEC to encode the audio track data into MP3. Windows has a lot of codec. The following are som

IPhone audio 101 (Part 1): file and Data Types

first of three audio development tutorials. In this article, we will start with the file and data format. File Format and Data Format First, we need to know that each audio file has two parts: 1 is the file format (or audio container), 2 is the data format (or audio encoding ). The file format (or audio container) describes its own format. The actual audio data can be encoded in many different ways. For example, a caf file is a file format that can contain audio data encoded in MP3, lin

Application of Digital microphone and array sound pickup Technology

into analog electrical signals that are amplified by internal buffering. Driven by the clock signal (SCL), the modulus is finally converted into one-bit PDM audio data, which is output from the data pin.    Figure 2a ECM analog microphone Circuit    Figure 2B MEMS analog microphone Circuit    Figure 2C ECM/memes digital microphone Circuit Comparison of various types of microphones: Table 1 lists the performance indicators and their respective advantages and disadvantages of ECM anal

ALSA struct process 2

The control interface allows the user space application (ALSA-Lib) to access and control multiple switches and sliding controls in the Audio Codec Chip. We need to actively call the snd_pcm_new () function to create the PCM device during driver initialization, while the control device is created in snd_card_create, Snd_card_create () creates a control device node by calling the snd_ctl_create () function. Therefore, you do not need to create a contro

Libmad Simple Description

MAD (Libmad) is an open source High-precision MPEG Audio Decoder library that supports MPEG-1 standards. Libmad provides 24-bit PCM output, fully fixed-point computing, and is ideal for use on embedded hardware platforms without floating-point support. The decoding of MP3 files can be achieved using a series of APIs provided by Libmad. The "mad.h" header file defines the LIBMAD data structure and API functions [15]. Table 4 The main data structures in

How to Design Large Oracle databases

multiple INSTANCE instances on different nodes to access a database at the same time to improve system availability, scalability, and performance. Each INSTANCE in the Oracle Parallel Server can read the table or index data blocks in the shared database into the local buffer, this means that a data block can exist in the SGA zone of multiple INSTANCE instances. Therefore, it is important to maintain data consistency in These buffers. Oracle uses the Parallel Cache Management lock to maintain th

How to Design Large Oracle databases

INSTANCE instances. Therefore, it is important to maintain data consistency in These buffers. Oracle uses the Parallel Cache Management lock to maintain the consistency of the buffer. Oracle also implements the PCM lock through the I DLM (integrated distributed lock manager, the dedicated LCK process is used to achieve data consistency between INSTANCE instances. Consider this situation: When INSTANCE1 modifies block x, INSTANCE2 also needs to modify

How to Design Large Oracle databases

Oracle Parallel Server Applications The Oracle Parallel Server allows multiple INSTANCE instances on different nodes to access a database at the same time to improve system availability, scalability, and performance. Each INSTANCE in the Oracle Parallel Server can read the table or index data blocks in the shared database into the local buffer, this means that a data block can exist in the SGA zone of multiple INSTANCE instances. Therefore, it is important to maintain data consistency in These

Myminimad. c

, struct mad_stream * stream){Struct buffer * buffer = data;If (! Buffer-> length)Return MAD_FLOW_STOP;Mad_stream_buffer (stream, buffer-> start, buffer-> length );Buffer-> length = 0;Return MAD_FLOW_CONTINUE;}/* This section is the pcm audio after processing the sample */Static inline signed int scale (mad_fixed_t sample){Sample + = (1L If (sample> = MAD_F_ONE)Sample = MAD_F_ONE-1;Else if (sample Sample =-MAD_F_ONE;Return sample> (MAD_F_FRACBITS + 1-

Audio recording format for iOS and Android devices with instant voice connectivity

Before I developed my iphone, I knew very little about the sound format. I know some of the differences between "wav" and "MP3" sound formats, but I certainly can't tell you exactly what the "AAC", "CAF" sound file is, and how the best way to convert audio files on your Mac is not known.I know that if you want to be a qualified iphone developer, you have to have a basic understanding of sound file data format, conversion, recording, and what audio manipulation API to use.This article is the firs

Description of waveformatex format

kHz, 11.025 kHz, 22.05 kHz, and 44.1 kHz. For example, to specify a sample frequency of 11.025 kHz, Set NsamplespersecTo 11025. For non-PCM formats, this member shocould be computed according to the manufacturer's Specification of the format tag. Set the sample frequency for playing and recording each audio channel. IfWformattag= Wave_format_pcm, thenNsamplespersecUsually 8.0 kHz, 11.025 kHz, 22.05 kHz and 44.

Ubuntu sound card exclusive solution

Http://wiki.ubuntu.org.cn/problem solving solution: software_mixing A much better way of getting ESD to work with non-ESD-aware apps is to make ALSA use its software mixer.Enable ALSA to use its own software mixer to work together with applications that are not recognized by ESD. $ Sudo/etc/init. d/ALSA-utils stop $ sudo gedit/var/lib/ALSA/asound. State(Add the following text verbatim to the start of this file) $ Sudo/etc/init. d/ALSA-utils start# This text shoshould be added to the beginning #/

Linux Driver: Audio driver (i) alsa__linux

116/* standardconfiguration * * 10, Sound Directory Description: In the "depth of practice embedded ..." "640 pages Second, "Driving framework" 1. Create sound card: Snd_card_new (Linux 2.6.22 snd_card_create replaced)---> 2, the creation of child devices: Snd_device_new (Snd_pcm_new, Snd_rawmidi_new, Snd_ctl_new, Snd_timer_new, snd_card_proc_new, snd_jack_new)---> 3, registered sound card: snd_card_register Third, "PCM function"

"Linux Kernel Programming" chapter II: ALSA Drive __arduino

This paper is based on mstar801 platform Linux2.6.35.11 kernel. First, ALSA drive to create a sound card process 1. Create sound card Snd_card_create (Linux2.6.30 and later APIs), Snd_card_new (Linux2.6.30 previous API). Description: First step to create a sound card2. Create a PCM device and add a sound card int snd_pcm_new (struct snd_card *card,const char *id,int device,int playback_count,int capture_count,struct SND_PCM * RPCM); Description: Param

You can read it. Embedded linux/android alsa_aplay alsa_amixer command line usage

A few days ago in the embedded Linux used to Alsa command, on-line check the data do not give force, only a little bit to check, and finally can use, will this use method to tell everyone, lest everyone less detours.0. First check which ALSA cmd the system supports:ll/system/bin/asla*After checking, my android4.4.2+linux3.10 system supports the following ALSA cmd-rwxr-xr-x root Shell 68242 2014-10-08 05:18 alsa_amixer-rwxr-xr-x root Shell 67639 2014-10-08 05:18 Alsa_aplay-rwxr-xr-x root Shell 13

iOS audio Development (recording + play + clip + compositing + compression transcoding)

], Avsampleratekey,//Sample rate[NSNumberNumberwithint:2], Avnumberofchannelskey,//Number of channels[NSNumberNumberwithint: -], Avlinearpcmbitdepthkey,//Sample bits (PCM exclusive)[NSNumberNumberwithbool:NO], avlinearpcmisnoninterleaved,//Whether to allow audio crossover (PCM exclusive)[NSNumberNumberwithbool:NO],avlinearpcmisfloatkey,//Whether the sampled signal is a floating point number (

Linux audio alsa Mechanism learning notes < a >

First macro view of the kernel exposed to the upper interface:[Email protected]:/# cat/dev/snd/ControlC0 pcmc0d10p pcmc0d13c pcmc0d15c pcmc0d2c pcmc0d3c pcmc0d5p pcmc0d8cpcmc0d0c pcmc0d11p pcmc0d13p pcmc0d15p pcmc0d2p pcmc0d3p pcmc0d6c pcmc0d9cpcmc0d0p pcmc0d12c pcmc0d14c pcmc0d1c pcmc0d31c pcmc0d4p pcmc0d6p pcmc0d9ppcmc0d10c pcmc0d12p pcmc0d14p pcmc0d1p pcmc0d32p pcmc0d5c pcmc0d7p TimerMainly by control and many PCM devices, wherein the control inter

You can read it. Embedded linux/android alsa_aplay alsa_amixer command line use method

A few days ago in the embedded Linux used to Alsa command, on-line check the data do not give the force, only a little hands to check, and finally can use, will this use to tell everyone, lest everyone less detours.0. First check which ALSA cmd the system supports:ll/system/bin/asla*After checking, my android4.4.2+linux3.10 system supports such as the following ALSA cmd-rwxr-xr-x root Shell 68242 2014-10-08 05:18 alsa_amixer-rwxr-xr-x root Shell 67639 2014-10-08 05:18 Alsa_aplay-rwxr-xr-x root S

Commonly used ITU speech codec Daquan (g.711, g.722.1, g.722.2, g.723.1, g.726, g.728, g.729, g.729.1)

I have summarized the commonly used ITU speech codec for your reference: 1. g.711: pulse code modulation (PCM) of voice Frequencies Encoding type: Expanded PCMEncoding rate: 64 Kbps Theoretical latency: the time for processing a sample (1/8000 seconds = 0.125 milliseconds ); Sound Quality: Long Distance quality;Advantages: Low algorithm complexity, low compression ratio (CD sound quality> 400 Kbps), minimum codec latency (relative to other technologi

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