bogen pcm

1. Codec Introduction In mobile devices, the role of codec can be summed up to 4 kinds, respectively: For PCM and other signals D/a conversion, the digital audio signal to analog signals to the mic, LineIn or other input source analog signals for A/D conversion, the analog sound signal conversion CPU can handle digital signals to control the audio channel, such as playing music, listening to FM radio, or when answering the phone, the flow of the audio

=%sclient _secret=%s "# Get Token's addressres = Requests.get (token_url% (Api_key, Secret_key)) # Send RequestRes_text = res.text # Get the text message in the requesttoken = json.loads (res_text) [' Access_token '] # Extract token information# define the voice to sendVoice_file = ' BAIDU_VOICE_TEST.PCM ' # Voice file to recognizeVOICE_FN = open (Voice_file, ' RB ') # opens file in binary modeOrg_voice_data = Voice_fn.read () # Read File contentsOrg_voice_len = Len (org_voice_data) # Get file l

' IEC958 ', 2scared, no configuration, no wonder I can't find "Master"a little skeptical that maybe the default sound card is incorrect,again amixerjun% amixer-c 1 scontrolssimple mixer control ' Master ', 0Simple mixer control ' headphone ', 0Simple mixer control ' Speaker ', 0Simple mixer control ' PCM ', 0Simple mixer control ' mic ', 0Simple mixer control ' mic Boost ', 0Simple mixer control ' IEC958 ', 0 Simple mixer control ' IEC958 Default

the most important and complex part of the whole system. By decoding, compressed-encoded video data output becomes uncompressed color data such as YUV420P,RGB and so on, and compressed-encoded audio data output becomes uncompressed audio sampling data, such as PCM data. The role of video and audio synchronization is based on the solution of the packaging module to obtain the parameters of the information, synchronized decoding of video and audio data

); } /*probetheCODECDAI*/ if (!codec_dai->probedcodec_dai->driver->probe_order== Order) { if (codec_dai->driver-> Probe) { ret= codec_dai->driver->probe (Codec_dai); } /*markcodec_daiasprobed andaddtocarddailist*/ codec_dai->probed=1; List_add (codec_dai->card_list,card->dai_dev_list); } /*completedaiprobeduringlast probe*/ if (Order!=snd_soc_comp_order_last) return0; ret=soc_post_component_init (card,codec,num,0); if (ret) returnret; NBSP;NBsp;...... /*createthe

the modules.conf. Use any text editor, such as VI, to open this file and add the following at the bottom: “ #ALSA portion Alias char-major-116 snd #注释: Main device number used by ALSA Alias snd-card-0 Snd-card-ymfpci #snd-card-0 is the first sound card device inside the system, if the system has more than two sound cards, #可以使用snd-card-1 Snd-card-2 and other methods mapping # SND-CARD-YMFPCI is the sound card device name. #OSS/free portion----Because ALSA

: "CREATE database character set Us7ascii;". There are the following error prompts: * CREATE DATABASE Character Set Us7ascii ERROR at line 1: Ora-01031:insufficient Privileges In fact, looking at V$nls_parameters, the character set has been changed successfully. But after restarting the database, the database character set changes back to the original. This command can be used for data switching between temporary and different character set servers. 9. How to query the number of

data of the microphone input is decoded by the audio codec to complete A/D conversion, the decoded audio data through the audio controller into the DSP or CPU for the corresponding processing, and then the data through the audio controller sent to the audio encoder, encoded D/a converted by the speaker output.(2) The format of digital audio has a variety of, the most commonly used is the following three kinds:A, digital audio (PCM): is the data forma

The audio card driver loading method, which may be helpful to new users-general Linux technology-Linux technology and application information, is described below. (1) install alsa-driver: Remove the previous dependency before installation. Tar zxvf alsa-driver-1.0.9b_26_evoc.tar.gz Grep-r? N 'southbridge chip './* Cd alsa-driver-1.0.9b_26 Make clean ./Configure -- with-kernel =/usr/src/linux-2.4 -- with-cards = intel8x0 Make Make install ./Snddevices Note: the following message appears when th

used to record PCM data. The audiotrack is used to play PCM data. PCM is the raw audio sample data that can be played with a VLC player. Of course, the channel sample rate and the like to set their own, because the original sample data is no file header, such as:VLC--demux=rawaud--rawaud-channels 2--rawaud-samplerate 44100 AUDIO.PCMGo back to the two classes of

The information of PCMlock is recorded in GRD, which is located in the SGA of each instance, but each instance only contains part of GRD, And the GRD of all instances is aggregated together. The PCM lock information is recorded in GRD, which is located in the SGA of each instance, but each instance only contains part of GRD, And the GRD of all instances is aggregated together. Record PCM lock informati

analogue in/out internal reference voltage (normally avdd/2, if not overdrive33 out4 analogue output auxillary output driver (speaker, line or headphone)34 spkgnd supply speaker ground (feeds output buffers on pins 33, 35, 36 and 335 spkl analogue output left speaker driver (speaker, line or headphone)36 spkr analogue output right speaker driver (speaker, line or headphone)37 out3 analogue output auxillary output driver (speaker, line or headphone)38 spkvdd supply speaker supply (feeds output b

This article describes in detail the AAC audio decoding algorithm that complies with ISO/IEC13818-7 (MPEG2 AAC audio codec), ISO/IEC 14496-3 (MPEG4 audio codec AAC lowcomplexity) for compression. 1. Program System Structure The following is the AAC decoding flowchart: AAC decoding Flowchart After the master module starts running, the master module puts a part of the AAC bit stream into the input buffer, and obtains the starting point of a frame by searching for the synchronous word, noislessdeco

can only start and compile it with the FC self-replaced Kernel 2.6.9-1.667 (which of the following methods can be used to compile it in the new kernel ).Compilation Method:1. Tar drops two packages and Su is the root user.2. Into alsa-driver-1.0.8rc2./ConfigureMakeMake install3. Enter the alsa-oss-1.0.8rc2./Configure -- With-aoss = YesMakeMake installThen you need to set the dmix plug-in of ALSA to implement soft multi-audio streams.Create the/etc/asound. conf file. The file content is as follo

calculated based on the sampling rate of a WAV file. The file information displayed by the mediainfo tool is as follows: Summary Complete name: audio \ WAV \ adele-rolling_in_the_deep.wav File Format: Wave File Size: 38.3 MIB Length: 3 minutes 47 seconds Average Mixed bit rate: 1 411 kbps Audio ID: 0 File Format: PCM Format settings, endianness: Little Encoding settings ID: 1 Encoding settings ID/prompt message: Microsoft Length: 3 minutes 47 second

Audiotrack and audioflinger exchange audio data (1) Reproduced from: http://www.eoeandroid.com/forum.php? MoD = viewthread tid = 98290. In the Audio Subsystem of the android framework, each audio stream corresponds to an audiotrack instance. Each audiotrack is registered to audioflinger at creation, audioflinger is used to mix all audiotracks and then deliver them to audiohardware for playback. Currently, froyo of Android allows you to create up to 32 audio streams at the same time. That is to

are guaranteed, should the recording file be kept as small as possible? Below are some audio playback formats supported by iPhone OS: AACHE-AACAmr (Adaptive multi-rate, which is a speech format)ALAC (Apple lossless)Ilbc (Internet low bitrate codec, another voice format)Ima4 (IMA/ADPCM)Linear PCM (no compression)Micro-law and a-LawMP3 (MPEG-1 audio layer 3rd)Below are some audio recording formats supported by iPhone OS: ALAC (Apple lossl

Detailed description of AAC decoding algorithm principles This document describes in detail ISO/IEC 13818-7 (MPEG2 AAC audio codec), ISO/IEC 14496-3 (MPEG4 audio codec AAC low complexity) the AAC audio decoding algorithm. 1. Program System StructureThe following is the AAC decoding flowchart:AAC decoding FlowchartAfter the master module starts running, the master module puts a part of the AAC bit stream into the input buffer, and obtains the starting point of a frame by searching for th

/*************************************** **************************************** ************** Author: conowen @ Dazhong* E-mail: conowen@hotmail.com* Http://blog.csdn.net/conowen* Note: This article is original and only used for learning and communication. For more information, indicate the author and its source. **************************************** **************************************** ************/ 1. Introduction to Android audiotrack In Android, mediaplayer and audiotrack can be

The Audio Subsystem in the Symbian OS phone mainly contains two types of independent audio data streams. One is telephone sound data, and the other is multimedia data.These two crucial use cases require sound quality and the ability to talk for a long time. A Digital Audio bus dedicated to audio data is used to meet these requirements.The original hardware audio format actually used on the Symbian OS phone is 16-bit pulse encoding modulation (PCM) dat