bogen pcm

In fact, asound. conf has little to do with dapm. The reason why it is included in the dapm series is to consider the consistency of knowledge. As mentioned in I. Understanding asound. conf The complexity and strength of ALSA-lib can be basically realized by ALSA-lib, which is embodied in configuration files such as asound. conf and. asoundrc. ALSA driver development is only one aspect. asound. conf and. asoundrc are essential for you to configure audio devices as you like. Fortunately, this

new interaction features is to process all the relevant audio elements in the player. The video on the DVD adopts the same mode: After decoding the main video, add subtitles or menus, and then output the video as a complete image, which can be a simulation (compound or component) or digital (DVI, HDMI) baseband signal. In HD (HD) disc players, audio is processed in the same way. Audio streams decoded from the optical disc, audio streams from the Internet or downloaded, or audio elements genera

,},.probe = samsung_asoc_platform_probe,.remove = __devexit_p(samsung_asoc_platform_remove),};module_platform_driver(asoc_dma_driver); Snd_soc_register_platform ()This function is used to register an snd_soc_platform. It can be used by the machine driver only after registration. Its Code clearly expresses its implementation process: Apply for memory for the snd_soc_platform instance; Obtain its name from platform_device for matching the machine driver; Initialize the snd_soc_platform field;

necessary rollback segments: Create rollback segment Rb2 tablespace RBS; 6) edit the initsid. ora file of the initialization parameter file and add the following items: Parallel_server = true Instance_number = 1 Thread = 1 Rollback_segments = (R01, r02, r03, r04) 7) Create the data dictionary required by OPS, that is, run catparr. SQL. 8) Start the instance on all nodes. 6. How can I determine whether a database is running in parallel? Show parameter parallel_server 7. How do I track activity i

: word; {indicates the number of channels for waveform data; 1 for single channel and 2 for STEREO}Nsamplespersec: DWORD; {specified sample rate (number of samples per second)} is generally 8000Navgbytespersec: DWORD; {average rate of data transmission (bytes per second)} bytes per second:Nblockalign: word; {specify block alignment (in bytes). Block align is the smallest unit of data}Wbitspersample: word; {sample size (bytes)} The number of BITs for each sample, usually 16Cbsize: word; {It shoul

bits represent the left and right channels. The data block of the wave file contains samples in Pulse-Coded Modulation (PCM) format. Wave Parts are organized by samples. In a single channel wave file, channel 0 indicates the left channel, and channel 1 indicates the right channel. Audio channel. In a multi-channel wave file, samples are generated alternately. Wave file format description Offset address byte data type capacity 　 File Header 00 H 4 cha

BSCP 协议将 CSR 芯片设计为封装 HCI数据分组，您必须使用 hciattach (hciattach ttySx bcsp) 通知 BlueZ 链接驱动程序，在这里 x 是连接到CSR 芯片组的 UART 通道号。现在 hci_uart 与 CSR 芯片交互并且传送 Bluetooth 数据往返于 BlueZ 栈。 Sony HBH-30 Bluetooth 耳机    前面的Bluetooth 设备示例展示了网络数据流。现在，通过查看 Sony Ericsson Bluetooth 耳机来考虑 Bluetooth音频 (SCO) 数据的传输。在耳机可以开始与 Linux 设备通信以前，它必须被 Linux 设备上的 Bluetooth链路层检测出来。因此，您必须将耳机置于发现模式（通过按下耳机上的一个按钮）。另外，您需要通过 Linux 设备上的 BlueZ 配置耳机的PIN。Linux Bluetooth 设备上使用 BlueZ SCO API 的应用程序现在可以发送音频数据到耳机上。音频数据应当是耳机可以理解的格式（例如，Sony 耳机的 A-law

to facilitate the expansion of functions of the instrument. Main technical indicators Input Port: high-impedance, 75Ω, and 120 Ω external configuration options)Input Interface speed: 2048 kbit/s ± 100ppmDisplay: 12.1 inch TFT LCDProcessor: Pentium/586 or abovePeripheral interface settings: 3.5 inch floppy disk drive, printer parallel interface, serial interface, external VGA display interface, LAN interfaceBody product: 40. 5cm long) X 20cm wide) X 27. 5cm high)Weight: 9 kgWork temperature: 0 ℃

transmits Bluetooth data to and from the BlueZ stack. Sony HBH-30 Bluetooth headsets The preceding Bluetooth device example shows network data streams. Now, you can view Sony Ericsson Bluetooth headphones to consider the transmission of Bluetooth audio (SCO) data. Before the headset can communicate with a Linux device, it must be detected by the Bluetooth link layer on the Linux device. Therefore, you must place the headset in discovery mode (by pressing a button on the headset ). In addition,

discussing the system architecture, let's take a look at the current situation of audio encoding. Currently, there are many Audio Encoding formats, including PCM, ADPCM, DM, PWM, WMA, Ogg, Amr, ACC, mp3pro, and MP3; for human speech has LPC, CELP and ACELP; other MPEG-2, MPEG-4, H. 264, VC-1 audio and video program encoding format. For more information about the market trend of mobile multimedia formats, see Figure 1 ). Figure 1 mobile multimedi

discussing the system architecture, let's take a look at the current situation of audio encoding. Currently, there are many Audio Encoding formats, including PCM, ADPCM, DM, PWM, WMA, Ogg, Amr, ACC, mp3pro, and MP3; for human speech has LPC, CELP and ACELP; other MPEG-2, MPEG-4, H. 264, VC-1 audio and video program encoding format. For more information about the market trend of mobile multimedia formats, see Figure 1 ). Figure 1 mobile multimedia

-ossAlias Sound-service-0-3 Snd-pcm-ossAlias Sound-service-0-8 Snd-seq-ossAlias Sound-service-0-12 Snd-pcm-oss Add the above paragraphs to/etc/modules.conf.After execution: Modprobe snd-the sound card module name. Note: Sometimes after the first execution of the ALSA module load operation, there will be some error messages, you can restart the system at this time, and then load a module to see if the proble

FFmpeg's understanding about Timebase One: FFmpeg's layered structure When writing this article, let's outline the FFmpeg layering problem, which we summarize in accordance with the most common "transcoding program" and "Camera capture" and "player" three models. 1: transcoding transcoding basic Flow: FLV Format Data-->H264/AAC data-->YUV/PCM data-->H264/AAC data-->mp4 format data. A: We call the FLV format data or the MP4 format data This layer is c

broadcasts the request to the LCK process for all remote instances, and after the remote instance receives the request, Update your information and feedback (if a process for the remote instance holds the lock in an incompatible manner, the LCK process for the instance notifies the bearer process to release the corresponding lock and then returns), and each instance contains only one LCK process by default.GRD: One way to save and organize all the resources related to memory fusion in the share

can be executed, or, without the frequently changing documents, only the code, Golang this is really perfect! The example package is not the same as the API package. This and utest is not the same, utest because to access the contents of the package, so the same package with the API, more convenient. Example because it is for the user's example, of course, and the API package can not be the same, not the same. Golang's example, forcing the API designer to be able to design the API very easily f

, 11.025 khz, 22.05 khz and 44.1 khz. For non-PCM formats, this must be calculated according to the specifications of the manufacturer's format tags.DWORD navgbytespersec; The average data transfer rate required, marked in bytes per second. If wFormatTag is wave_format_pcm,navgbytespersec, it should be equal to the product of Nsamplespersec and nblockalign. For non-PCM formats, this must be calculated accor

WaveFormatExWaveFormatEx{WORD wFormatTag;WORD nchannels;DWORD nsamplespersec;DWORD navgbytespersec;WORD nblockalign;WORD wBitsPerSample;WORD cbsize;};This structure defines the format of the wave audio data. Only the general format information contained in all audio is included in this structure, and for formats that require additional information, the structure is included and as the first parameter.Member variableswFormatTagWave sound format. The format label is the one that Microsoft has regi

As one of the acoustic file formats used in multimedia, the wave file header adopts the riff format as the standard. Riff is short for resource interchange file format. The first four bytes of each wave file are "riff ". Reasonable Use of the wave file header can be more effective in speech decoding. Generally speaking, voice encoding refers to compressing 8 kHz sampled, 16-bit quantified linear PCM voice signals into other formats of voice signals, d

Today I have made a good research on the speech compression encoding in Android, and I have some tips: First of all on the collection of sound source format is the PCM-16bit format, What Is PCM format, you can see the following article: Http://blog.csdn.net/tkboy/archive/2009/12/19/5038947.aspx PCM is a non-compressed sound source.

I have been studying the implementation of voice calls recently. Now I have recorded my Implementation ideas here. However, this is a simple idea and a learning conclusion after Google and Baidu because of the first access to voice calls. I think a voice call system has at least four modules. These are PCM (Pulse Code Modulation) Speech collection, encoding/decoding, network transmission, and audio playback. If UI interaction is counted, five modules