best audio software

(mediaPlayer. isPlaying () mediaPlayer. stop (); mediaPlayer. release (); mediaPlayer = null; break; case AudioManager. AUDIOFOCUS_LOSS_TRANSIENT: // indicates that the audio focus is lost. Pause the playback and wait for the audio focus to be obtained again. if (mediaPlayer. isPlaying () mediaPlayer. pause (); break; case AudioManager. AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK: // if (mediaPlayer. isPlaying () {

. dwTagFlag = MITA_TAGREAD_NONE;Param. hDecoderChain = MITA_NULL;Param. hInstance = gInstance;Param. hItemArray = MITA_NULL;Param. hStream = MITA_NULL;Param. nStreamBeginPos = 0;Param. lpPath = L "../media/stereo.ogg ";MITA_Ins_ParseSong ( Param );MITA_CheckError ();If (Param. hItemArray = MITA_NULL | MITA_Array_GetCount (Param. hItemArray) = 0){Printf ("ERROR: No valid audio data. \ n ");If (Param. hItemArray)MITA_CloseHandle (Param. hItemArray );MIT

Based on the analysis in the first three articles, you only need to write a MAKEFILE file to compile main. cpp and aaccodechelper. cpp, as shown below: Makefile: Cxx = g ++ Objects = Main. O aaccodechelper. oLibs =-L/usr/local/lib-lfaac-wl,-rpath,/usr/local/lib-I/usrWavtoaac: $ (objects)$ (Cxx) $ (objects)-O darling $ (libs)Aaccodechelper. O: aaccodechelper. cpp aaccodechelper. h typedef. h$ (Cxx)-C aaccodechelper. cpp $ (libs)Main. O: Main. cpp aaccodechelper. h$ (Cxx)-C main. cpp $ (libs)Cl

codec coding and decoding should hurry up! Another extremely complex change is that audio defines several output flags (see audio_output_flags_t enumeration definitions of audio. h ). According to the annotation, this value has two functions. One is that at users can specify what outputdevice they want to use. The other is that the device manufacturer can declare the output device that it supports (it see

driver writers, they can let the driver adapt to all the changes that can be found, but it's like a guessing game without an end. So, if you have a special sound card chip and trapped in its Linux driver, please do not blame the writing driver, they have spring silkworm to die silk side. Today, the most famous and active Ubuntu/linux-driven writing team is counting ALSA and OSS, and what we can do about this extremely difficult task that they accept should be a tribute to them.Speaking of the d

understand the decoding process of ffmpeg. * * This software decode audio streams (Aac,mp3 ...) to PCM data. * Suitable for beginner of FFmpeg. * */#include After running the resulting program, the following audio files are decoded.The decoded PCM sample data is saved as a file. You can view the contents of the PCM after you set up the sample rate with Adobe Aud

tutorial on FFmpeg audio decoding. * By studying this example, you can understand the FFmpeg decoding process. ** This software decode audio streams (AAC, MP3 ...) to PCM data. * Suitable for beginner of FFmpeg. **/# include After the running result program is run, the following audio files are decoded. The decoded P

In a big aspect, the android audio system architecture has not changed much compared with Android. The analysis of the audio architecture of 2.2 still applies to 2.3. Many people have already elaborated on this aspect in detail and I will not repeat it here. The storage location of files in each module has changed, so you need to understand this. 1. A major improvement is the introduction of the mixable

iphone.· Μ-law and A-law: As far as I know, this encoding is an alternating coded analog data for digital format data, but is better than linear PCM in speech optimization.· MP3: This format is what we all know and like, although it's been a long time, but MP3 is still a very popular encoding format, and it can be well supported by the iphone.What encoding format do we choose?The above looks like a big table, but in fact there are only a few of us that are preferred for development. When making

Transferred from: http://www.cnblogs.com/javawebsoa/archive/2013/05/20/3089511.htmlThe students who have been in touch with iOS audio development know that core audio is the basis for the digital audio processing of iOS and MAC, which provides a set of software frameworks that the application uses to process

"Data Recovery failure description"Company financial personnel for data maintenance, misoperation, in the financial software to initialize the data, because recently did not do backup, it caused a lot of financial documents lost.Because the financial data is very important, the customer is anxious to get back.The data Recovery Analysis "Engineer detection, in SQL Server database Enterprise Manager, discover what the

codec can be completed by pure software, you can also use dedicated hardware chips. Playback Process Extract related files from the storage device and perform Decoding Based on the encoding method used in the recording process. The audio system selects the final matching audio playback device for the playback instance. The decoded data is tran

other factors, the resulting audio files may have a certain degree of distortion, in addition, audio and video codec can be done by pure software, but also by the use of specialized hardware chip to complete. Playback process Remove the relevant files from the storage device and decode them according to the encoding used in the recording process.

application scenarios to analyze (mainly RTP this piece). The students who make up the decoding, hold on! Another super-complicated change is that audio defines several output flags (see Audio.h's audio_output_flags_t enumeration definition). According to the note, this value has two functions, and one is that the user of at is able to indicate what kind of outputdevice they want to use. The other is the device manufacturer can declare its own

Due to historical reasons, there are multiple API systems available for sound programming in Linux. However, without proper guidance, it is difficult to find a system that suits your needs. Here is a guide written by Lennart Poettering 　　The simplest guide is to ask yourself: "What do I want (using the Linux Sound API) to do? ". The answer is as follows:I want to write a media player-like software! Then use GStreamer. Unless you only want to program

. There are, of course, more channel numbers. For example, the channel is many, the effect is good, two channels, indicating that only the left and right side of the voice transmitted over, four-channel, explained before and after the voice passed overBit rate (bitrate)Also called bit rate. For encoding format, indicates the amount of audio data per second after compression encoding. Calculation formula: Bit rate = sample rate × Sample accuracy X numb

are:PS: The synthesized samplerate is lower than the output frequency of the sound card work, must pass the software src (Sample rate Converter), otherwise come out is the quick release effect, the default parameter when creating the source is to open its built-in SRC.Required header files and static libraries#include #include #pragma comment (lib, "Xaudio2.lib")/* #include "SpeechSynthesis.h" This is Speechsynthesis's header file */speechsynthesis S

formats required); 2. Player software support. At present, the remux source on the network already carries these next-generation HD audio tracks, so whether the player can provide the source code output function becomes an important performance benchmark. Multi-channel lpcm: the original format of lossless audio tracks. It is equivalent to a wave file and does n

Two ways of synchronizing audio and video synchronization with the time stamp as the benchmark with the time stamp of the video as the benchmark with the external clock as the reference to the standard audio-video synchronization This is a digression. This blog theme is not write audio and video synchronization but write audi