aac to m4r

Player 9,0,115,0 and higher.Speex is supported in Flash Player and higher. Soundrate UB [2] Sampling rate. The following values are defined:0 = 5.5 KHz1 = one kHz2 = KHz3 = About KHz Soundsize UB [1] Size of each audio sample. This parameter only pertains toUncompressed formats. Compressed formats always decodeto + bits internally.0 = 8-bit Samples1 = 16-bit Samples Soundtype UB [1] Mono or stereo sound0 = Mono Sound1 =

how to make video and audio time-stampinghttp://blog.csdn.net/wfqxx/article/details/5497138 1. Video time stamp PTS = inc++ * (1000/fps); Where Inc is a static, initial value of 0, each time the timestamp Inc plus 1. In FFmpeg, the code in pkt.pts= m_nvideotimestamp++ * (M_vctx->time_base.num * 1000/m_vctx->time_base.den); 2. Audio time stamp PTS = inc++ * (frame_size * 1000/sample_rate) The code in FFmpeg is pkt.pts= m_naudiotimestamp++ * (m_actx->frame_size * 1000/m_actx->sample_rate); The s

Http://article.ityran.com/archives/253 This translation is produced by tairan translation team. Please indicate the source for reprinting and inform tairan !! Translation Before I developed the iPhone, Ray had little knowledge about sound formats. I know the differences between some "WAV" and "MP3" sound formats, but I certainly cannot tell you exactly what formats "AAC" and "caf" are, I also don't know what the best way to convert audio files on Mac

IPhoneDeveloped audio 101 (Part 1): file and Data Types Translator: Xia laiyou Disclaimer (read only !) : The original translations of all tutorials provided by this blog are from the Internet and are only for learning and communication purposes. Do not conduct commercial communications. At the same time, do not remove this statement when reprinting. In the event of any dispute, it has nothing to do with the owner of this blog and the person who published the translation. Thank you for

What is AAC? AAC, Advanced audio coding, is a technique for compressing digital audio files. officially part of the MPEG-4 standard, itIsMost widely used to create small digital audio files. The current variant is specified in ISO/IEC Standard 14496-3. AACIs conceptually similar to the ubiquitous MP3 format. like MP3, it takes advantage of deficiencies in human hearing to discard Digital bits correspondi

open source project android-aac-enc on github (Address: https://github.com/timsu/android-aac-enc ), this open-source project can perfectly encode the binary data in the original pcm format into a data file in the m4a format. Compared with the FFmpeg library, this library has the following advantages: 1. The aac-enc library is smaller than the FFmpeg library; 2.

FFmpeg is so powerful that you can do your job with a few simple commands.Put Darkdoor. [001-100].jpg sequence frames and 001.mp3 audio files use MPEG4 encoding to synthesize video files Darkdoor.avi:$ ffmpeg-i 001.mp3-i darkdoor.%3d.jpg-s 1024x768-author skypp-vcodec MPEG4 Darkdoor.aviFFmpeg also supports MOV format:$ ffmpeg-i darkdoor.%3d.jpg Darkdoor.movTo see which formats your ffmpeg supports, you can use the following command:$ ffmpeg-formats | LessYou can also export video files to JPG se

-Endian integer 16-bit variant of linear PCM), and iPhone preferred file format (reminder: Core audio file format, that is, CAFF ), you can enter the following command line: Afconvert-D lei16-F 'caff' input_file.xxx output_file.caf Note: I didn't point out the extension of this input file, because afconvert can intelligently detect the audio file type and convert it accordingly, therefore, the input file can be any data format or audio file format. Also note: you can a

and 172.16.8 .*C. SIPEntities: the entity that the SIP entity connects to. Generally, there are sessionmanager, CM, conference, and presence. For example, name (presence) and FQDNorIPAddress (ps address), Type (other), Location (ceibs-asm), TimeZone (shanghai), SIPLinkMonitoring (enabled), EntityLinks (ceibs-asm \ tls \ 5061 \ presence \ 5061 \ trusted) for example, ceibs-asm \ tcp udp \ 5060 \ conference \ 5060 \ trusted), and cm is also tlsD, EntityLinks, commonly used: asm-ps \ asm \ tls \

to three Hz. It uses the MS-frame to provide 40ms algorithm latency.The G722.1 achieves lower bit rates and greater compression than the g.722 codec. The goal is to achieve roughly the same quality as the g.722 at approximately half the bit rate. This license to use the code requires the authorization of Polycom Corporation.5, G722.1 Annex CThe G722.1 Annex C is based on Polycom's siren 14 compression technology, which uses 32kHz sampling frequency to support audio sampling from a range of up t

Http://www.cnblogs.com/ipinka/archive/2012/09/24/2699816.html This week, I completed a basic recording and playback function. I didn't know where to start at the beginning, and I also found a lot of related information. At the same time, I also learned a lot about audio, which is also helpful for the subsequent recording configuration. See the audio for iPhone development:File and data type (1), which is more enlightening. I. Audio Format There are still many Audio Encoding formats supported o

specific versions such as Android 3.0 + Table 1.Supported core media formats and codecs. Type Format/Codec Encoding Decoding Details Supported file types/contained formats Audio Aac lc/LTP • • Single-channel/Stereo content with a standard bitrate of up to 160 kbps and a sampling rate ranging from 8 to 48 khz in any combination • 3GPP (. 3GP)• MPEG-4 (.mp4,. m4a)• ADTs raw

In the previous article, we used ffmpeg to separate the audio and video data from a multimedia container, but it is possible that the data could not be decoded correctly. Why is it? Because the decoder needs to be configured before decoding the data, it is typical of the current popular HD coded "Golden Partner" combination H264 + AAC . This article will describe the key decoding configuration parameters of H264 and

Decoder Details Supported file types/containers and formats Audio Aac lc/LTP • • Use a standard bit rate of up to kbps and a sampling frequency of 8 to 48 khz to freely combine single-channel/Stereo content. • 3GPP (. 3gp)• MPEG-4 (.mp4,. m4a)• ADTS raw AAC (. aac, decoding: Android 3.1 +, encoding: Android 4.0

Today, I encountered a strange problem at oracle10g. One SQL statement was very fast on database 1 and very slow on Database 2. Database 2's data was imported from database 1 and the data volume was similar. Run 0.01 s on database 1. SQL> SELECT .*,2 B. INCREASE_ID,3 B. TRANSACTION_ID,4 B. LINK_CARD_ID,5 B. VALIDATE_FLAG,6 B. ASSET_VALUE_SHARING,7 B. RELATED_DEVICE_ID,8 B. PARENT_CARD_CODE,9 B. PROJECT_VALUE,10 B. DELETE_FLAG,11 B. DEPRECIATION_ADJUST_VALUE,12 T. TRANSACTION_MODE_CODE,13 T. TRAN

than 40 seconds! 5, after the change, in the playlist click on this music, right-click menu to choose to create the ACC version, created in the original music below you have changed to the ringtone music files, to see the length of music can be distinguished (see photo) 6, then select the modified Ringtone file, right-click menu in the Finder in the display, into the finder to find this ring, click to modify the suffix name, from the original "M4A" changed

Audio and video storage using the MPEG4IP library interface Mp4writesample Write-Audio Video frame (requires precise control of timestamp, time stamp can use relative value, that is, the current frame timestamp minus the timestamp of the previous frame) Mp4addh264videotrack adding H264 Track (timescale time factor parameter is important, based on the timestamp type definition of the video frame, 90000 (video standard sample timestamp) or (MS Timestamp)) Mp4addaudiotrack Add audio

ffmpeg.exe-I f: \ hacker house \ hacker Song-AB 56-ar 22050-B 500-R 15-s 320x240 F: \ 11.flv FFMPEG-I f: \ 01.wmv-AB 56-ar 22050-B 500-R 15-s 320x240 F: \ test. FLV use the-SS parameter (time_off set the Start Time Offset) to start the conversion task from a specified time point. For example: capture the thumbnail while converting the file format: FFMPEG-I "test. avi "-y-F image2-SS 8-T 0.001-s 350x240 'test.jpg ' capture existing FLV images: FFMPEG-I" test. FLV "-y-F image2-SS 8-T 0.001-s

kbit/s, at a frequency of up to a rate of up to three Hz. It uses the MS-frame to provide 40ms algorithm latency.The G722.1 achieves lower bit rates and greater compression than the g.722 codec. The goal is to achieve roughly the same quality as the g.722 at approximately half the bit rate. This license to use the code requires the authorization of Polycom Corporation.5, G722.1 Annex CThe G722.1 Annex C is based on Polycom's siren 14 compression technology, which uses 32kHz sampling frequency t

libvdpau-dev libsdl2-dev2.3 Dependent development libraries requiring stand-alone installationFirst create FFmpeg code directory, all the source code is placed in this directory# mkdir ~/ffmpeg_sources2.3.1 Installing fdk-aac-0.1.5:# cd ~/ffmpeg_sources# wget http://downloads.sourceforge.net/opencore-amr/fdk-aac-0.1.5.tar.gz# tar -zxvf fdk-aac-0.1.5.tar.gz mv f