lossless audio

suitable for this encoding format. · AMR: The full name of AMR is "adaptive multi-rate". It is also another encoding format specially optimized for "speech" and is also suitable for Low Bit Rate environments. · ALAC: The full name is "Apple lossless". This is an audio encoding method without any quality loss, that is, lossless compression. In actual use, it can

, AAC has a better compression rate than MP3, especially at the same bit rate. HE-AAC: HE-AAC is an extension of AAC, this "he" is actually "high efficiency ". The HE-AAC is the best audio at low bit rates, just like streaming audio. AMR: AMR is actually "adaptive multi-rate", which is another fast and effective encoding method at low bit rates. ALAC: it is actually "Apple

bandwidth for the audio file format is 20KHZ. According to the Nyquist theory, only if the sampling frequency is higher than twice times the maximum frequency of the sound signal, the voice of the digital signal can be restored to the original sound, so the audio file sampling rate is generally 40~50khz, such as the most common CD quality sampling rate of 44.1KHZ. The process of sampling and quantifying so

To play and record audio on iOS devices, Apple recommends that we use the Avaudioplayer and Avaudiorecorder classes in the Avfoundation framework. Although the usage is simpler, it does not support streaming; This means that before playing the audio, you must wait until the entire audio load is complete before you can start playing the

This article mainly introduces the composition of AudioUnit This article is made up of your own understanding. If you have any errors, please contact our netizens for correction. Understand the architecture of Audio Unit Before we started, we realized audioUnit through an AudioUnit object, which is an Effect type AudioUnit of this type. This unit consists of many small scopes, scope has element, elementt has channel, stream format, and properties. Do

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

. after a simulated signal is sampled and quantified, a quantified pulse amplitude modulation signal is obtained, which is only a finite number. EncodingA group of binary code groups is used to represent each quantified value with a fixed level. however, quantization is actually completed in the encoding process at the same time. Therefore, the encoding process is also called a modulo/number transformation, which can be recorded as A/D. [Features] PCM signals are not encoded or compressed (

Real-time audio and video domain UDP is the king In the Internet, audio and video real-time interaction using the Transport Layer Scheme has TCP (such as: RTMP) and UDP (such as: RTP) two kinds. The TCP protocol can provide a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks

BasisLet's take a quick look at some basic audio knowledge.At present, we need to rely on audio files for audio playback on the computer, audio file generation process is the sound information sampling, quantization and encode the digital signal generated by the process, the sound that the human ear can hear, the lowes

for the audio file format is 20KHZ. According to the Nyquist theory, only if the sampling frequency is higher than twice times the maximum frequency of the sound signal, the voice of the digital signal can be restored to the original sound, so the audio file sampling rate is generally 40~50khz, such as the most common CD quality sampling rate of 44.1KHZ.The process of sampling and quantifying sound is call

(speech)" Optimized encoding format, is also suitable for low bit rate environment to adopt.· ALAC: Its full name is "Apple Lossless", which is an audio encoding without any loss of quality, which is what we call lossless compression. It is capable of compressing 40%-60% of raw data during actual use. This encoding format is very fast decoding, which is ideal fo

of these quality depends on the sound bit rate. When the bit rate is suitable, the loss is hard to hear. In fact, AAC ratio MP3 better compression rate, especially when the bit rate is less than 128bit/S . HE-AAC:HE-AACYesAACOneSuperset, this"He"It indicates "high efficiency ". HE-AAC is a kind of audio encoding format specially optimized for low bit rate, such as streaming audioThis encoding format is particularly suitable. Amr:AmrFu

MP3 is only a type of audio format. However, audio has several important parameters, such as kHz, bit, audio channel, and kbps. the formats are different, and the algorithms are different. Therefore, when the above parameters are the same, the quality of different formats varies greatly. VBR is a kind of dynamic sampling. For more information, see the following d

the convenience of audio algorithm processing/transmission, it is generally set to Ms ~ 60 ms indicates that the unit of data is one frame of audio. This time is called "sampling time", and there is no special standard in its length. It is determined based on the requirements of the decoder and specific applications, we can calculate the size of an audio frame:

Recently my wife gave me a death order, let me the computer's "broken speakers" to replace, she really can not endure to listen to music without surround sound, watching movies without overweight bass, play no game stereo sound, singing KTV simply do not sound ... Vulgar speech: "Head can be broken, blood flow, wife orders can not be lost." "Yes, this time must be excellent to complete the task of wife confessed!" Can change the speaker to spend money, if not change the speaker can also bring th

article shows that in the case does not support HTML5, will be flash play, but in support of HTML5 Firefox play MP4, but encountered great difficulty, although the call to Flash, but has been unable to play (but I have always suspected that my Firefox flash has a problem , don't know if it's true). But if you follow Videojs's advice and put in two-format videos, you won't have this problem.In addition to the writing of video also has a special for flash writing, of course, you can also use this

This article focuses on how to capture a single frame of audio data on the Android platform. Before reading this article, it is recommended to read my previous article, "Android Audio Development (1): Fundamentals", because in the audio development process, often involves these basic knowledge, after mastering these important concepts, the development process of