bitrate

问题 I've use pydub to output a file(chop the file into shorter one), everything is great, but the bitrate has changed from 256k to 124k(why I will get this number instead 128k?). I know that AudioSegment has an argument to set bitrate, but I just want the same bitrate instead manually set every time. Any way to fix this issue? 回答1: This has mainly to do with ffmpeg/avlib, but you can pass a flag to the AudioSegment().export() method to specify the bitrate you'd like: from pydub import

问题 I am new to video encoding so bear with me. I am using FFMPEG. I have an mp4 file which is 640 x 350 with an average bitrate of around 2000kb (I think) and a filesize of 80Mb. I want to convert this to an ogv file with a much lower bit rate (128kb) but the same width and height. I am using the following command... ffmpeg -i input.mp4 -b:v 128k output.ogv ... but FFMPEG seems to ignore my bitrate option and outputs a file with a bitrate of around 600kb and a filesize of around 3Mb. I can do

问题 I am new to video encoding so bear with me. I am using FFMPEG. I have an mp4 file which is 640 x 350 with an average bitrate of around 2000kb (I think) and a filesize of 80Mb. I want to convert this to an ogv file with a much lower bit rate (128kb) but the same width and height. I am using the following command... ffmpeg -i input.mp4 -b:v 128k output.ogv ... but FFMPEG seems to ignore my bitrate option and outputs a file with a bitrate of around 600kb and a filesize of around 3Mb. I can do

问题 I am new to video encoding so bear with me. I am using FFMPEG. I have an mp4 file which is 640 x 350 with an average bitrate of around 2000kb (I think) and a filesize of 80Mb. I want to convert this to an ogv file with a much lower bit rate (128kb) but the same width and height. I am using the following command... ffmpeg -i input.mp4 -b:v 128k output.ogv ... but FFMPEG seems to ignore my bitrate option and outputs a file with a bitrate of around 600kb and a filesize of around 3Mb. I can do

问题 I am trying to record an audio file in Android, I'm setting the output file bit rate and sampling rate, everything is working right but whenever i record anything in a different device the file size differs a lot. I have made some tests with a Z2 and a Moto G changing bit rate and sampling rate, obtaining very different file sizes on same recording time. I have noticed that the file depends the most on bit rate rather than sampling rate. The problem is that i would expect and actually need

问题 How can I get the properties of an audio file (like the number of channels, sample rate and bit rate) programmatically in Objective-C for iOS? 回答1: To get the properties for any audio filetype that is supported on iOS, load the audio data into an AVAudioPlayer and examine its properties (in particular, the settings dictionary). 来源： https://stackoverflow.com/questions/4929825/how-can-i-read-the-properties-of-an-audio-file-in-objective-c-for-ios

问题 How can you list all files (recursively) within a directory where the file (audio) bit rate is greater than 32kbps using powershell? 回答1: Starting with the link (new link location) from Johannes' post, here is a simple function that uses GetDetailsOf to find all mp3 files in a directory with a minimum bitrate: function Get-Mp3Files( [string]$directory = "$pwd", [int]$minimumBitrate = 32 ) { $shellObject = New-Object -ComObject Shell.Application $bitrateAttribute = 0 # Find all mp3 files under

问题 I am trying to fetch bit rate at which the AVPlayer is playing the video stream . I tried with observed bit rate property of AVPlayerItemAccessLogEvent , but it gives very high value.After exploring further on Observed-bit-rate property I understood that this property shows the value of download rate of a segment .I need to know the bit rate at which the player is playing the stream. The graph below shows Observed bit vs Indicated bit rate for the LIVE stream. Kindly give your suggestion on

问题 Currently, I'm using mic.rate = 100; This only gives 63kbps. Is it possible for Flash AS3 to set bitrate higher than 63kbps? 回答1: From the docs Acceptable values are 5, 8, 11, 22, and 44 So enter one of those. And it's measured in kHz, not kbps, also according to the docs 回答2: It's flash. Great quality also depends on users' hardware. You didn't post full settings for your microphone. Also value that you are using isn't valid. Here a small snippet, for mic settings, that will give you good

问题 I have a 1 second PCM data which I write into an AAC file successfully. However, I can not control the bitrate of the output file. Here is the configuration of my AAC codec: AudioStreamBasicDescription clientFormat = {0}; clientFormat.mSampleRate = 44100; clientFormat.mFormatID = kAudioFormatMPEG4AAC; clientFormat.mFormatFlags = kMPEG4Object_AAC_Main; clientFormat.mChannelsPerFrame = 2; clientFormat.mBytesPerPacket = 0; clientFormat.mBytesPerFrame = 0; clientFormat.mFramesPerPacket = 1024;