aac

问题 I've successfully registered widevine in my audio player, so I can log in to Spotify Web Player, but I can't play any audio track becuase the appropriate codec is still missing. As I know the web player is using AAC format so somehow I should add AAC support to CEF. Have you guys got any idea how should I do this? 回答1: You need a custom build of CEF with proprietary codecs enabled (use proprietary_codecs=true ffmpeg_branding=Chrome build flags). You can find instructions for building CEF with

问题 I am trying to modify an iOS app that can send audio files (securely/encrypted) to other iOS devices. It uses AVAudioRecorder to record the audio files, and it uses AVAudioPlayer to play back any received audio files. I am now trying to modify the app to create and send files that are compatible with Android. I modified the existing AVAudioRecorder code to this: [settings setValue:[NSNumber numberWithInt:kAudioFormatMPEG4AAC] forKey:AVFormatIDKey]; [settings setValue:[NSNumber numberWithInt

问题 I am trying to modify an iOS app that can send audio files (securely/encrypted) to other iOS devices. It uses AVAudioRecorder to record the audio files, and it uses AVAudioPlayer to play back any received audio files. I am now trying to modify the app to create and send files that are compatible with Android. I modified the existing AVAudioRecorder code to this: [settings setValue:[NSNumber numberWithInt:kAudioFormatMPEG4AAC] forKey:AVFormatIDKey]; [settings setValue:[NSNumber numberWithInt

问题 I want to covert a AAC to MP3 and give metadata of comment to the MP3 file using ffmpeg. But -metadata comment doesn't work and ffmpeg doesn't return any error. My code is ffmpeg -i "test.aac" -ab 128k -metadata comment='this is test' "test.mp3" I tried -metadata description='this is test' and also update ffmpeg. Other function such as -metadata artist and -metadata album works well. What's wrong with this code? Output Stream mapping: Stream #0:0 -> #0:0 (aac (native) -> mp3 (libmp3lame))

问题 I have two commands, one in which I specify the output format as -f mp4 : ffmpeg -i input.wav -ac 1 -c:a libfdk_aac -profile:a aac_he -vbr 2 -f mp4 test_export_1.m4a And one which lets ffmpeg guess based on the extension: ffmpeg -i input.wav -ac 1 -c:a libfdk_aac -profile:a aac_he -vbr 2 test_export_2.m4a This gives me two different results. The first one: $ffprobe test_export_1.m4a Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test_export_1.m4a': Metadata: major_brand : isom minor_version : 512

问题 I try to find any info about AAC HEv2 (PS) in an MPEG Transport Stream (TS) for HLS. According to the HLS Authoring Specification for Apple Devices AAC HEv2 is a supported format. AAC HEv2 is part of MPEG-4, but I cannot understand how HEv2 could fit into a transport stream. SBR (or HEv1) can be in TS by implicit signaling. In case of MP4 we have an audio specific config. But how can I multiplex AAC Parametric Stereo into the TS? Is it available or not? I cannot find any info from the Apple

问题 i was trying to play AAC audio stream in android MediaPlayer, as mentioned here and also here author claimed the problem was ignoring position argument so i made a little setup to test this i will record using recorder and save it to a buffer and feed this buffer to MediaPlayer according to this // parcel pipe: 1: write // 0: read ParcelFileDescriptor[] pfd = ParcelFileDescriptor.createPipe(); // a Good Recorder! final MediaRecorder mMediaRecorder = new MediaRecorder(); mMediaRecorder

问题 i was trying to play AAC audio stream in android MediaPlayer, as mentioned here and also here author claimed the problem was ignoring position argument so i made a little setup to test this i will record using recorder and save it to a buffer and feed this buffer to MediaPlayer according to this // parcel pipe: 1: write // 0: read ParcelFileDescriptor[] pfd = ParcelFileDescriptor.createPipe(); // a Good Recorder! final MediaRecorder mMediaRecorder = new MediaRecorder(); mMediaRecorder

来源： https://stackoverflow.com/questions/57198419/when-webm-extracts-aac-the-duration-is-inconsistent

问题 We have an encoding process in place using ffmpeg on Mac OS X. This process will take a source video and a couple files from that: .m3u8 video, .mp4 video and .mp3 audio file. By default we've used the video from our m3u8 process which is a h264 (via libx264 ) video with AAC (via libfaac ) audio. We are mostly using these videos on mobile devices (hence the m3u8 files) but we also use the .mp4 files for Android, Windows Phone, etc. More and more we also need to offer these same videos on the