[FFmpeg-user] audio only AAC in HLS
Guido Holz
guido at sportograf.de
Wed Feb 17 13:19:20 CET 2016
Hi,
I'm trying to encode a aac file that is playable in an embedded player. All
I have tried doesn't work.
I've downloaded an example from a radio-live-stream.
auido.aac is my generated AAC file and the long one is what I've copied
from the live tream
\> ffmpeg -i audio_root.aac -c:a libfdk_aac -b:a 64k audio.aac
\> ffmpeg -i audio.aac
Input #0, aac, from 'audio.aac':
Duration: 00:04:35.31, bitrate: 63 kb/s
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 63 kb/s
\> ffmpeg -i
../../tmp/media_w370587926_b160000_ao_slen_t64RW5nbGlzaA\=\=_4.aac
Input #0, aac, from
'../../tmp/media_w370587926_b160000_ao_slen_t64RW5nbGlzaA==_4.aac':
Duration: 00:00:05.80, bitrate: 165 kb/s
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 165 kb/s
\> ffprobe -v quiet -print_format json -show_format -show_streams audio.aac
{
"streams": [
{
"index": 0,
"codec_name": "aac",
"codec_long_name": "AAC (Advanced Audio Coding)",
"profile": "LC",
"codec_type": "audio",
"codec_time_base": "1/44100",
"codec_tag_string": "[0][0][0][0]",
"codec_tag": "0x0000",
"sample_fmt": "fltp",
"sample_rate": "44100",
"channels": 2,
"channel_layout": "stereo",
"bits_per_sample": 0,
"r_frame_rate": "0/0",
"avg_frame_rate": "0/0",
"time_base": "1/28224000",
"duration_ts": 7770301271,
"duration": "275.308293",
"bit_rate": "63738",
"disposition": {
"default": 0,
"dub": 0,
"original": 0,
"comment": 0,
"lyrics": 0,
"karaoke": 0,
"forced": 0,
"hearing_impaired": 0,
"visual_impaired": 0,
"clean_effects": 0,
"attached_pic": 0
}
}
],
"format": {
"filename": "audio.aac",
"nb_streams": 1,
"nb_programs": 0,
"format_name": "aac",
"format_long_name": "raw ADTS AAC (Advanced Audio Coding)",
"duration": "275.308293",
"size": "2193450",
"bit_rate": "63738",
"probe_score": 51
}
}
\>ffprobe -v quiet -print_format json -show_format -show_streams
../../tmp/media_w370587926_b160000_ao_slen_t64RW5nbGlzaA\=\=_4.aac
{
"streams": [
{
"index": 0,
"codec_name": "aac",
"codec_long_name": "AAC (Advanced Audio Coding)",
"profile": "LC",
"codec_type": "audio",
"codec_time_base": "1/44100",
"codec_tag_string": "[0][0][0][0]",
"codec_tag": "0x0000",
"sample_fmt": "fltp",
"sample_rate": "44100",
"channels": 2,
"channel_layout": "stereo",
"bits_per_sample": 0,
"r_frame_rate": "0/0",
"avg_frame_rate": "0/0",
"time_base": "1/28224000",
"duration_ts": 163676160,
"duration": "5.799184",
"bit_rate": "165375",
"disposition": {
"default": 0,
"dub": 0,
"original": 0,
"comment": 0,
"lyrics": 0,
"karaoke": 0,
"forced": 0,
"hearing_impaired": 0,
"visual_impaired": 0,
"clean_effects": 0,
"attached_pic": 0
}
}
],
"format": {
"filename":
"../../tmp/media_w370587926_b160000_ao_slen_t64RW5nbGlzaA==_4.aac",
"nb_streams": 1,
"nb_programs": 0,
"format_name": "aac",
"format_long_name": "raw ADTS AAC (Advanced Audio Coding)",
"duration": "5.799184",
"size": "119953",
"bit_rate": "165475",
"probe_score": 51
}
}
I can't see any differences.. but I know I have to add ID3 and ADTS to the
aac-files.
\>exiftool audio.aac
ExifTool Version Number : 9.46
File Name : audio.aac
Directory : .
File Size : 2.1 MB
File Modification Date/Time : 2016:02:17 13:12:10+01:00
File Access Date/Time : 2016:02:17 13:12:47+01:00
File Inode Change Date/Time : 2016:02:17 13:12:10+01:00
File Permissions : rw-rw-r--
Error : Unknown file type
\>exiftool
../../tmp/media_w370587926_b160000_ao_slen_t64RW5nbGlzaA\=\=_4.aac
ExifTool Version Number : 9.46
File Name :
media_w370587926_b160000_ao_slen_t64RW5nbGlzaA==_4.aac
Directory : ../../tmp
File Size : 117 kB
File Modification Date/Time : 2015:12:14 23:17:45+01:00
File Access Date/Time : 2016:02:17 12:27:03+01:00
File Inode Change Date/Time : 2016:02:13 21:33:53+01:00
File Permissions : rw-rw-r--
File Type : MP3
MIME Type : audio/mpeg
ID3 Size : 73
Private : (Binary data 8 bytes, use -b option to
extract)
You see ID3 and mp3 as file type? What command can I use to see differences
for ID3 and ADTS?
thanks
More information about the ffmpeg-user
mailing list