what "buffer too early (), down-sampling" means

[rhbc73]

Hi mates,

I'm developing a mepg ts muxer which could produce ts file with H.264 encoded video and AAC encoded audio.
But when I play my video in VLC player (win 2.0.1) there's warning in log window

main debug: processing request item: 1_0.ts, node: Playlist, skip: 0
main debug: resyncing on 1_0.ts
main debug: 1_0.ts is at 0
main debug: starting playback of the new playlist item
main debug: creating new input thread
main debug: Creating an input for '1_0.ts'
main debug: using timeshift granularity of 50 MiB, in path 'C:\Users\Alex\AppData\Local\Temp'
main debug: `file:///C:/ProgramData/Web/LiveHigh/1_0.ts' gives access `file' demux `' path `/C:/ProgramData/Web/LiveHigh/1_0.ts'
main debug: creating demux: access='file' demux='' location='/C:/ProgramData/Web/LiveHigh/1_0.ts' file='C:\ProgramData\Web\LiveHigh\1_0.ts'
main debug: looking for access_demux module: 3 candidates
main debug: no access_demux module matching "file" could be loaded
main debug: TIMER module_need() : 0.378 ms - Total 0.378 ms / 1 intvls (Avg 0.378 ms)
main debug: creating access 'file' location='/C:/ProgramData/Web/LiveHigh/1_0.ts', path='C:\ProgramData\Web\LiveHigh\1_0.ts'
main debug: looking for access module: 3 candidates
filesystem debug: opening file `C:\ProgramData\Web\LiveHigh\1_0.ts'
main debug: using access module "filesystem"
main debug: TIMER module_need() : 0.266 ms - Total 0.266 ms / 1 intvls (Avg 0.266 ms)
main debug: Using stream method for AStream*
main debug: starting pre-buffering
main debug: received first data after 0 ms
main debug: pre-buffering done 1024 bytes in 0s - 22727 KiB/s
main debug: looking for stream_filter module: 4 candidates
main debug: no stream_filter module matching "any" could be loaded
main debug: TIMER module_need() : 0.130 ms - Total 0.130 ms / 1 intvls (Avg 0.130 ms)
main debug: looking for stream_filter module: 1 candidate
main debug: using stream_filter module "stream_filter_record"
main debug: TIMER module_need() : 0.110 ms - Total 0.110 ms / 1 intvls (Avg 0.110 ms)
main debug: creating demux: access='file' demux='' location='/C:/ProgramData/Web/LiveHigh/1_0.ts' file='C:\ProgramData\Web\LiveHigh\1_0.ts'
main debug: looking for demux module: 55 candidates
ts warning: lost synchro
ts debug: skipping 94 bytes of garbage
qt4 debug: IM: Setting an input
ts warning: lost synchro
ts debug: skipping 94 bytes of garbage
ts warning: lost synchro
ts debug: skipping 94 bytes of garbage
ts warning: lost synchro
ts debug: skipping 94 bytes of garbage
ts warning: lost synchro
ts debug: skipping 94 bytes of garbage
ts debug: PATCallBack called
ts debug: new PAT ts_id=1 version=0 current_next=1
ts debug: * number=0 pid=31
ts debug: * number=1 pid=256
ts debug: PMTCallBack called
ts debug: new PMT program number=1 version=0 pid_pcr=4097
ts error: MPEG-4 descriptor not found
ts debug: * es pid=4352 type=15 fcc=mp4a
main debug: selecting program id=1
ts debug: * es pid=4113 type=27 fcc=h264
filesystem warning: unimplemented query 4097 in control
main debug: using demux module "ts"
main debug: TIMER module_need() : 18.119 ms - Total 18.119 ms / 1 intvls (Avg 18.119 ms)
main debug: looking for a subtitle file in C:\ProgramData\Web\LiveHigh
main debug: looking for decoder module: 31 candidates
main debug: using decoder module "faad"
main debug: TIMER module_need() : 0.134 ms - Total 0.134 ms / 1 intvls (Avg 0.134 ms)
main debug: looking for packetizer module: 21 candidates
packetizer_mpeg4audio debug: running MPEG4 audio packetizer
packetizer_mpeg4audio debug: no decoder specific info, must be an ADTS or LOAS stream
main debug: using packetizer module "packetizer_mpeg4audio"
main debug: TIMER module_need() : 0.071 ms - Total 0.071 ms / 1 intvls (Avg 0.071 ms)
main debug: looking for decoder module: 31 candidates
avcodec debug: libavcodec already initialized
avcodec debug: trying to use direct rendering
avcodec debug: allowing 4 thread(s) for decoding
avcodec debug: ffmpeg codec (H264 - MPEG-4 AVC (part 10)) started
avcodec debug: using frame thread mode with 4 threads
main debug: using decoder module "avcodec"
main debug: TIMER module_need() : 1.082 ms - Total 1.082 ms / 1 intvls (Avg 1.082 ms)
main debug: looking for packetizer module: 21 candidates
main debug: using packetizer module "packetizer_h264"
main debug: TIMER module_need() : 0.090 ms - Total 0.090 ms / 1 intvls (Avg 0.090 ms)
ts debug: DEMUX_SET_GROUP 0 00000000
main debug: looking for meta reader module: 2 candidates
lua debug: Trying Lua scripts in C:\Users\Alex\AppData\Roaming\vlc\lua\meta\reader
lua debug: Trying Lua scripts in C:\Program Files\VideoLAN\VLC\lua\meta\reader
lua debug: Trying Lua playlist script C:\Program Files\VideoLAN\VLC\lua\meta\reader\filename.luac
main debug: no meta reader module matching "any" could be loaded
main debug: TIMER module_need() : 0.792 ms - Total 0.792 ms / 1 intvls (Avg 0.792 ms)
main debug: `file:///C:/ProgramData/Web/LiveHigh/1_0.ts' successfully opened
ts debug: pid[31] unknown
ts warning: first packet for pid=4113 cc=0x1
packetizer_h264 debug: found NAL_SPS (sps_id=0)
packetizer_h264 debug: found NAL_PPS (pps_id=0 sps_id=0)
main debug: looking for text renderer module: 2 candidates
freetype debug: Using Arial as font from file C:\Windows\Fonts\arial.ttf
freetype debug: using fontsize: 2
main debug: using text renderer module "freetype"
main debug: TIMER module_need() : 0.710 ms - Total 0.710 ms / 1 intvls (Avg 0.710 ms)
main debug: looking for video filter2 module: 18 candidates
ts warning: first packet for pid=4352 cc=0x1
packetizer_mpeg4audio debug: detected ADTS format
packetizer_mpeg4audio info: AAC channels: 2 samplerate: 22050
ts debug: pid[4097] unknown
main debug: Buffering 0%
main debug: Buffering 30%
main debug: Buffering 60%
main debug: Buffering 90%
swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality)
main debug: using video filter2 module "swscale"
main debug: TIMER module_need() : 0.406 ms - Total 0.406 ms / 1 intvls (Avg 0.406 ms)
main debug: looking for video filter2 module: 18 candidates
main debug: Stream buffering done (360 ms in 0 ms)
yuvp debug: YUVP to YUVA converter
main debug: using video filter2 module "yuvp"
main debug: TIMER module_need() : 0.083 ms - Total 0.083 ms / 1 intvls (Avg 0.083 ms)
main debug: Deinterlacing available
faad warning: decoded zero sample
main debug: deinterlace 0, mode blend, is_needed 0
main debug: Opening vout display wrapper
main debug: looking for vout display module: 6 candidates
direct3d debug: creating Vout EventThread
direct3d debug: DirectXCreateWindow
main debug: looking for vout window hwnd module: 2 candidates
qt4 debug: requesting video...
qt4 debug: Video was requested 0, 0
main debug: using vout window hwnd module "qt4"
main debug: TIMER module_need() : 60.796 ms - Total 60.796 ms / 1 intvls (Avg 60.796 ms)
direct3d debug: created video sub-window
direct3d debug: Vout EventThread running
direct3d debug: disabling screen saver
main debug: VoutDisplayEvent 'resize' 656x518 window
direct3d warning: trying surface pixel format: YV12
direct3d debug: selected surface pixel format is YV12
direct3d debug: Using D3DTEXF_LINEAR for minification
direct3d debug: Using D3DTEXF_LINEAR for magnification
direct3d debug: Direct3D scene created successfully
direct3d debug: Direct3D device adapter successfully initialized
main debug: using vout display module "direct3d"
main debug: TIMER module_need() : 102.179 ms - Total 102.179 ms / 1 intvls (Avg 102.179 ms)
main debug: original format sz 640x480, of (0,0), vsz 640x480, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0
main debug: removing module "freetype"
main debug: looking for text renderer module: 2 candidates
main debug: VoutDisplayEvent 'resize' 640x480 window
freetype debug: Using Arial as font from file C:\Windows\Fonts\arial.ttf
freetype debug: using fontsize: 2
main debug: using text renderer module "freetype"
main debug: TIMER module_need() : 0.679 ms - Total 0.679 ms / 1 intvls (Avg 0.679 ms)
avcodec debug: using direct rendering
main debug: recycling audio output
main debug: looking for audio output module: 2 candidates
aout_directx debug: Opening DirectSound Audio Output
aout_directx debug: found device: Primary Sound Driver
aout_directx debug: found device: Speakers (Conexant HD Audio)
aout_directx debug: using device: Speakers (Conexant HD Audio)
main debug: End of video preroll
main debug: Received first picture
direct3d debug: Created 83x33 texture for OSD
aout_directx debug: device supports 2 channels (DEFAULT!)
main debug: Post-processing available
aout_directx debug: device supports 1 channel
aout_directx debug: Windows says your SpeakerConfig is stereo
aout_directx debug: creating DirectSoundThread
main debug: using audio output module "aout_directx"
main debug: TIMER module_need() : 16.856 ms - Total 16.856 ms / 1 intvls (Avg 16.856 ms)
aout_directx debug: DirectSoundThread ready
main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes
main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes
main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo
main debug: looking for audio filter module: 14 candidates
audio_format debug: f32l->s16l, bits per sample: 32->16
main debug: using audio filter module "audio_format"
main debug: TIMER module_need() : 0.115 ms - Total 0.115 ms / 1 intvls (Avg 0.115 ms)
main debug: conversion pipeline completed
main debug: looking for audio mixer module: 2 candidates
main debug: using audio mixer module "float32_mixer"
main debug: TIMER module_need() : 0.056 ms - Total 0.056 ms / 1 intvls (Avg 0.056 ms)
main debug: input 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes
main debug: looking for audio filter module: 1 candidate
scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32
scaletempo debug: params: 30 stride, 0.200 overlap, 14 search
scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode
main debug: using audio filter module "scaletempo"
main debug: TIMER module_need() : 0.119 ms - Total 0.119 ms / 1 intvls (Avg 0.119 ms)
main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo
main debug: conversion pipeline completed
main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo
main debug: conversion pipeline completed
main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo
main debug: looking for audio filter module: 14 candidates
main debug: using audio filter module "samplerate"
main debug: TIMER module_need() : 0.142 ms - Total 0.142 ms / 1 intvls (Avg 0.142 ms)
main debug: conversion pipeline completed
main debug: End of audio preroll
main debug: Decoder buffering done in 154 ms
main warning: buffer too late (86037), up-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too late (172517), up-sampling
main warning: buffer way too late (215735), dropping buffer
main warning: buffer way too late (212568), dropping buffer
main warning: buffer way too late (209346), dropping buffer
main warning: buffer way too late (206123), dropping buffer
main warning: buffer way too late (202946), dropping buffer
main warning: resampling stopped after 145674 usec (drift: 21923)
main warning: buffer too early (-55444), down-sampling
main warning: timing screwed, stopping resampling
main warning: audio output out of sync, adjusting dates (101946 us)
main warning: buffer too early (-114302), down-sampling
main warning: not synchronized (101947 us), resampling
main debug: picture might be displayed late (missing 0 ms)
main warning: resampling stopped after 180015 usec (drift: 71612)
main warning: buffer too early (-90973), down-sampling
main warning: buffer way too early (-133338), clearing queue
main warning: timing screwed, stopping resampling
main debug: audio output is starving (-605062), playing silence
main debug: auto hiding mouse cursor
main warning: buffer too early (-58229), down-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too early (-115687), down-sampling
main warning: buffer way too early (-128481), clearing queue
main warning: timing screwed, stopping resampling
main warning: buffer too early (-44167), down-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too early (-78207), down-sampling
main warning: timing screwed, stopping resampling
main warning: buffer way too early (-121723), clearing queue
main warning: buffer too late (62586), up-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too late (96980), up-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too late (153921), up-sampling
main warning: buffer way too late (184537), dropping buffer
main warning: buffer way too late (184449), dropping buffer
main warning: buffer way too late (186738), dropping buffer
main warning: buffer way too late (180526), dropping buffer
main warning: buffer way too late (185513), dropping buffer
main warning: resampling stopped after 9450064 usec (drift: -3125)
main warning: buffer too early (-41405), down-sampling
main warning: timing screwed, stopping resampling
main warning: buffer too early (-74983), down-sampling
main warning: resampling stopped after 6745807 usec (drift: 43383)
main warning: buffer too early (-44109), down-sampling
ts debug: eof ?
main debug: EOF reached
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: finished input
main debug: removing module "faad"
main debug: killing decoder fourcc `mp4a', 0 PES in FIFO
main debug: removing module "scaletempo"
main debug: removing module "samplerate"
main debug: removing module "aout_directx"
aout_directx debug: closing audio device
aout_directx debug: DirectSoundThread exiting
main debug: removing module "audio_format"
main debug: removing module "float32_mixer"
main debug: releasing audio output
main debug: removing module "packetizer_mpeg4audio"
main debug: removing module "avcodec"
avcodec debug: ffmpeg codec (H264 - MPEG-4 AVC (part 10)) stopped
main debug: killing decoder fourcc `h264', 0 PES in FIFO
main debug: saving a free vout
main debug: reusing provided vout
main debug: removing module "packetizer_h264"
main debug: removing module "ts"
ts debug: pid list:
ts debug: - pid[0] seen
ts debug: - pid[31] seen
ts debug: - pid[256] seen
ts debug: - pid[4097] seen
ts debug: - pid[4113] seen
main debug: Program doesn't contain anymore ES
ts debug: - pid[4352] seen
ts debug: - pid[8191] seen
main debug: removing module "stream_filter_record"
main debug: removing module "filesystem"
main debug: dead input
main debug: changing item without a request (current 0/1)
main debug: nothing to play
main debug: destroying useless vout
main debug: removing module "direct3d"
direct3d debug: Direct3D scene released successfully
direct3d debug: DirectXEventThread terminating
direct3d debug: DirectXCloseWindow
direct3d debug: WinProc WM_DESTROY
qt4 debug: releasing video...
qt4 debug: IM: Deleting the input
main debug: TIMER input launching for '1_0.ts' : 23.160 ms - Total 23.160 ms / 1 intvls (Avg 23.160 ms)
qt4 debug: Video is not needed anymore
main debug: removing module "freetype"
main debug: removing module "yuvp"
main debug: removing module "swscale"


Audio sounds good in VLC player, as the log said, player may do some up/down sampling to make the
sound as smooth/good as possible.

But my ts file is targeted for iPhone, when it's played with iPhone's Quicktime, the audio scratched. I
think that's because iPhone Quicktime doesn't do any sophisticated compensation for this.

So I'm eager to now, what will cause these kind of up/down sampling, so that I can check my source code to avoid that.

Great thanks!