Refer Here for more context to my question: https://gamedev.stackexchange.com/questions/136817/how-to-get-sdl2-to-play-32bit-wav-files
I have a 24bit WAV file that has an Audio Format PCM of 1, refer here: http://soundfile.sapp.org/doc/WaveFormat/ to AudioFormat
When converting my WAV file (24bit) to 16 bit using: ffmpeg -i input.wav -ar 48000 -ac 2 -acodec pcm_s16le output.wav it retains the Audio Format PCM = 0x001.
When using ffmpeg -i input.wav -ar 48000 -ac 2 -acodec pcm_s32le output.wav the Audio Format PCM = 0xfffe.
SDL2 (as seen in the parent question) only allows files to play with Linear PCM Audio Format (1), and I am unsure how using sox or ffmpeg how to convert my 24bit WAV files upwards to 32bit (as SDL2 only plays 32bit and 16bit).
Is what I'm asking possible? Some more information on WAV files and why ffmpeg changes the header number would be greatly appreciated.
FFmpeg uses the following code to set the codec tag
...
waveformatextensible = (par->channels > 2 && par->channel_layout) ||
par->channels == 1 && par->channel_layout && par->channel_layout != AV_CH_LAYOUT_MONO ||
par->channels == 2 && par->channel_layout && par->channel_layout != AV_CH_LAYOUT_STEREO ||
par->sample_rate > 48000 ||
par->codec_id == AV_CODEC_ID_EAC3 ||
av_get_bits_per_sample(par->codec_id) > 16;
if (waveformatextensible)
avio_wl16(pb, 0xfffe);
...
A crude attempt would be to just replace the 2 bytes at 0x20 with 01 00 and try. If that doesn't work and this behaviour is out-of-spec then file a bug report.
Related
I have a raw image that was taken with v4l2-ctl after the camera had been setup like:
# media-ctl -d /dev/media0 -l "'rzg2l_csi2 10830400.csi2':1 -> 'CRU output':0 [1]"
# media-ctl -d /dev/media0 -V "'rzg2l_csi2 10830400.csi2':1 [fmt:UYVY8_2X8/1280x960 field:none]"
# media-ctl -d /dev/media0 -V "'ov5645 0-003c':0 [fmt:UYVY8_2X8/1280x960 field:none]"
and then the picture got snapped with:
# v4l2-ctl --device /dev/video0 --stream-mmap --stream-to=frame.raw --stream-count=1
now I've tried multiple methods to convert this into a jpeg but nothing seems to yield the expected output
the raw file can be downloaded here: https://drive.google.com/file/d/1VqXnrJDYbzdtSsWfTlm2mX9rl1-Rl_7F/view?usp=sharing
I tried out the following command:
convert -verbose -size 1280x960 UYVY:frame.raw frame.bmp
which I found on Converting from YUV(UYVY) to RGB using imagemagick
but it doesn't do the trick
Your frame is 2457600 bytes and your pixel dimensions are 1280x960, so you have:
bits per pixel = 2457600 * 8 / (1280 * 960) = 16
You can get a list of the pixel formats that ffmpeg supports using:
ffmpeg -pix_fmts 2> /dev/null
Sample Output
FLAGS NAME NB_COMPONENTS BITS_PER_PIXEL
-----
IO... yuv420p 3 12
IO... yuyv422 3 16
IO... rgb24 3 24
IO... bgr24 3 24
IO... yuv422p 3 16
IO... yuv444p 3 24
IO... yuv410p 3 9
...
...
That means you can get a list of pixel formats that contain Y, U and V with 16 bits per pixel like this:
ffmpeg -pix_fmts 2> /dev/null | awk '/y/ && /u/ && /16$/ {print}'
IO... yuyv422 3 16
IO... yuv422p 3 16
IO... yuvj422p 3 16
IO... uyvy422 3 16
IO... yuv440p 3 16
IO... yuvj440p 3 16
IO... yvyu422 3 16
Now you can run a loop, iterating over all the 16-bit per pixel YUV formats and see what ffmpeg makes of your image - naming each result after the format so you can identify which is which:
ffmpeg -pix_fmts 2> /dev/null |
awk '/y/ && /u/ && /16$/ {print $2}' |
while read f; do
ffmpeg -y -s:v 1280x960 -pix_fmt $f -i frame.raw $f.jpg
done
That gives you these files:
-rw-r--r-- 1 mark staff 304916 3 Feb 09:38 yuv440p.jpg
-rw-r--r-- 1 mark staff 227123 3 Feb 09:38 yuvj422p.jpg
-rw-r--r-- 1 mark staff 39543 3 Feb 09:38 yuyv422.jpg
-rw-r--r-- 1 mark staff 39545 3 Feb 09:38 yvyu422.jpg
And I guess that yuyv422.jpg is your image, so that means you can extract it with:
ffmpeg -y -s:v 1280x960 -pix_fmt yuyv422 -i frame.raw result.jpg
If you wanted to do that with ImageMagick, you could do something like this:
#!/bin/bash
python3 <<EOF
import numpy as np
h, w = 960, 1280
# Load raw file into Numpy array
raw = np.fromfile('frame.raw', np.uint8)
raw[0::2].tofile('Y') # Starting at the 1st byte, write every 2nd byte to file "Y"
raw[1::4].tofile('U') # Starting at the 2nd byte, write every 4th byte to file "U"
raw[3::4].tofile('V') # Starting at the 3rd byte, write every 4th byte to file "V"
EOF
# Load the Y channel, then the U and V channels forcibly resizing them, then combine and go to sRGB
magick -depth 8 -size 1280x960 gray:Y \
\( -size 640x960 gray:U gray:V -resize 1280x960\! \) \
-set colorspace YUV -combine -colorspace sRGB result.jpg
If yo don't like/have Python, that part can be replaced with some basic C as follows:
#include <stdint.h>
#include <stdio.h>
// Split YUYV file called "frame.raw" into separate channels with filenames "Y", "U" and "V"
// Compile with: clang -O3 splitter.c -o splitter
int main(){
FILE *in, *Y, *U, *V;
uint8_t buffer[4];
size_t bytesRead;
// Open input file and 1 output file per channel
in = fopen("frame.raw", "rb");
Y = fopen("Y", "wb");
U = fopen("U", "wb");
V = fopen("V", "wb");
// read up to sizeof(buffer) bytes
while ((bytesRead = fread(buffer, 1, sizeof(buffer), in)) > 0)
{
fputc(buffer[0], Y);
fputc(buffer[1], U);
fputc(buffer[2], Y);
fputc(buffer[3], V);
}
}
Having had so much fun doing ffmpeg, Python, and C versions, I thought I'd try just doing it in the shell - converting bytes to lines and so I could pick alternate lines instead of alternate bytes. This works the same as the above:
#!/bin/bash
# Build JPEG image from YUYV image with packed bytes in order YUYVYUYV...
# Use "xxd" to convert bytes into lines, then extract alternate lines - which is easier than extracting bytes
H=960
W=1280
INPUT="frame.raw"
# Take top byte of every uint16 and put into "Y.pgm"
xxd -c1 -p "$INPUT" | sed -n 'p;n' | xxd -r -p | magick -size ${W}x${H} -depth 8 gray:- Y.pgm
# Take bottom byte of every 2nd uint16, starting at the 1st, resize up to full width and put into "U.pgm"
xxd -c1 -p "$INPUT" | sed -n 'n;p' | sed -n 'p;n' | xxd -r -p | magick -size $((W/2))x${H} -depth 8 gray:- -resize ${W}x${H}\! U.pgm
# Take bottom byte of every 2nd uint16, starting at the 2nd, resize up to full width and put into "V.pgm"
xxd -c1 -p "$INPUT" | sed -n 'n;p' | sed -n 'n;p' | xxd -r -p | magick -size $((W/2))x${H} -depth 8 gray:- -resize ${W}x${H}\! V.pgm
# Load the 3 channels, combine and convert to JPEG
magick {Y,U,V}.pgm -set colorspace YUV -combine -colorspace sRGB result.jpg
# Remove litter
rm {Y,U,V}.pgm
As regards colour cast removal, as I said in the comments, the " normal" way, AFAIK, is to get the average colour of the image and invert its Hue then blend that "negated cast" back with the original image to offset the original colour cast. Here is a crude attempt - if anyone knows better please ping me!
Step 1: Get average colour cast
magick result.jpg -resize 1x1\! cast.png
Step 2: Invert the cast
magick cast.png -modulate 100,100,0 correction.png
Step 3: Blend the original with the correction and brighten maybe
magick result.jpg correction.png -define compose:args=50,50 -compose blend -composite -auto-level result.jpg
Here are the original and corrected versions:
Obviously you can change the percentages for different degrees of "correction".
I have a standard mp4 (audio + video)
I am trying to merge a 1.4 second mini mp4 clip into this track, replacing the video for the length of the mini clip but merging the audios together at a specific time
Would anyone know how to do this using ffmpeg?
I've tried quite a few different filters, however can't seem to get what I want
V <------->
miniclip.mp4 A <=======>
V <-----------> ↓ + ↓ <--->
standard.mp4 A <=========================>
Example to show miniclip.mp4 (1.4 seconds long) at timestamp 5.
ffmpeg -i main.mp4 -i miniclip.mp4 -filter_complex "[0:v]drawbox=t=fill:enable='between(t,5,6.4)'[bg];[1:v]setpts=PTS+5/TB[fg];[bg][fg]overlay=x=(W-w)/2:y=(H-h)/2:eof_action=pass;[1:a]adelay=5s:all=1[a1];[0:a][a1]amix" output.mp4
drawbox covers the main video with black. Only needed if miniclip.mp4 has a smaller width or height than main.mp4. You can omit it if miniclip.mp4 width and height is ≥ than main.mp4. Alternatively you could use the scale2ref filter to make miniclip.mp4 have the same width and height as main.mp4.
setpts add a 5 second offset to miniclip.mp4 video.
overlay overlays miniclip.mp4 video over main.mp4 video.
adelay adds a 5 second delay to miniclip.mp4 audio.
amix mixes miniclip.mp4 and main.mp4 audio.
More info
See FFmpeg Filter Documentation for info on each filter.
How to get video duration
Edited (now I understood the question):
First Get 1.4 seconds of standard.mp4 and audio1.mp3
-ss is the start for get the small video that will be 1.4 seconds of length (with -t option you can specify the duration, in this case 1.4 seconds) summary: cut video from min 5, 1.4 seconds
-an is for audio none copy, because you want to add a new audio1.mp3
video_only.mp4
ffmpeg -ss 00:05:00 -i standard.mp4 -t 1.4 -map 0:v -c copy -an small_only_video.mp4
audio_only.mp4
ffmpeg -ss 00:05:00 -i audio1.mp3 -t 1.4 -c copy small_only_audio.mp3
now you can to create a small_clip_audiovideo.mp4
ffmpeg -i small_only_video.mp4 -c:a mp3 -i small_only_audio.mp3 -c copy -map 0:v -map 1:a:0 -disposition:a:0 default -disposition:a:1 default -strict -2 -sn -dn -map_metadata -1 -map_chapters -1 -movflags faststart small_clip_audiovideo.mp4
V <------->
miniclip.mp4 A <=======>
V <-----------> ↓ + ↓ <------->
standard.mp4 A <=============================>
|--|--|--|--|--|--|--|--|--|--|
0 1 2 3 4 5 6 7 8 9 10
standard.mp4 have 10 seconds (aprox) of duration, have audio and video
miniclip.mp4 have 03 seconds (aprox) of duration, have video and audio
ffmpeg -i standard.mp4 |
} have same codes of video and audio?*
ffmpeg -i miniclip.mp4 |
if are not a same audio code or video code of files standard.mp4 and miniclip.mp4, you will be to recode for continue, if you want a good work.
ffmpeg -ss 00:00:00 -i standard.mp4 -t 4 -c copy 01.part_project.mp4
and 7 to 10, in 03.part_project.mp4
ffmpeg -ss 00:00:04.000 -i standard.mp4 -t 3.0000 -c copy 03.part_project.mp4
changue name or create a copy of miniclip.mp4 to 02.part_project.mp4
cp miniclip.mp4 02.part_project.mp4
(the part of 4 second to 7 seconds, of standard.mp4 will be used if you choice the OPTION 2 copy only the audio, santadard_part2_audio.mp4)
NOW THE OPTION N 1: IS TO CONTACT (UNITED) the 3 video parts
make a folder "option1" and copy 01.part_project.mp4 02.part_project.mp4 03.part_project.mp4
mkdir option1 && cp 01.part_project.mp4 02.part_project.mp4 03.part_project.mp4 ./option1 && cd ./option1
now you concat 01.part_project.mp4 + 02.part_project.mp4 + 03.part_project.mp4 into a unique file fin_option1.mp4
ffmpeg -f concat -safe 0 -i <(for f in ./*.mp4; do echo "file '$PWD/$f'"; done) -c copy fin_option1.mp4
V <------->
miniclip.mp4 A <=======>
V <-----------> ↓ + ↓ <------->
standard.mp4 A <============XXXXXXXXX========>
|--|--|--|--|--|--|--|--|--|--|
0 1 2 3 4 5 6 7 8 9 10
THE SECOND OPTION IS TO CONTACT (UNITED) the 3 video parts, BUT MIX
THE AUDIO OF miniclip.mp4 with santadard_part2_audio.mp4
get the audio stream from santadard_part2_audio.mp4 and get the audio
file only from miniclip.mp4
ffmpeg -i santadard_part2_audio.mp4 -map 0:a -c copy -vn -strict -2 mix_audio_santadad.mp4
ffmpeg -i miniclip.mp4 -map 0:a -c copy -vn -strict -2 mix_audio_miniclip.mp4
MIX ALL AUDIOS** IN ONE AND PUT THE VIDEO FROM miniclip.mp4
ffmpeg -i mix_audio_miniclip.mp4 -i mix_audio_santadad.mp4 -filter_complex amix=inputs=2:duration=longest -strict -2 audio_mixed_miniclip.mp4
get only video from miniclip.mp4
ffmpeg -i miniclip.mp4 -c copy -an miniclip_video.mp4
and get miniclip but with mixed audios, I think that it is the solution that you are looking for
ffmpeg -i miniclip_video.mp4 -i audio_mixed_miniclip.mp4 -c copy -map 0:v -map 1:a:0 -disposition:a:0 default -disposition:a:1 default -strict -2 -sn -dn -map_metadata -1 -map_chapters -1 -movflags faststart 02.part_project_OPTION2.mp4
santadard_part2_audio.mp4
+
audio_miniclip.mp4
V <------->
miniclip.mp4 A <MMMMMMMM> (audio miniclip mixed with standard.mp4)
V <-----------> ↓ + ↓ <------->
standard.mp4 A <============ ========>
|--|--|--|--|--|--|--|--|--|--|
0 1 2 3 4 5 6 7 8 9 10
make a folder "option2" and copy 01.part_project.mp4 02.part_project_OPTION2.mp4 03.part_project.mp4
mkdir option2 && cp 01.part_project.mp4 02.part_project_OPTION2.mp4 03.part_project.mp4 ./option2 && cd ./option2
ffmpeg -f concat -safe 0 -i <(for f in ./*.mp4; do echo "file '$PWD/$f'"; done) -c copy fin_option2.mp4
NOTES
** YOU CAN USE A LOT OF AUDIO MANIPULATIONS https://trac.ffmpeg.org/wiki/AudioChannelManipulation
I have a USB audio-video capture device, something used to digitize video cassettes. I want to record both the video and audio from the device to a video file that has dimensions 720x576 and video codec H.264 and good audio quality.
I am able to record video from the device using ffmpeg and I am able to see video from the device using MPlayer. I am able also to see that audio is being delivered from the device to the computer by looking at Input tab of the Sound Preferences window or by recording the audio using Audacity, however the audio gets delivered from the device apparently only when the video is being accessed using ffmpeg or MPlayer.
I have tried to get ffmpeg to record the audio and I have tried to get MPlayer to play the audio and my efforts have not been successful.
The device is "Pinnacle Dazzle DVC 90/100/101" (as returned by v4l2-ctl --list-devices). The sound cards listing shows it as "DVC100":
$ cat /proc/asound/cards
0 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0x601d118000 irq 171
1 [DVC100 ]: USB-Audio - DVC100
Pinnacle Systems GmbH DVC100 at usb-0000:00:14.0-4, high speed
29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control
ThinkPad Console Audio Control at EC reg 0x30, fw N2LHT33W
The PulseAudio listing for the device is as follows:
$ pactl list cards short
0 alsa_card.pci-0000_00_1f.3 module-alsa-card.c
14 alsa_card.usb-Pinnacle_Systems_GmbH_DVC100-01 module-alsa-card.c
The following ffmpeg command successfully records video, but records severely distorted, broken and out-of-sync audio:
ffmpeg -y -f rawvideo -f alsa -thread_queue_size 2048 -ar 48000 -i hw:0 \
-c:a aac -video_size 720x576 -pixel_format uyvy422 -i /dev/video2 out.mp4
The following MPlayer command successfully displays the video but does not play the audio:
mplayer -tv driver=v4l2:norm=PAL:device=/dev/video2:width=720:height=576 \
-ao alsa:device=hw=1.0 -vf pp=lb tv://
Now, when the above MPlayer command is running (not the ffmpeg command) and displaying the input video in a window, Audacity can be opened and set recording audio, and it records the audio from the device clearly and in good quality. While Audacity is doing this, the input device is listed in pavucontrol as "Dazzle DVC Audio Device Analogue Stereo". Equivalently, arecord can be used also to record the audio using the following command (with output shown):
$ arecord -vv -D plughw:DVC100 -fdat out.wav
Recording WAVE 'out.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Plug PCM: Hardware PCM card 1 'DVC100' device 0 subdevice 0
Its setup is:
stream : CAPTURE
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 1
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 6755399441055744000
appl_ptr : 0
hw_ptr : 0
Looking at the output of arecord -L, I tried a variety of audio device input names with ffmpeg and none of them seemed to work. So, for example, I tried commands like the following:
ffmpeg -y -f rawvideo -f alsa -i plughw:DVC100 \
-video_size 720x576 -pixel_format uyvy422 -i /dev/video2 out.mp4
And tried the following audio device names:
plughw:DVC100
plughw:CARD=DVC100,DEV=0
hw:CARD=DVC100,DEV=0
plughw:CARD=DVC100
sysdefault:CARD=DVC100
iec958:CARD=DVC100,DEV=0
dsnoop:CARD=DVC100,DEV=0
So, how might I get ffmpeg to record the audio successfully to the video file? Is there some alternative approach to this problem?
EDIT: The relevant output from the command pactl list sources is as follows:
Source #20
State: SUSPENDED
Name: alsa_input.usb-Pinnacle_Systems_GmbH_DVC100-01.analog-stereo
Description: Dazzle DVC100 Audio Device Analogue Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 48000Hz
Channel Map: front-left,front-right
Owner Module: 45
Mute: no
Volume: front-left: 99957 / 153% / 11.00 dB, front-right: 99957 / 153% / 11.00 dB
balance 0.00
Base Volume: 35466 / 54% / -16.00 dB
Monitor of Sink: n/a
Latency: 0 usec, configured 0 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "DVC100"
alsa.long_card_name = "Pinnacle Systems GmbH DVC100 at usb-0000:00:14.0-4, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:14.0-usb-0:4:1.1"
sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-4/1-4:1.1/sound/card1"
udev.id = "usb-Pinnacle_Systems_GmbH_DVC100-01"
device.bus = "usb"
device.vendor.id = "2304"
device.vendor.name = "Pinnacle Systems, Inc."
device.product.id = "021a"
device.product.name = "Dazzle DVC100 Audio Device"
device.serial = "Pinnacle_Systems_GmbH_DVC100"
device.string = "front:1"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analogue Stereo"
device.description = "Dazzle DVC100 Audio Device Analogue Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB2304:021a"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
Ports:
analog-input-linein: Line In (priority: 8100)
Active Port: analog-input-linein
Formats:
pcm
I tested the name from this with ffmpeg (version 4.3.1, compiled with -enable-libpulse) in the following way:
ffmpeg -y -f video4linux2 -f pulse \
-i alsa_input.usb-Pinnacle_Systems_GmbH_DVC100-01.analog-stereo \
-video_size 720x576 -pixel_format uyvy422 -i /dev/video2 out.mp4
Unfortunately this hasn't worked.
I also use Dazzle DVC100 to capture video and -f alsa -i hw:1 works well to me. For instance:
ffmpeg -f alsa -i hw:1 -i /dev/video2 \
-codec:v ffv1 -codec:a pcm_s16le raw.mkv
The number of the device can be found using:
cat /proc/asound/cards
Use the number in the first column after hw: prefix. In your case it is hw:1.
Keep in mind FFmpeg fails opening the device when PulsAudio device is opened. It happens to me when I am runnning pavucontrol at the same time for example. In practice I need to wait about a half of a minute after closing pavucontrol before running FFmpeg successfully.
You can check the output of FFmpeg in real time using:
ffmpeg -f alsa -i hw:1 -i /dev/video2 \
-codec:v ffv1 -codec:a pcm_s16le -f matroska - | ffplay -
You can find more information on capturing video using Dazzle DVC100 in my post.
This is a small python3 script reading off picam using OpenCV :
#picamStream.py
import sys, os
from picamera.array import PiRGBArray
from picamera import PiCamera
import time
import cv2
# initialize the camera and grab a reference to the raw camera capture
camera = PiCamera()
camera.resolution = (960, 540)
camera.framerate = 30
rawCapture = PiRGBArray(camera, size=(960, 540))
# allow the camera to warmup
time.sleep(0.1)
# capture frames from the camera
for frame in camera.capture_continuous(rawCapture, format="bgr", use_video_port=True):
image = frame.array
# ---------------------------------
# .
# Opencv image processing goes here
# .
# ---------------------------------
os.write(1, image.tostring())
# clear the stream in preparation for the next frame
rawCapture.truncate(0)
# end
And I am trying to pipe it to ffmpeg to Youtube stream
My understanding is that I need to reference below two commands to somehow come up with a new ffmpeg command.
Piping picam live video to ffmpeg for Youtube streaming.
raspivid -o - -t 0 -vf -hf -w 960 -h 540 -fps 25 -b 1000000 | ffmpeg -re -ar 44100 -ac 2 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero -f h264 -i - -vcodec copy -acodec aac -ab 128k -g 50 -strict experimental -f flv rtmp://a.rtmp.youtube.com/live2/[STREAMKEY]
Piping OPENCV raw video to ffmpeg for mp4 file.
python3 picamStream.py | ffmpeg -f rawvideo -pixel_format bgr24 -video_size 960x540 -framerate 30 -i - foo.mp4
So far I've had no luck. Can anyone help me with this?
This is the program I use in raspberry pi.
#main.py
import subprocess
import cv2
cap = cv2.VideoCapture(0)
cap.set(cv2.CAP_PROP_FRAME_WIDTH, 640)
cap.set(cv2.CAP_PROP_FRAME_HEIGHT, 480)
command = ['ffmpeg',
'-f', 'rawvideo',
'-pix_fmt', 'bgr24',
'-s','640x480',
'-i','-',
'-ar', '44100',
'-ac', '2',
'-acodec', 'pcm_s16le',
'-f', 's16le',
'-ac', '2',
'-i','/dev/zero',
'-acodec','aac',
'-ab','128k',
'-strict','experimental',
'-vcodec','h264',
'-pix_fmt','yuv420p',
'-g', '50',
'-vb','1000k',
'-profile:v', 'baseline',
'-preset', 'ultrafast',
'-r', '30',
'-f', 'flv',
'rtmp://a.rtmp.youtube.com/live2/[STREAMKEY]']
pipe = subprocess.Popen(command, stdin=subprocess.PIPE)
while True:
_, frame = cap.read()
pipe.stdin.write(frame.tostring())
pipe.kill()
cap.release()
Youtube needs an audio source, so use -i /dev/zero.
I hope it helps you.
I have a video and an audio file. I'm trying joining them and slice a piece of video and it's working:
ffmpeg -ss 0:0:1.950 -i "video.avi" -ss 0:0:1.950 -i "audio.mp3" -target pal-dvd -bufsize 9175040 -muxrate 50400000 -acodec ac3 -ac 2 -ab 128k -ar 44100 -t 0:0:5.997 -y "output.mpg"
The problem is when I try resize the video using the -vf filter, example:
ffmpeg -ss 0:0:1.950 -i "video.avi" -ss 0:0:1.950 -i "audio.mp3" -vf scale="1024:420" -target pal-dvd -bufsize 9175040 -muxrate 50400000 -acodec ac3 -ac 2 -ab 128k -ar 44100 -t 0:0:5.997 -y "output.mpg"
It doesn't work because of the argument: -target pal-dvd. If I remove this argument, the video resize but doesn't keep the quality I want.
-target pal-dvd is equal to -c:v mpeg2video -c:a ac3 -f dvd -s 720x576 -r 25 -pix_fmt yuv420p -g 15 -b:v 6000000 -maxrate:v 9000000 -minrate:v 0 -bufsize:v 1835008 -packetsize 2048 -muxrate 10080000 -b:a 448000 -ar 48000. Your other options override these defaults, so you can simply use these options directly and remove the -s 720x576 and use your own size instead.
I'm not sure why you want to resize to 1024x420 and then use -target pal-dvd, but this option implies additional options. From ffmpeg_opt.c:
} else if (!strcmp(arg, "dvd")) {
opt_video_codec(o, "c:v", "mpeg2video");
opt_audio_codec(o, "c:a", "ac3");
parse_option(o, "f", "dvd", options);
parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options);
parse_option(o, "r", frame_rates[norm], options);
parse_option(o, "pix_fmt", "yuv420p", options);
opt_default(NULL, "g", norm == PAL ? "15" : "18");
opt_default(NULL, "b:v", "6000000");
opt_default(NULL, "maxrate:v", "9000000");
opt_default(NULL, "minrate:v", "0"); // 1500000;
opt_default(NULL, "bufsize:v", "1835008"); // 224*1024*8;
opt_default(NULL, "packetsize", "2048"); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
opt_default(NULL, "muxrate", "10080000"); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
opt_default(NULL, "b:a", "448000");
parse_option(o, "ar", "48000", options);
Also, option placement matters. If you want to resize and use -target then place the filtering after -target. Note that this will probably resize twice.
Or omit -target and manually declare each option and modify them to your desired specifications.