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Autres articles (59)
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MediaSPIP version 0.1 Beta
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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
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Sur d’autres sites (11342)
-
What's the most desireable way to capture system display and audio in the form of individual encoded audio and video packets in go (language) ? [closed]
11 janvier 2023, par Tiger YangQuestion (read the context below first) :


For those of you familiar with the capabilities of go, Is there a better way to go about all this ? Since ffmpeg is so ubiquitous, I'm sure it's been optomized to perfection, but what's the best way to capture system display and audio in the form of individual encoded audio and video packets in go (language), so that they can be then sent via webtransport-go ? I wish for it to prioritize efficiency and low latency, and ideally capture and encode the framebuffer directly like ffmpeg does.


Thanks ! I have many other questions about this, but I think it's best to ask as I go.


Context and what I've done so far :


I'm writing a remote desktop software for my personal use because of grievances with current solutions out there. At the moment, it consists of a web app that uses the webtransport API to send input datagrams and receive AV packets on two dedicated unidirectional streams, and the webcodecs API to decode these packets. On the serverside, I originally planned to use python with the aioquic library as a webtransport server. Upon connection and authentication, the server would start ffmpeg as a subprocess with this command :


ffmpeg -init_hw_device d3d11va -filter_complex ddagrab=video_size=1920x1080:framerate=60 -vcodec hevc_nvenc -tune ll -preset p7 -spatial_aq 1 -temporal_aq 1 -forced-idr 1 -rc cbr -b:v 400K -no-scenecut 1 -g 216000 -f hevc -


What I really appreciate about this is that it uses windows' desktop duplication API to copy the framebuffer of my GPU and hand that directly to the on-die hardware encoder with zero round trips to the CPU. I think it's about as efficient and elegant a solution as I can manage. It then outputs the encoded stream to the stdout, which python can read and send to the client.


As for the audio, there is another ffmpeg instance :


ffmpeg -f dshow -channels 2 -sample_rate 48000 -sample_size 16 -audio_buffer_size 15 -i audio="RD Audio (High Definition Audio Device)" -acodec libopus -vbr on -application audio -mapping_family 0 -apply_phase_inv true -b:a 25K -fec false -packet_loss 0 -map 0 -f data -


which listens to a physical loopback interface, which is literally just a short wire bridging the front panel headphone and microphone jacks (I'm aware of the quality loss of converting to analog and back, but the audio is then crushed down to 25kbps so it's fine) ()


Unfortunately, aioquic was not easy to work with IMO, and I found webtransport-go https://github.com/adriancable/webtransport-go, which was a hell of a lot better in both simplicity and documentation. However, now I'm dealing with a whole new language, and I wanna ask : (above)


EDIT : Here's the code for my server so far :




package main

import (
 "bytes"
 "context"
 "fmt"
 "log"
 "net/http"
 "os/exec"
 "time"

 "github.com/adriancable/webtransport-go"
)

func warn(str string) {
 fmt.Printf("\n===== WARNING ===================================================================================================\n %s\n=================================================================================================================\n", str)
}

func main() {

 password := []byte("abc")

 videoString := []string{
 "ffmpeg",
 "-init_hw_device", "d3d11va",
 "-filter_complex", "ddagrab=video_size=1920x1080:framerate=60",
 "-vcodec", "hevc_nvenc",
 "-tune", "ll",
 "-preset", "p7",
 "-spatial_aq", "1",
 "-temporal_aq", "1",
 "-forced-idr", "1",
 "-rc", "cbr",
 "-b:v", "500K",
 "-no-scenecut", "1",
 "-g", "216000",
 "-f", "hevc", "-",
 }

 audioString := []string{
 "ffmpeg",
 "-f", "dshow",
 "-channels", "2",
 "-sample_rate", "48000",
 "-sample_size", "16",
 "-audio_buffer_size", "15",
 "-i", "audio=RD Audio (High Definition Audio Device)",
 "-acodec", "libopus",
 "-mapping_family", "0",
 "-b:a", "25K",
 "-map", "0",
 "-f", "data", "-",
 }

 connected := false

 http.HandleFunc("/", func(writer http.ResponseWriter, request *http.Request) {
 session := request.Body.(*webtransport.Session)

 session.AcceptSession()
 fmt.Println("\nAccepted incoming WebTransport connection.")
 fmt.Println("Awaiting authentication...")

 authData, err := session.ReceiveMessage(session.Context()) // Waits here till first datagram
 if err != nil { // if client closes connection before sending anything
 fmt.Println("\nConnection closed:", err)
 return
 }

 if len(authData) >= 2 && bytes.Equal(authData[2:], password) {
 if connected {
 session.CloseSession()
 warn("Client has authenticated, but a session is already taking place! Connection closed.")
 return
 } else {
 connected = true
 fmt.Println("Client has authenticated!\n")
 }
 } else {
 session.CloseSession()
 warn("Client has failed authentication! Connection closed. (" + string(authData[2:]) + ")")
 return
 }

 videoStream, _ := session.OpenUniStreamSync(session.Context())

 videoCmd := exec.Command(videoString[0], videoString[1:]...)
 go func() {
 videoOut, _ := videoCmd.StdoutPipe()
 videoCmd.Start()

 buffer := make([]byte, 15000)
 for {
 len, err := videoOut.Read(buffer)
 if err != nil {
 break
 }
 if len > 0 {
 videoStream.Write(buffer[:len])
 }
 }
 }()

 time.Sleep(50 * time.Millisecond)

 audioStream, err := session.OpenUniStreamSync(session.Context())

 audioCmd := exec.Command(audioString[0], audioString[1:]...)
 go func() {
 audioOut, _ := audioCmd.StdoutPipe()
 audioCmd.Start()

 buffer := make([]byte, 15000)
 for {
 len, err := audioOut.Read(buffer)
 if err != nil {
 break
 }
 if len > 0 {
 audioStream.Write(buffer[:len])
 }
 }
 }()

 for {
 data, err := session.ReceiveMessage(session.Context())
 if err != nil {
 videoCmd.Process.Kill()
 audioCmd.Process.Kill()

 connected = false

 fmt.Println("\nConnection closed:", err)
 break
 }

 if len(data) == 0 {

 } else if data[0] == byte(0) {
 fmt.Printf("Received mouse datagram: %s\n", data)
 }
 }

 })

 server := &webtransport.Server{
 ListenAddr: ":1024",
 TLSCert: webtransport.CertFile{Path: "SSL/fullchain.pem"},
 TLSKey: webtransport.CertFile{Path: "SSL/privkey.pem"},
 QuicConfig: &webtransport.QuicConfig{
 KeepAlive: false,
 MaxIdleTimeout: 3 * time.Second,
 },
 }

 fmt.Println("Launching WebTransport server at", server.ListenAddr)
 ctx, cancel := context.WithCancel(context.Background())
 if err := server.Run(ctx); err != nil {
 log.Fatal(err)
 cancel()
 }

}







-
Libavformat/FFMPEG : Muxing into mp4 with AVFormatContext drops the final frame, depending on the number of frames
27 octobre 2020, par Galen LynchI am trying to use libavformat to create a
.mp4
video
with a single h.264 video stream, but the final frame in the resulting file
often has a duration of zero and is effectively dropped from the video.
Strangely enough, whether the final frame is dropped or not depends on how many
frames I try to add to the file. Some simple testing that I outline below makes
me think that I am somehow misconfiguring either theAVFormatContext
or the
h.264 encoder, resulting in two edit lists that sometimes chop off the final
frame. I will also post a simplified version of the code I am using, in case I'm
making some obvious mistake. Any help would be greatly appreciated : I've been
struggling with this issue for the past few days and have made little progress.

I can recover the dropped frame by creating a new mp4 container using
ffmpeg

binary with the copy codec if I use the-ignore_editlist
option. Inspecting
the file with a missing frame usingffprobe
,mp4trackdump
, ormp4file --dump
, shows that the final frame is dropped if its sample time is exactly the
same the end of the edit list. When I make a file that has no dropped frames, it
still has two edit lists : the only difference is that the end time of the edit
list is beyond all samples in files that do not have dropped frames. Though this
is hardly a fair comparison, if I make a.png
for each frame and then generate
a.mp4
withffmpeg
using theimage2
codec and similar h.264 settings, I
produce a movie with all frames present, only one edit list, and similar PTS
times as my mangled movies with two edit lists. In this case, the edit list
always ends after the last frame/sample time.

I am using this command to determine the number of frames in the resulting stream,
though I also get the same number with other utilities :


ffprobe -v error -count_frames -select_streams v:0 -show_entries stream=nb_read_frames -of default=nokey=1:noprint_wrappers=1 video_file_name.mp4



Simple inspection of the file with ffprobe shows no obviously alarming signs to
me, besides the framerate being affected by the missing frame (the target was
24) :


$ ffprobe -hide_banner testing.mp4
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testing.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.45.100
 Duration: 00:00:04.13, start: 0.041016, bitrate: 724 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 100x100, 722 kb/s, 24.24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
 Metadata:
 handler_name : VideoHandler



The files that I generate programatically always have two edit lists, one of
which is very short. In files both with and without a missing frame, the
duration one of the frames is 0, while all the others have the same duration
(512). You can see this in the
ffmpeg
output for this file that I tried to put
100 frames into, though only 99 are visible despite the file containing all 100
samples.

$ ffmpeg -hide_banner -y -v 9 -loglevel 99 -i testing.mp4 
...
<edited to="to" remove="remove" the="the" class="class" printing="printing">
type:'edts' parent:'trak' sz: 48 100 948
type:'elst' parent:'edts' sz: 40 8 40
track[0].edit_count = 2
duration=41 time=-1 rate=1.000000
duration=4125 time=0 rate=1.000000
type:'mdia' parent:'trak' sz: 808 148 948
type:'mdhd' parent:'mdia' sz: 32 8 800
type:'hdlr' parent:'mdia' sz: 45 40 800
ctype=[0][0][0][0]
stype=vide
type:'minf' parent:'mdia' sz: 723 85 800
type:'vmhd' parent:'minf' sz: 20 8 715
type:'dinf' parent:'minf' sz: 36 28 715
type:'dref' parent:'dinf' sz: 28 8 28
Unknown dref type 0x206c7275 size 12
type:'stbl' parent:'minf' sz: 659 64 715
type:'stsd' parent:'stbl' sz: 151 8 651
size=135 4CC=avc1 codec_type=0
type:'avcC' parent:'stsd' sz: 49 8 49
type:'stts' parent:'stbl' sz: 32 159 651
track[0].stts.entries = 2
sample_count=99, sample_duration=512
sample_count=1, sample_duration=0
...
AVIndex stream 0, sample 99, offset 5a0ed, dts 50688, size 3707, distance 0, keyframe 1
Processing st: 0, edit list 0 - media time: -1, duration: 504
Processing st: 0, edit list 1 - media time: 0, duration: 50688
type:'udta' parent:'moov' sz: 98 1072 1162
...
</edited>


The last frame has zero duration :


$ mp4trackdump -v testing.mp4
...
mp4file testing.mp4, track 1, samples 100, timescale 12288
sampleId 1, size 6943 duration 512 time 0 00:00:00.000 S
sampleId 2, size 3671 duration 512 time 512 00:00:00.041 S
...
sampleId 99, size 3687 duration 512 time 50176 00:00:04.083 S
sampleId 100, size 3707 duration 0 time 50688 00:00:04.125 S



Non-mangled videos that I generate have similar structure, as you can see in
this video that had 99 input frames, all of which are visible in the output.
Even though the sample_duration is set to zero for one of the samples in the
stss box, it is not dropped from the frame count or when reading the frames back
in with ffmpeg.


$ ffmpeg -hide_banner -y -v 9 -loglevel 99 -i testing_99.mp4 
...
type:'elst' parent:'edts' sz: 40 8 40
track[0].edit_count = 2
duration=41 time=-1 rate=1.000000
duration=4084 time=0 rate=1.000000
...
track[0].stts.entries = 2
sample_count=98, sample_duration=512
sample_count=1, sample_duration=0
...
AVIndex stream 0, sample 98, offset 5d599, dts 50176, size 3833, distance 0, keyframe 1
Processing st: 0, edit list 0 - media time: -1, duration: 504
Processing st: 0, edit list 1 - media time: 0, duration: 50184
...



$ mp4trackdump -v testing_99.mp4
...
sampleId 98, size 3814 duration 512 time 49664 00:00:04.041 S
sampleId 99, size 3833 duration 0 time 50176 00:00:04.083 S



One difference that jumps out to me is that the mangled file's second edit list
ends at time 50688, which coincides with the last sample, while the non-mangled
file's edit list ends at 50184, which is after the time of the last sample
at 50176. As I mentioned before, whether the last frame is clipped depends on
the number of frames I encode and mux into the container : 100 input frames
results in 1 dropped frame, 99 results in 0, 98 in 0, 97 in 1, etc...


Here is the code that I used to generate these files, which is a MWE script
version of library functions that I am modifying. It is written in Julia,
which I do not think is important here, and calls the FFMPEG library version
4.3.1. It's more or less a direct translation from of the FFMPEG muxing
demo, although the codec
context here is created before the format context. I am presenting the code that
interacts with ffmpeg first, although it relies on some helper code that I will
put below.


The helper code just makes it easier to work with nested C structs in Julia, and
allows
.
syntax in Julia to be used in place of C's arrow (->
) operator for
field access of struct pointers. Libav structs such asAVFrame
appear as a
thin wrapper typeAVFramePtr
, and similarlyAVStream
appears as
AVStreamPtr
etc... These act like single or double pointers for the purposes
of function calls, depending on the function's type signature. Hopefully it will
be clear enough to understand if you are familiar with working with libav in C,
and I don't think looking at the helper code should be necessary if you don't
want to run the code.

# Function to transfer array to AVPicture/AVFrame
function transfer_img_buf_to_frame!(frame, img)
 img_pointer = pointer(img)
 data_pointer = frame.data[1] # Base-1 indexing, get pointer to first data buffer in frame
 for h = 1:frame.height
 data_line_pointer = data_pointer + (h-1) * frame.linesize[1] # base-1 indexing
 img_line_pointer = img_pointer + (h-1) * frame.width
 unsafe_copyto!(data_line_pointer, img_line_pointer, frame.width) # base-1 indexing
 end
end

# Function to transfer AVFrame to AVCodecContext, and AVPacket to AVFormatContext
function encode_mux!(packet, format_context, frame, codec_context; flush = false)
 if flush
 fret = avcodec_send_frame(codec_context, C_NULL)
 else
 fret = avcodec_send_frame(codec_context, frame)
 end
 if fret < 0 && !in(fret, [-Libc.EAGAIN, VIO_AVERROR_EOF])
 error("Error $fret sending a frame for encoding")
 end

 pret = Cint(0)
 while pret >= 0
 pret = avcodec_receive_packet(codec_context, packet)
 if pret == -Libc.EAGAIN || pret == VIO_AVERROR_EOF
 break
 elseif pret < 0
 error("Error $pret during encoding")
 end
 stream = format_context.streams[1] # Base-1 indexing
 av_packet_rescale_ts(packet, codec_context.time_base, stream.time_base)
 packet.stream_index = 0
 ret = av_interleaved_write_frame(format_context, packet)
 ret < 0 && error("Error muxing packet: $ret")
 end
 if !flush && fret == -Libc.EAGAIN && pret != VIO_AVERROR_EOF
 fret = avcodec_send_frame(codec_context, frame)
 if fret < 0 && fret != VIO_AVERROR_EOF
 error("Error $fret sending a frame for encoding")
 end
 end
 return pret
end

# Set parameters of test movie
nframe = 100
width, height = 100, 100
framerate = 24
gop = 0
codec_name = "libx264"
filename = "testing.mp4"

((width % 2 !=0) || (height % 2 !=0)) && error("Encoding error: Image dims must be a multiple of two")

# Make test images
imgstack = map(x->rand(UInt8,width,height),1:nframe);

pix_fmt = AV_PIX_FMT_GRAY8
framerate_rat = Rational(framerate)

codec = avcodec_find_encoder_by_name(codec_name)
codec == C_NULL && error("Codec '$codec_name' not found")

# Allocate AVCodecContext
codec_context_p = avcodec_alloc_context3(codec) # raw pointer
codec_context_p == C_NULL && error("Could not allocate AVCodecContext")
# Easier to work with pointer that acts like a c struct pointer, type defined below
codec_context = AVCodecContextPtr(codec_context_p)

codec_context.width = width
codec_context.height = height
codec_context.time_base = AVRational(1/framerate_rat)
codec_context.framerate = AVRational(framerate_rat)
codec_context.pix_fmt = pix_fmt
codec_context.gop_size = gop

ret = avcodec_open2(codec_context, codec, C_NULL)
ret < 0 && error("Could not open codec: Return code $(ret)")

# Allocate AVFrame and wrap it in a Julia convenience type
frame_p = av_frame_alloc()
frame_p == C_NULL && error("Could not allocate AVFrame")
frame = AVFramePtr(frame_p)

frame.format = pix_fmt
frame.width = width
frame.height = height

# Allocate picture buffers for frame
ret = av_frame_get_buffer(frame, 0)
ret < 0 && error("Could not allocate the video frame data")

# Allocate AVPacket and wrap it in a Julia convenience type
packet_p = av_packet_alloc()
packet_p == C_NULL && error("Could not allocate AVPacket")
packet = AVPacketPtr(packet_p)

# Allocate AVFormatContext and wrap it in a Julia convenience type
format_context_dp = Ref(Ptr{AVFormatContext}()) # double pointer
ret = avformat_alloc_output_context2(format_context_dp, C_NULL, C_NULL, filename)
if ret != 0 || format_context_dp[] == C_NULL
 error("Could not allocate AVFormatContext")
end
format_context = AVFormatContextPtr(format_context_dp)

# Add video stream to AVFormatContext and configure it to use the encoder made above
stream_p = avformat_new_stream(format_context, C_NULL)
stream_p == C_NULL && error("Could not allocate output stream")
stream = AVStreamPtr(stream_p) # Wrap this pointer in a convenience type

stream.time_base = codec_context.time_base
stream.avg_frame_rate = 1 / convert(Rational, stream.time_base)
ret = avcodec_parameters_from_context(stream.codecpar, codec_context)
ret < 0 && error("Could not set parameters of stream")

# Open the AVIOContext
pb_ptr = field_ptr(format_context, :pb)
# This following is just a call to avio_open, with a bit of extra protection
# so the Julia garbage collector does not destroy format_context during the call
ret = GC.@preserve format_context avio_open(pb_ptr, filename, AVIO_FLAG_WRITE)
ret < 0 && error("Could not open file $filename for writing")

# Write the header
ret = avformat_write_header(format_context, C_NULL)
ret < 0 && error("Could not write header")

# Encode and mux each frame
for i in 1:nframe # iterate from 1 to nframe
 img = imgstack[i] # base-1 indexing
 ret = av_frame_make_writable(frame)
 ret < 0 && error("Could not make frame writable")
 transfer_img_buf_to_frame!(frame, img)
 frame.pts = i
 encode_mux!(packet, format_context, frame, codec_context)
end

# Flush the encoder
encode_mux!(packet, format_context, frame, codec_context; flush = true)

# Write the trailer
av_write_trailer(format_context)

# Close the AVIOContext
pb_ptr = field_ptr(format_context, :pb) # get pointer to format_context.pb
ret = GC.@preserve format_context avio_closep(pb_ptr) # simply a call to avio_closep
ret < 0 && error("Could not free AVIOContext")

# Deallocation
avcodec_free_context(codec_context)
av_frame_free(frame)
av_packet_free(packet)
avformat_free_context(format_context)



Below is the helper code that makes accessing pointers to nested c structs not a
total pain in Julia. If you try to run the code yourself, please enter this in
before the logic of the code shown above. It requires
VideoIO.jl, a Julia wrapper to libav.


# Convenience type and methods to make the above code look more like C
using Base: RefValue, fieldindex

import Base: unsafe_convert, getproperty, setproperty!, getindex, setindex!,
 unsafe_wrap, propertynames

# VideoIO is a Julia wrapper to libav
#
# Bring bindings to libav library functions into namespace
using VideoIO: AVCodecContext, AVFrame, AVPacket, AVFormatContext, AVRational,
 AVStream, AV_PIX_FMT_GRAY8, AVIO_FLAG_WRITE, AVFMT_NOFILE,
 avformat_alloc_output_context2, avformat_free_context, avformat_new_stream,
 av_dump_format, avio_open, avformat_write_header,
 avcodec_parameters_from_context, av_frame_make_writable, avcodec_send_frame,
 avcodec_receive_packet, av_packet_rescale_ts, av_interleaved_write_frame,
 avformat_query_codec, avcodec_find_encoder_by_name, avcodec_alloc_context3,
 avcodec_open2, av_frame_alloc, av_frame_get_buffer, av_packet_alloc,
 avio_closep, av_write_trailer, avcodec_free_context, av_frame_free,
 av_packet_free

# Submodule of VideoIO
using VideoIO: AVCodecs

# Need to import this function from Julia's Base to add more methods
import Base: convert

const VIO_AVERROR_EOF = -541478725 # AVERROR_EOF

# Methods to convert between AVRational and Julia's Rational type, because it's
# hard to access the AV rational macros with Julia's C interface
convert(::Type{Rational{T}}, r::AVRational) where T = Rational{T}(r.num, r.den)
convert(::Type{Rational}, r::AVRational) = Rational(r.num, r.den)
convert(::Type{AVRational}, r::Rational) = AVRational(numerator(r), denominator(r))

"""
 mutable struct NestedCStruct{T}

Wraps a pointer to a C struct, and acts like a double pointer to that memory.
The methods below will automatically convert it to a single pointer if needed
for a function call, and make interacting with it in Julia look (more) similar
to interacting with it in C, except '->' in C is replaced by '.' in Julia.
"""
mutable struct NestedCStruct{T}
 data::RefValue{Ptr{T}}
end
NestedCStruct{T}(a::Ptr) where T = NestedCStruct{T}(Ref(a))
NestedCStruct(a::Ptr{T}) where T = NestedCStruct{T}(a)

const AVCodecContextPtr = NestedCStruct{AVCodecContext}
const AVFramePtr = NestedCStruct{AVFrame}
const AVPacketPtr = NestedCStruct{AVPacket}
const AVFormatContextPtr = NestedCStruct{AVFormatContext}
const AVStreamPtr = NestedCStruct{AVStream}

function field_ptr(::Type{S}, struct_pointer::Ptr{T}, field::Symbol,
 index::Integer = 1) where {S,T}
 fieldpos = fieldindex(T, field)
 field_pointer = convert(Ptr{S}, struct_pointer) +
 fieldoffset(T, fieldpos) + (index - 1) * sizeof(S)
 return field_pointer
end

field_ptr(a::Ptr{T}, field::Symbol, args...) where T =
 field_ptr(fieldtype(T, field), a, field, args...)

function check_ptr_valid(p::Ptr, err::Bool = true)
 valid = p != C_NULL
 err && !valid && error("Invalid pointer")
 valid
end

unsafe_convert(::Type{Ptr{T}}, ap::NestedCStruct{T}) where T =
 getfield(ap, :data)[]
unsafe_convert(::Type{Ptr{Ptr{T}}}, ap::NestedCStruct{T}) where T =
 unsafe_convert(Ptr{Ptr{T}}, getfield(ap, :data))

function check_ptr_valid(a::NestedCStruct{T}, args...) where T
 p = unsafe_convert(Ptr{T}, a)
 GC.@preserve a check_ptr_valid(p, args...)
end

nested_wrap(x::Ptr{T}) where T = NestedCStruct(x)
nested_wrap(x) = x

function getproperty(ap::NestedCStruct{T}, s::Symbol) where T
 check_ptr_valid(ap)
 p = unsafe_convert(Ptr{T}, ap)
 res = GC.@preserve ap unsafe_load(field_ptr(p, s))
 nested_wrap(res)
end

function setproperty!(ap::NestedCStruct{T}, s::Symbol, x) where T
 check_ptr_valid(ap)
 p = unsafe_convert(Ptr{T}, ap)
 fp = field_ptr(p, s)
 GC.@preserve ap unsafe_store!(fp, x)
end

function getindex(ap::NestedCStruct{T}, i::Integer) where T
 check_ptr_valid(ap)
 p = unsafe_convert(Ptr{T}, ap)
 res = GC.@preserve ap unsafe_load(p, i)
 nested_wrap(res)
end

function setindex!(ap::NestedCStruct{T}, i::Integer, x) where T
 check_ptr_valid(ap)
 p = unsafe_convert(Ptr{T}, ap)
 GC.@preserve ap unsafe_store!(p, x, i)
end

function unsafe_wrap(::Type{T}, ap::NestedCStruct{S}, i) where {S, T}
 check_ptr_valid(ap)
 p = unsafe_convert(Ptr{S}, ap)
 GC.@preserve ap unsafe_wrap(T, p, i)
end

function field_ptr(::Type{S}, a::NestedCStruct{T}, field::Symbol,
 args...) where {S, T}
 check_ptr_valid(a)
 p = unsafe_convert(Ptr{T}, a)
 GC.@preserve a field_ptr(S, p, field, args...)
end

field_ptr(a::NestedCStruct{T}, field::Symbol, args...) where T =
 field_ptr(fieldtype(T, field), a, field, args...)

propertynames(ap::T) where {S, T<:NestedCStruct{S}} = (fieldnames(S)...,
 fieldnames(T)...)




Edit : Some things that I have already tried


- 

- Explicitly setting the stream duration to be the same number as the number of frames that I add, or a few more beyond that
- Explicitly setting the stream start time to zero, while the first frame has a PTS of 1
- Playing around with encoder parameters, as well as
gop_size
, using B frames, etc. - Setting the private data for the mov/mp4 muxer to set the movflag
negative_cts_offsets
- Changing the framerate
- Tried different pixel formats, such as AV_PIX_FMT_YUV420P














Also to be clear while I can just transfer the file into another while ignoring the edit lists to work around this problem, I am hoping to not make damaged mp4 files in the first place.


-
Final Rendered Video is Sped Up Compared to Animation Played in Processing
15 janvier 2016, par NightlifeRecently I made an Audio Visualizer in Processing. From there I wanted to render the animation created in Processing into a mp4 file. I am on a windows computer, and am using ffmpeg to convert my TIF files produced in Processing into mp4.
When I do this I am able to render the images into an mp4 file, but when I playback this file the animation is sped up compared to the animation when I play it on Processing. Because of this the animation does not sync with the audio when I combine the mp4 file and audio on a video editing program.
When I set my frame rate to 25 and have the limit on the number of frames to be 250 and render it into a mp4 file it is 10 seconds long like it should be, but it contains more than 10 seconds of the animation when compared to the animation played directly in Processing.
I have no idea why this is so any help will be much appreciated.
My Processing code :
import ddf.minim.*;
import ddf.minim.analysis.*;
Minim minim;
AudioPlayer player;
PImage img;
FFT fft;
void setup() {
size(728, 546);
minim = new Minim(this);
// this loads mysong.wav from the data folder as a stream with a internal buffer of size 1024
player = minim.loadFile("new_years_good.mp3");
fft = new FFT(player.bufferSize(), player.sampleRate());
player.play();
img= loadImage("cat-in-shades-.jpg");
frameRate(25);
}
void draw() {
image(img, 0, 0);
//tint(0, 100, 150);
stroke(255);
strokeWeight(4);
float a = 0;
float angle = (2*PI) / 200;
fft.forward(player.mix);
for(int i=0; i < player.bufferSize() - 1; i++) {
//player.mix.get(i) is a value between [-1,1]
float x = 250 + cos(a) * (20 * player.mix.get(i) + 100);
float x2 = 540 + cos(a) * (20 * player.mix.get(i) + 100);
float y = 230 + sin(a) * (20 * player.mix.get(i) + 100);
float y2 = 240 + sin(a) * (20 * player.mix.get(i) + 100);
float xFinal = 250 + cos(a+angle) * (20 * player.mix.get(i+1) + 100);
float x2Final = 540 + cos(a+angle) * (20 * player.mix.get(i+1) + 100);
float yFinal = 230 + sin(a+angle) * (20 * player.mix.get(i+1) + 100);
float y2Final = 240 + sin(a+angle) * (20 * player.mix.get(i+1) + 100);
line(x,y,xFinal,yFinal);
line(x2,y2,x2Final,y2Final);
a += angle;
}
noStroke();
fill(255, 0, 0, 128);
for(int i = 0; i < 250; i++)
{
float b = fft.getBand(i);
float yAxis = random(-b, b) + 480;
float xAxis = i*3;
ellipse(xAxis, yAxis, b, b);
}
saveFrame("frame-####.tif");
if(frameCount>250)
{
noLoop();
stop();
}
}
void stop() {
player.close();
minim.stop();
super.stop();
}What I input into the command line (as one line) on the cmd :
C:\Users\Robert\Documents\Processing\AudioVisulizer>ffmpeg -i C:\Users\Robert\Do
cuments\Processing\AudioVisulizer\frame-%04d.tif -r 25 -pix_fmt yuv420p smallVid
.mp4What it outputted :
ffmpeg version N-77836-g62dfe1d Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.2.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl
e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --
enable-lzma --enable-decklink --enable-zlib
libavutil 55. 13.100 / 55. 13.100
libavcodec 57. 22.100 / 57. 22.100
libavformat 57. 21.101 / 57. 21.101
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, image2, from 'C:\Users\Robert\Documents\Processing\AudioVisulizer\fram
e-%04d.tif':
Duration: 00:00:10.04, start: 0.000000, bitrate: N/A
Stream #0:0: Video: tiff, rgb24, 728x546, 25 fps, 25 tbr, 25 tbn, 25 tbc
[libx264 @ 00000092cd5e3700] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 00000092cd5e3700] profile High, level 3.0
[libx264 @ 00000092cd5e3700] 264 - core 148 r2638 7599210 - H.264/MPEG-4 AVC cod
ec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 r
ef=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed
_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pski
p=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 deci
mate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_
adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=2
5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.6
0 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'smallVid.mp4':
Metadata:
encoder : Lavf57.21.101
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 728x54
6, q=-1--1, 25 fps, 12800 tbn, 25 tbc
Metadata:
encoder : Lavc57.22.100 libx264
Side data:
unknown side data type 10 (24 bytes)
Stream mapping:
Stream #0:0 -> #0:0 (tiff (native) -> h264 (libx264))
Press [q] to stop, [?] for help
frame= 52 fps=0.0 q=28.0 size= 77kB time=00:00:00.00 bitrate=N/A speed=
frame= 74 fps= 65 q=28.0 size= 127kB time=00:00:00.88 bitrate=1178.0kbits/
frame= 93 fps= 57 q=28.0 size= 164kB time=00:00:01.64 bitrate= 820.0kbits/
frame= 113 fps= 52 q=28.0 size= 201kB time=00:00:02.44 bitrate= 676.3kbits/
frame= 136 fps= 51 q=28.0 size= 245kB time=00:00:03.36 bitrate= 596.3kbits/
frame= 157 fps= 49 q=28.0 size= 282kB time=00:00:04.20 bitrate= 550.2kbits/
frame= 178 fps= 48 q=28.0 size= 324kB time=00:00:05.04 bitrate= 527.2kbits/
frame= 199 fps= 47 q=28.0 size= 362kB time=00:00:05.88 bitrate= 504.1kbits/
frame= 219 fps= 46 q=28.0 size= 403kB time=00:00:06.68 bitrate= 494.2kbits/
frame= 242 fps= 46 q=28.0 size= 452kB time=00:00:07.60 bitrate= 486.8kbits/
frame= 251 fps= 38 q=-1.0 Lsize= 623kB time=00:00:09.96 bitrate= 512.3kbits
/s speed=1.52x
video:619kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.607807%
[libx264 @ 00000092cd5e3700] frame I:2 Avg QP:21.74 size: 56596
[libx264 @ 00000092cd5e3700] frame P:66 Avg QP:23.36 size: 2523
[libx264 @ 00000092cd5e3700] frame B:183 Avg QP:31.50 size: 1932
[libx264 @ 00000092cd5e3700] consecutive B-frames: 0.8% 4.0% 6.0% 89.2%
[libx264 @ 00000092cd5e3700] mb I I16..4: 10.9% 72.6% 16.5%
[libx264 @ 00000092cd5e3700] mb P I16..4: 0.0% 0.0% 0.2% P16..4: 4.4% 2.3
% 3.3% 0.0% 0.0% skip:89.7%
[libx264 @ 00000092cd5e3700] mb B I16..4: 0.0% 0.0% 0.5% B16..8: 3.2% 2.0
% 2.3% direct: 1.3% skip:90.7% L0:50.9% L1:42.2% BI: 7.0%
[libx264 @ 00000092cd5e3700] 8x8 transform intra:50.2% inter:10.1%
[libx264 @ 00000092cd5e3700] coded y,uvDC,uvAC intra: 83.4% 32.6% 14.2% inter: 3
.5% 0.2% 0.0%
[libx264 @ 00000092cd5e3700] i16 v,h,dc,p: 22% 8% 8% 62%
[libx264 @ 00000092cd5e3700] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 12% 19% 6% 9%
9% 9% 9% 9%
[libx264 @ 00000092cd5e3700] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 14% 22% 6% 9%
8% 8% 5% 6%
[libx264 @ 00000092cd5e3700] i8c dc,h,v,p: 74% 12% 12% 2%
[libx264 @ 00000092cd5e3700] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 00000092cd5e3700] ref P L0: 41.8% 3.9% 24.3% 30.0%
[libx264 @ 00000092cd5e3700] ref B L0: 64.5% 25.0% 10.5%
[libx264 @ 00000092cd5e3700] ref B L1: 82.2% 17.8%
[libx264 @ 00000092cd5e3700] kb/s:504.56