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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
.mp4video
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 theAVFormatContextor 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_editlistoption. 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.pngfor each frame and then generate
a.mp4withffmpegusing theimage2codec 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
ffmpegoutput 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 asAVFrameappear as a
thin wrapper typeAVFramePtr, and similarlyAVStreamappears as
AVStreamPtretc... 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.


-
How to play and seek fragmented MP4 audio using MSE SourceBuffer ?
29 juin 2024, par Stefan FalkNote :




If you end up here, you might want to take a look at shaka-player and the accompanying shaka-streamer. Use it. Don't implement this yourself unless you really have to.




I am trying for quite some time now to be able to play an audio track on Chrome, Firefox, Safari, etc. but I keep hitting brick walls. My problem is currently that I am just not able to seek within a fragmented MP4 (or MP3).


At the moment I am converting audio files such as MP3 to fragmented MP4 (fMP4) and send them chunk-wise to the client. What I do is defining a
CHUNK_DURACTION_SEC(chunk duration in seconds) and compute a chunk size like this :

chunksTotal = Math.ceil(this.track.duration / CHUNK_DURATION_SEC);
chunkSize = Math.ceil(this.track.fileSize / this.chunksTotal);


With this I partition the audio file and can fetch it entirely jumping
chunkSize-many bytes for each chunk :

-----------------------------------------
| chunk 1 | chunk 2 | ... | chunk n |
-----------------------------------------


How audio files are converted to fMP4


ffmpeg -i input.mp3 -acodec aac -b:a 256k -f mp4 \
 -movflags faststart+frag_every_frame+empty_moov+default_base_moof \
 output.mp4


This seems to work with Chrome and Firefox (so far).


How chunks are appended


After following this example, and realizing that it's simply not working as it is explained here, I threw it away and started over from scratch. Unfortunately without success. It's still not working.


The following code is supposed to play a track from the very beginning to the very end. However, I also need to be able to seek. So far, this is simply not working. Seeking will just stop the audio after the
seekingevent got triggered.

The code


/* Desired chunk duration in seconds. */
const CHUNK_DURATION_SEC = 20;

const AUDIO_EVENTS = [
 'ended',
 'error',
 'play',
 'playing',
 'seeking',
 'seeked',
 'pause',
 'timeupdate',
 'canplay',
 'loadedmetadata',
 'loadstart',
 'updateend',
];


class ChunksLoader {

 /** The total number of chunks for the track. */
 public readonly chunksTotal: number;

 /** The length of one chunk in bytes */
 public readonly chunkSize: number;

 /** Keeps track of requested chunks. */
 private readonly requested: boolean[];

 /** URL of endpoint for fetching audio chunks. */
 private readonly url: string;

 constructor(
 private track: Track,
 private sourceBuffer: SourceBuffer,
 private logger: NGXLogger,
 ) {

 this.chunksTotal = Math.ceil(this.track.duration / CHUNK_DURATION_SEC);
 this.chunkSize = Math.ceil(this.track.fileSize / this.chunksTotal);

 this.requested = [];
 for (let i = 0; i < this.chunksTotal; i++) {
 this.requested[i] = false;
 }

 this.url = `${environment.apiBaseUrl}/api/tracks/${this.track.id}/play`;
 }

 /**
 * Fetch the first chunk.
 */
 public begin() {
 this.maybeFetchChunk(0);
 }

 /**
 * Handler for the "timeupdate" event. Checks if the next chunk should be fetched.
 *
 * @param currentTime
 * The current time of the track which is currently played.
 */
 public handleOnTimeUpdate(currentTime: number) {

 const nextChunkIndex = Math.floor(currentTime / CHUNK_DURATION_SEC) + 1;
 const hasAllChunks = this.requested.every(val => !!val);

 if (nextChunkIndex === (this.chunksTotal - 1) && hasAllChunks) {
 this.logger.debug('Last chunk. Calling mediaSource.endOfStream();');
 return;
 }

 if (this.requested[nextChunkIndex] === true) {
 return;
 }

 if (currentTime < CHUNK_DURATION_SEC * (nextChunkIndex - 1 + 0.25)) {
 return;
 }

 this.maybeFetchChunk(nextChunkIndex);
 }

 /**
 * Fetches the chunk if it hasn't been requested yet. After the request finished, the returned
 * chunk gets appended to the SourceBuffer-instance.
 *
 * @param chunkIndex
 * The chunk to fetch.
 */
 private maybeFetchChunk(chunkIndex: number) {

 const start = chunkIndex * this.chunkSize;
 const end = start + this.chunkSize - 1;

 if (this.requested[chunkIndex] == true) {
 return;
 }

 this.requested[chunkIndex] = true;

 if ((end - start) == 0) {
 this.logger.warn('Nothing to fetch.');
 return;
 }

 const totalKb = ((end - start) / 1000).toFixed(2);
 this.logger.debug(`Starting to fetch bytes ${start} to ${end} (total ${totalKb} kB). Chunk ${chunkIndex + 1} of ${this.chunksTotal}`);

 const xhr = new XMLHttpRequest();
 xhr.open('get', this.url);
 xhr.setRequestHeader('Authorization', `Bearer ${AuthenticationService.getJwtToken()}`);
 xhr.setRequestHeader('Range', 'bytes=' + start + '-' + end);
 xhr.responseType = 'arraybuffer';
 xhr.onload = () => {
 this.logger.debug(`Range ${start} to ${end} fetched`);
 this.logger.debug(`Requested size: ${end - start + 1}`);
 this.logger.debug(`Fetched size: ${xhr.response.byteLength}`);
 this.logger.debug('Appending chunk to SourceBuffer.');
 this.sourceBuffer.appendBuffer(xhr.response);
 };
 xhr.send();
 };

}

export enum StreamStatus {
 NOT_INITIALIZED,
 INITIALIZING,
 PLAYING,
 SEEKING,
 PAUSED,
 STOPPED,
 ERROR
}

export class PlayerState {
 status: StreamStatus = StreamStatus.NOT_INITIALIZED;
}


/**
 *
 */
@Injectable({
 providedIn: 'root'
})
export class MediaSourcePlayerService {

 public track: Track;

 private mediaSource: MediaSource;

 private sourceBuffer: SourceBuffer;

 private audioObj: HTMLAudioElement;

 private chunksLoader: ChunksLoader;

 private state: PlayerState = new PlayerState();

 private state$ = new BehaviorSubject<playerstate>(this.state);

 public stateChange = this.state$.asObservable();

 private currentTime$ = new BehaviorSubject<number>(null);

 public currentTimeChange = this.currentTime$.asObservable();

 constructor(
 private httpClient: HttpClient,
 private logger: NGXLogger
 ) {
 }

 get canPlay() {
 const state = this.state$.getValue();
 const status = state.status;
 return status == StreamStatus.PAUSED;
 }

 get canPause() {
 const state = this.state$.getValue();
 const status = state.status;
 return status == StreamStatus.PLAYING || status == StreamStatus.SEEKING;
 }

 public playTrack(track: Track) {
 this.logger.debug('playTrack');
 this.track = track;
 this.startPlayingFrom(0);
 }

 public play() {
 this.logger.debug('play()');
 this.audioObj.play().then();
 }

 public pause() {
 this.logger.debug('pause()');
 this.audioObj.pause();
 }

 public stop() {
 this.logger.debug('stop()');
 this.audioObj.pause();
 }

 public seek(seconds: number) {
 this.logger.debug('seek()');
 this.audioObj.currentTime = seconds;
 }

 private startPlayingFrom(seconds: number) {
 this.logger.info(`Start playing from ${seconds.toFixed(2)} seconds`);
 this.mediaSource = new MediaSource();
 this.mediaSource.addEventListener('sourceopen', this.onSourceOpen);

 this.audioObj = document.createElement('audio');
 this.addEvents(this.audioObj, AUDIO_EVENTS, this.handleEvent);
 this.audioObj.src = URL.createObjectURL(this.mediaSource);

 this.audioObj.play().then();
 }

 private onSourceOpen = () => {

 this.logger.debug('onSourceOpen');

 this.mediaSource.removeEventListener('sourceopen', this.onSourceOpen);
 this.mediaSource.duration = this.track.duration;

 this.sourceBuffer = this.mediaSource.addSourceBuffer('audio/mp4; codecs="mp4a.40.2"');
 // this.sourceBuffer = this.mediaSource.addSourceBuffer('audio/mpeg');

 this.chunksLoader = new ChunksLoader(
 this.track,
 this.sourceBuffer,
 this.logger
 );

 this.chunksLoader.begin();
 };

 private handleEvent = (e) => {

 const currentTime = this.audioObj.currentTime.toFixed(2);
 const totalDuration = this.track.duration.toFixed(2);
 this.logger.warn(`MediaSource event: ${e.type} (${currentTime} of ${totalDuration} sec)`);

 this.currentTime$.next(this.audioObj.currentTime);

 const currentStatus = this.state$.getValue();

 switch (e.type) {
 case 'playing':
 currentStatus.status = StreamStatus.PLAYING;
 this.state$.next(currentStatus);
 break;
 case 'pause':
 currentStatus.status = StreamStatus.PAUSED;
 this.state$.next(currentStatus);
 break;
 case 'timeupdate':
 this.chunksLoader.handleOnTimeUpdate(this.audioObj.currentTime);
 break;
 case 'seeking':
 currentStatus.status = StreamStatus.SEEKING;
 this.state$.next(currentStatus);
 if (this.mediaSource.readyState == 'open') {
 this.sourceBuffer.abort();
 }
 this.chunksLoader.handleOnTimeUpdate(this.audioObj.currentTime);
 break;
 }
 };

 private addEvents(obj, events, handler) {
 events.forEach(event => obj.addEventListener(event, handler));
 }

}
</number></playerstate>

Running it will give me the following output :






Apologies for the screenshot but it's not possible to just copy the output without all the stack traces in Chrome.




What I also tried was following this example and call
sourceBuffer.abort()but that didn't work. It looks more like a hack that used to work years ago but it's still referenced in the docs (see "Example" -> "You can see something similar in action in Nick Desaulnier's bufferWhenNeeded demo ..").

case 'seeking':
 currentStatus.status = StreamStatus.SEEKING;
 this.state$.next(currentStatus); 
 if (this.mediaSource.readyState === 'open') {
 this.sourceBuffer.abort();
 } 
 break;


Trying with MP3


I have tested the above code under Chrome by converting tracks to MP3 :


ffmpeg -i input.mp3 -acodec aac -b:a 256k -f mp3 output.mp3


and creating a
SourceBufferusingaudio/mpegas type :

this.mediaSource.addSourceBuffer('audio/mpeg')


I have the same problem when seeking.


The issue wihout seeking


The above code has another issue :


After two minutes of playing, the audio playback starts to stutter and comes to a halt prematurely. So, the audio plays up to a point and then it stops without any obvious reason.


For whatever reason there is another
canplayandplayingevent. A few seconds after, the audio simply stops..



-
ffmpeg takes too long to start
17 octobre 2020, par SuspendedI have this command in python script, in a loop :


ffmpeg -i somefile.mp4 -ss 00:03:12 -t 00:00:35 piece.mp4 -loglevel error -stats


It cuts out pieces of input file (-i). Input filename, as well as start time (-ss) and length of the piece I cut out (-t) varies, so it reads number of mp4 files and cuts out number of pieces from each one. During execution of the script it might be called around 100 times. My problem is that each time before it starts, there is a delay of 6-15 seconds and it adds up to significant time. How can I get it to start immediately ?


Initially I thought it was process priority problem, but I noticed that even during the "pause", all processors work at 100%, so apparently some work is being done.


The script (process_videos.py) :


import subprocess
import sys
import math
import time

class TF:
 """TimeFormatter class (TF).
This class' reason for being is to convert time in short
form, e.g. 1:33, 0:32, or 23 into long form accepted by
mp4cut function in bash, e.g. 00:01:22, 00:00:32, etc"""

def toLong(self, shrt):
 """Converts time to its long form"""
 sx = '00:00:00'
 ladd = 8 - len(shrt)
 n = sx[:ladd] + shrt
 return n

def toShort(self, lng):
 """Converts time to short form"""
 if lng[0] == '0' or lng[0] == ':':
 return self.toShort(lng[1:])
 else:
 return lng

def toSeconds(self, any_time):
 """Converts time to seconds"""
 if len(any_time) < 3:
 return int(any_time)
 tt = any_time.split(':')
 if len(any_time) < 6: 
 return int(tt[0])*60 + int(tt[1])
 return int(tt[0])*3600 + int(tt[1])*60 + int(tt[2])

def toTime(self, secsInt):
 """"""
 tStr = ''
 hrs, mins, secs = 0, 0, 0
 if secsInt >= 3600:
 hrs = math.floor(secsInt / 3600)
 secsInt = secsInt % 3600
 if secsInt >= 60:
 mins = math.floor(secsInt / 60)
 secsInt = secsInt % 60
 secs = secsInt
 return str(hrs).zfill(2) + ':' + str(mins).zfill(2) + ':' + str(secs).zfill(2)

def minus(self, t_start, t_end):
 """"""
 t_e = self.toSeconds(t_end)
 t_s = self.toSeconds(t_start)
 t_r = t_e - t_s
 hrs, mins, secs = 0, 0, 0
 if t_r >= 3600:
 hrs = math.floor(t_r / 3600)
 t_r = t_r - (hrs * 3600)
 if t_r >= 60:
 mins = math.floor(t_r / 60)
 t_r = t_r - (mins * 60)
 secs = t_r
 hrsf = str(hrs).zfill(2)
 minsf = str(mins).zfill(2)
 secsf = str(secs).zfill(2)
 t_fnl = hrsf + ':' + minsf + ':' + secsf
 return t_fnl

def go_main():
 tf = TF()
 vid_n = 0
 arglen = len(sys.argv)
 if arglen == 2:
 with open(sys.argv[1], 'r') as f_in:
 lines = f_in.readlines()
 start = None
 end = None
 cnt = 0
 for line in lines:
 if line[:5] == 'BEGIN':
 start = cnt
 if line[:3] == 'END':
 end = cnt
 cnt += 1
 if start == None or end == None:
 print('Invalid file format. start = {}, end = {}'.format(start,end))
 return
 else:
 lines_r = lines[start+1:end]
 del lines
 print('videos to process: {}'.format(len(lines_r)))
 f_out_prefix = ""
 for vid in lines_r:
 vid_n += 1
 print('\nProcessing video {}/{}'.format(vid_n, len(lines_r)))
 f_out_prefix = 'v' + str(vid_n) + '-'
 dat = vid.split('!')[1:3]
 title = dat[0]
 dat_t = dat[1].split(',')
 v_pieces = len(dat_t)
 piece_n = 0
 video_pieces = []
 cmd1 = "echo -n \"\" > tmpfile"
 subprocess.run(cmd1, shell=True) 
 print(' new tmpfile created')
 for v_times in dat_t:
 piece_n += 1
 f_out = f_out_prefix + str(piece_n) + '.mp4'
 video_pieces.append(f_out)
 print(' piece filename {} added to video_pieces list'.format(f_out))
 v_times_spl = v_times.split('-')
 v_times_start = v_times_spl[0]
 v_times_end = v_times_spl[1]
 t_st = tf.toLong(v_times_start)
 t_dur = tf.toTime(tf.toSeconds(v_times_end) - tf.toSeconds(v_times_start))
 cmd3 = ["ffmpeg", "-i", title, "-ss", t_st, "-t", t_dur, f_out, "-loglevel", "error", "-stats"]
 print(' cutting out piece {}/{} - {}'.format(piece_n, len(dat_t), t_dur))
 subprocess.run(cmd3)
 for video_piece_name in video_pieces:
 cmd4 = "echo \"file " + video_piece_name + "\" >> tmpfile"
 subprocess.run(cmd4, shell=True)
 print(' filename {} added to tmpfile'.format(video_piece_name))
 vname = f_out_prefix[:-1] + ".mp4"
 print(' name of joined file: {}'.format(vname))
 cmd5 = "ffmpeg -f concat -safe 0 -i tmpfile -c copy joined.mp4 -loglevel error -stats"
 to_be_joined = " ".join(video_pieces)
 print(' joining...')
 join_cmd = subprocess.Popen(cmd5, shell=True)
 join_cmd.wait()
 print(' joined!')
 cmd6 = "mv joined.mp4 " + vname
 rename_cmd = subprocess.Popen(cmd6, shell=True)
 rename_cmd.wait()
 print(' File joined.mp4 renamed to {}'.format(vname))
 cmd7 = "rm " + to_be_joined
 rm_cmd = subprocess.Popen(cmd7, shell=True)
 rm_cmd.wait()
 print('rm command completed - pieces removed')
 cmd8 = "rm tmpfile"
 subprocess.run(cmd8, shell=True)
 print('tmpfile removed')
 print('All done')
 else:
 print('Incorrect number of arguments')

############################
if __name__ == '__main__':
 go_main()


process_videos.py is called from bash terminal like this :


$ python process_videos.py video_data 


video_data file has the following format :


BEGIN
!first_video.mp4!3-23,55-1:34,2:01-3:15,3:34-3:44!
!second_video.mp4!2-7,12-44,1:03-1:33!
END


My system details :


System: Host: snowflake Kernel: 5.4.0-52-generic x86_64 bits: 64 Desktop: Gnome 3.28.4
 Distro: Ubuntu 18.04.5 LTS
Machine: Device: desktop System: Gigabyte product: N/A serial: N/A
Mobo: Gigabyte model: Z77-D3H v: x.x serial: N/A BIOS: American Megatrends v: F14 date: 05/31/2012
CPU: Quad core Intel Core i5-3570 (-MCP-) cache: 6144 KB 
 clock speeds: max: 3800 MHz 1: 1601 MHz 2: 1601 MHz 3: 1601 MHz 4: 1602 MHz
Drives: HDD Total Size: 1060.2GB (55.2% used)
 ID-1: /dev/sda model: ST31000524AS size: 1000.2GB
 ID-2: /dev/sdb model: Corsair_Force_GT size: 60.0GB
Partition: ID-1: / size: 366G used: 282G (82%) fs: ext4 dev: /dev/sda1
 ID-2: swap-1 size: 0.70GB used: 0.00GB (0%) fs: swap dev: /dev/sda5
Info: Processes: 313 Uptime: 16:37 Memory: 3421.4/15906.9MB Client: Shell (bash) inxi: 2.3.56



UPDATE :


Following Charles' advice, I used performance sampling :


# perf record -a -g sleep 180


...and here's the report :


Samples: 74K of event 'cycles', Event count (approx.): 1043554519767
 Children Self Command Shared Object
- 50.56% 45.86% ffmpeg libavcodec.so.57.107.100 
 - 3.10% 0x4489480000002825 
 0.64% 0x7ffaf24b92f0 
 - 2.12% 0x5f7369007265646f 
 av_default_item_name 
 1.39% 0 
- 44.48% 40.59% ffmpeg libx264.so.152 
 5.78% x264_add8x8_idct_avx2.skip_prologue 
 3.13% x264_add8x8_idct_avx2.skip_prologue 
 2.91% x264_add8x8_idct_avx2.skip_prologue 
 2.31% x264_add8x8_idct_avx.skip_prologue 
 2.03% 0 
 1.78% 0x1 
 1.26% x264_add8x8_idct_avx2.skip_prologue 
 1.09% x264_add8x8_idct_avx.skip_prologue 
 1.06% x264_me_search_ref 
 0.97% x264_add8x8_idct_avx.skip_prologue 
 0.60% x264_me_search_ref 
- 38.01% 0.00% ffmpeg [unknown] 
 4.10% 0 
 - 3.49% 0x4489480000002825 
 0.70% 0x7ffaf24b92f0 
 0.56% 0x7f273ae822f0 
 0.50% 0x7f0c4768b2f0 
 - 2.29% 0x5f7369007265646f 
 av_default_item_name 
 1.99% 0x1 
 10.13% 10.12% ffmpeg [kernel.kallsyms] 
- 3.14% 0.73% ffmpeg libavutil.so.55.78.100 
 2.34% av_default_item_name 
- 1.73% 0.21% ffmpeg libpthread-2.27.so 
 - 0.70% pthread_cond_wait@@GLIBC_2.3.2 
 - 0.62% entry_SYSCALL_64_after_hwframe 
 - 0.62% do_syscall_64 
 - 0.57% __x64_sys_futex 
 0.52% do_futex 
 0.93% 0.89% ffmpeg libc-2.27.so 
- 0.64% 0.64% swapper [kernel.kallsyms] 
 0.63% secondary_startup_64 
 0.21% 0.18% ffmpeg libavfilter.so.6.107.100 
 0.20% 0.11% ffmpeg libavformat.so.57.83.100 
 0.12% 0.11% ffmpeg ffmpeg 
 0.11% 0.00% gnome-terminal- [unknown] 
 0.09% 0.07% ffmpeg libm-2.27.so 
 0.08% 0.07% ffmpeg ld-2.27.so 
 0.04% 0.04% gnome-terminal- libglib-2.0.so.0.5600.4




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