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Mise à jour de la version 0.1 vers 0.2
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Ecrire une actualité
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FFMPEG - Convert MKV to MP4 with all streams (subtitles) and watermark
24 juillet 2016, par ShinJiiI spent few hours researching and still have problem...
I want to convert .mkv file to .mp4 with watermark and subtitles (from mkv).I tried few commands but always error or success with watermark without subtitles.
Here are some commands I tried (not all) :Partial Success - some files are converted with subtitles, but I have 153 files that are converted without subtitles... (both from .mkv to .mp4) :
ffmpeg -i 025.mkv -vf "movie=logovideo2.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:10,subtitles=025.mkv [out]" 25.mp4
ffmpeg -i 001.mkv -i logovideo2.png -filter_complex "[0:v][1:v]overlay=0:0,subtitles=001.mkv[out]" -map "[out]" -map 0:a -c:s ass 001.mp4Fail - command with error :
ffmpeg -i 001.mkv -i logovideo2.png -filter_complex "[0:v][1:v]overlay=0:0[out]" -map "[out]" -map 0:a -map 0:s -c:s ass 001.mp4CODE (with error) :
ffmpeg version N-81036-g2b14204 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libass
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 50.100 / 57. 50.100
libavformat 57. 42.100 / 57. 42.100
libavdevice 57. 0.102 / 57. 0.102
libavfilter 6. 47.100 / 6. 47.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from '001.mkv':
Metadata:
title : DB
encoder : libebml v1.2.1 + libmatroska v1.1.1
creation_time : 2011-09-15 16:44:37
Duration: 00:24:39.37, start: 0.000000, bitrate: 1429 kb/s
Stream #0:0(jpn): Audio: vorbis, 48000 Hz, stereo, fltp (default)
Stream #0:1: Video: h264 (High), yuv420p, 720x540 [SAR 1:1 DAR 4:3], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Stream #0:2(pol): Subtitle: ass (default)
Metadata:
title : Grupa
Input #1, png_pipe, from 'logovideo2.png':
Duration: N/A, bitrate: N/A
Stream #1:0: Video: png, rgba(pc), 235x50 [SAR 2835:2835 DAR 47:10], 25 tbr, 25 tbn, 25 tbc
File '001.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x43c7a80] using SAR=1/1
[libx264 @ 0x43c7a80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x43c7a80] profile High, level 3.0
[libx264 @ 0x43c7a80] 264 - core 148 r2 3f5ed56 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=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_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=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=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
**[mp4 @ 0x442c9e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Last message repeated 2 times
[mp4 @ 0x442c9e0] Could not find tag for codec ass in stream #2, codec not currently supported in container**
Output #0, mp4, to '001.mp4':
Metadata:
title : DB
encoder : Lavf57.42.100
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 720x540 [SAR 1:1 DAR 4:3], q=-1--1, 23.98 fps, 24k tbn, 23.98 tbc (default)
Metadata:
encoder : Lavc57.50.100 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(jpn): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
encoder : Lavc57.50.100 aac
Stream #0:2(pol): Subtitle: ass (default)
Metadata:
title : Grupa
encoder : Lavc57.50.100 ass
Stream mapping:
Stream #0:1 (h264) -> overlay:main (graph 0)
Stream #1:0 (png) -> overlay:overlay (graph 0)
overlay (graph 0) -> Stream #0:0 (libx264)
Stream #0:0 -> #0:1 (vorbis (native) -> aac (native))
Stream #0:2 -> #0:2 (ass (ssa) -> ass (native))
**Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument**
[aac @ 0x43bf600] Qavg: -nan -
FFMPEG - Convert MKV to MP4 with all streams (subtitles) and watermark
13 mars 2024, par ShinJiiI spent few hours researching and still have problem...
I want to convert .mkv file to .mp4 with watermark and subtitles (from mkv).



I tried few commands but always error or success with watermark without subtitles.
Here are some commands I tried (not all) :



Partial Success - some files are converted with subtitles, but I have 153 files that are converted without subtitles... (both from .mkv to .mp4) :



ffmpeg -i 025.mkv -vf "movie=logovideo2.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:10,subtitles=025.mkv [out]" 25.mp4

ffmpeg -i 001.mkv -i logovideo2.png -filter_complex "[0:v][1:v]overlay=0:0,subtitles=001.mkv[out]" -map "[out]" -map 0:a -c:s ass 001.mp4



Fail - command with error :



ffmpeg -i 001.mkv -i logovideo2.png -filter_complex "[0:v][1:v]overlay=0:0[out]" -map "[out]" -map 0:a -map 0:s -c:s ass 001.mp4



CODE (with error) :



ffmpeg version N-81036-g2b14204 Copyright (c) 2000-2016 the FFmpeg developers
 built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
 configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libass
 libavutil 55. 28.100 / 55. 28.100
 libavcodec 57. 50.100 / 57. 50.100
 libavformat 57. 42.100 / 57. 42.100
 libavdevice 57. 0.102 / 57. 0.102
 libavfilter 6. 47.100 / 6. 47.100
 libswscale 4. 1.100 / 4. 1.100
 libswresample 2. 1.100 / 2. 1.100
 libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from '001.mkv':
 Metadata:
 title : DB
 encoder : libebml v1.2.1 + libmatroska v1.1.1
 creation_time : 2011-09-15 16:44:37
 Duration: 00:24:39.37, start: 0.000000, bitrate: 1429 kb/s
 Stream #0:0(jpn): Audio: vorbis, 48000 Hz, stereo, fltp (default)
 Stream #0:1: Video: h264 (High), yuv420p, 720x540 [SAR 1:1 DAR 4:3], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
 Stream #0:2(pol): Subtitle: ass (default)
 Metadata:
 title : Grupa
Input #1, png_pipe, from 'logovideo2.png':
 Duration: N/A, bitrate: N/A
 Stream #1:0: Video: png, rgba(pc), 235x50 [SAR 2835:2835 DAR 47:10], 25 tbr, 25 tbn, 25 tbc
File '001.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x43c7a80] using SAR=1/1
[libx264 @ 0x43c7a80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x43c7a80] profile High, level 3.0
[libx264 @ 0x43c7a80] 264 - core 148 r2 3f5ed56 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=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_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=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=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
**[mp4 @ 0x442c9e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
 Last message repeated 2 times
[mp4 @ 0x442c9e0] Could not find tag for codec ass in stream #2, codec not currently supported in container**
Output #0, mp4, to '001.mp4':
 Metadata:
 title : DB
 encoder : Lavf57.42.100
 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 720x540 [SAR 1:1 DAR 4:3], q=-1--1, 23.98 fps, 24k tbn, 23.98 tbc (default)
 Metadata:
 encoder : Lavc57.50.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
 Stream #0:1(jpn): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s (default)
 Metadata:
 encoder : Lavc57.50.100 aac
 Stream #0:2(pol): Subtitle: ass (default)
 Metadata:
 title : Grupa
 encoder : Lavc57.50.100 ass
Stream mapping:
 Stream #0:1 (h264) -> overlay:main (graph 0)
 Stream #1:0 (png) -> overlay:overlay (graph 0)
 overlay (graph 0) -> Stream #0:0 (libx264)
 Stream #0:0 -> #0:1 (vorbis (native) -> aac (native))
 Stream #0:2 -> #0:2 (ass (ssa) -> ass (native))
**Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument**
[aac @ 0x43bf600] Qavg: -nan


-
I want to take any Audio from a file and encode it as PCM_ALAW. My Example is a .m4a file to .wav file
22 novembre 2023, par ClockmanI have been working on this for a while now while am generally new to ffmpeg library, I have studied it a bit. The challenge I have that at the point of witting to file I get the following exception.


"Exception thrown at 0x00007FFACA8305B3 (avformat-60.dll) in FfmpegPractice.exe : 0xC0000005 : Access violation writing location 0x0000000000000000.". I understand this means am writing to an uninitialized buffer am unable to discover why this is happening. The exception call stack shows the following


avformat-60.dll!avformat_write_header() C
avformat-60.dll!ff_write_chained() C
avformat-60.dll!ff_write_chained() C
avformat-60.dll!av_write_frame() C
FfmpegPractice.exe!main() Line 215 C++


Some things I have tried


This code is part of a larger project built with CMake but for some reason I could no step into ffmpeg library while debugging, So I recompiled ffmpeg ensured debugging was enabled so I could drill down to the root cause but I still could not step into the ffmpeg library.


I then created a minimal project using visual studio c++ console project and I still could not step into the code.


I have read through many ffmpeg docs and some I could find on the internet and I still could not solve it.


This is the code


#include <iostream>

extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libswresample></libswresample>swresample.h>
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>audio_fifo.h>
}

using namespace std;

//in audio file
string filename{ "rapid_caller_test.m4a" };
AVFormatContext* pFormatCtx{};
AVCodecContext* pCodecCtx{};
AVStream* pStream{};

//out audio file
string outFilename{ "output.wav" };
AVFormatContext* pOutFormatCtx{ nullptr };
AVCodecContext* pOutCodecCtx{ nullptr };
AVIOContext* pOutIoContext{ nullptr };
const AVCodec* pOutCodec{ nullptr };
AVStream* pOutStream{ nullptr };
const int OUTPUT_CHANNELS = 1;
const int SAMPLE_RATE = 8000;
const int OUT_BIT_RATE = 64000;
uint8_t** convertedSamplesBuffer{ nullptr };
int64_t dstNmbrSamples{ 0 };
int dstLineSize{ 0 };
static int64_t pts{ 0 };

//conversion context;
SwrContext* swr{};

uint32_t i{ 0 };
int audiostream{ -1 };


void cleanUp() 
{
 avcodec_free_context(&pOutCodecCtx);;
 avio_closep(&(pOutFormatCtx)->pb);
 avformat_free_context(pOutFormatCtx);
 pOutFormatCtx = nullptr;
}

int main()
{

/*
* section to setup input file
*/
if (avformat_open_input(&pFormatCtx, filename.data(), nullptr, nullptr) != 0) {
 cout << "could not open file " << filename << endl;
 return -1;
}
if (avformat_find_stream_info(pFormatCtx, nullptr) < 0) {
 cout << "Could not retrieve stream information from file " << filename << endl;
 return -1;
}
av_dump_format(pFormatCtx, 0, filename.c_str(), 0);

for (i = 0; i < pFormatCtx->nb_streams; i++) {
 if (pFormatCtx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 audiostream = i;
 break;
 }
}
if (audiostream == -1) {
 cout << "did not find audio stream" << endl;
 return -1;
}

pStream = pFormatCtx->streams[audiostream];
const AVCodec* pCodec{ avcodec_find_decoder(pStream->codecpar->codec_id) };
pCodecCtx = avcodec_alloc_context3(pCodec);
avcodec_parameters_to_context(pCodecCtx, pStream->codecpar);
if (avcodec_open2(pCodecCtx, pCodec, nullptr)) {
 cout << "could not open codec" << endl;
 return -1;
}

/*
* section to set up output file which is a G711 audio
*/
if (avio_open(&pOutIoContext, outFilename.data(), AVIO_FLAG_WRITE)) {
 cout << "could not open out put file" << endl;
 return -1;
}
if (!(pOutFormatCtx = avformat_alloc_context())) {
 cout << "could not create format conext" << endl;
 cleanUp();
 return -1;
}
pOutFormatCtx->pb = pOutIoContext;
if (!(pOutFormatCtx->oformat = av_guess_format(nullptr, outFilename.data(), nullptr))) {
 cout << "could not find output file format" << endl;
 cleanUp();
 return -1;
}
if (!(pOutFormatCtx->url = av_strdup(outFilename.data()))) {
 cout << "could not allocate file name" << endl;
 cleanUp();
 return -1;
}
if (!(pOutCodec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW))) {
 cout << "codec not found" << endl;
 cleanUp();
 return -1;
}
if (!(pOutStream = avformat_new_stream(pOutFormatCtx, nullptr))) {
 cout << "could not create new stream" << endl;
 cleanUp();
 return -1;
}
if (!(pOutCodecCtx = avcodec_alloc_context3(pOutCodec))) {
 cout << "could not allocate codec context" << endl;
 return -1;
}
av_channel_layout_default(&pOutCodecCtx->ch_layout, OUTPUT_CHANNELS);
pOutCodecCtx->sample_rate = SAMPLE_RATE;
pOutCodecCtx->sample_fmt = pOutCodec->sample_fmts[0];
pOutCodecCtx->bit_rate = OUT_BIT_RATE;

//setting sample rate for the container
pOutStream->time_base.den = SAMPLE_RATE;
pOutStream->time_base.num = 1;
if (pOutFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
 pOutCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

if (avcodec_open2(pOutCodecCtx, pOutCodec, nullptr)) {
 cout << "could not open output codec" << endl;
 cleanUp();
 return -1;
}
if ((avcodec_parameters_from_context(pOutStream->codecpar, pOutCodecCtx)) < 0) {
 cout << "could not initialize stream parameters" << endl;
} 

AVPacket* packet = av_packet_alloc();

swr = swr_alloc();
swr_alloc_set_opts2(&swr, &pOutCodecCtx->ch_layout, pOutCodecCtx->sample_fmt, pOutCodecCtx->sample_rate,&pCodecCtx->ch_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0, nullptr);
swr_init(swr);

int ret{};
int bSize{};
while (av_read_frame(pFormatCtx, packet) >= 0) {
 AVFrame* pFrame = av_frame_alloc();
 AVFrame* pOutFrame = av_frame_alloc();
 if (packet->stream_index == audiostream) {
 ret = avcodec_send_packet(pCodecCtx, packet);
 while (ret >= 0) {
 ret = avcodec_receive_frame(pCodecCtx, pFrame);
 if (ret == AVERROR(EAGAIN))
 continue;
 else if (ret == AVERROR_EOF)
 break;
 dstNmbrSamples = av_rescale_rnd(swr_get_delay(swr, pCodecCtx->sample_rate) + pFrame->nb_samples, pOutCodecCtx->sample_rate, pCodecCtx->sample_rate, AV_ROUND_UP);
 if ((av_samples_alloc_array_and_samples(&convertedSamplesBuffer, &dstLineSize, pOutCodecCtx->ch_layout.nb_channels,dstNmbrSamples, pOutCodecCtx->sample_fmt, 0)) < 0) {
 cout << "coult not allocate samples array and buffer" << endl;
 }
 int channel_samples_count{ 0 };
 channel_samples_count = swr_convert(swr, convertedSamplesBuffer, dstNmbrSamples, (const uint8_t**)pFrame->data, pFrame->nb_samples);
 bSize = av_samples_get_buffer_size(&dstLineSize, pOutCodecCtx->ch_layout.nb_channels, channel_samples_count, pOutCodecCtx->sample_fmt, 0);
 cout << "no of samples is " << channel_samples_count << " the buffer size " << bSize << endl;
 pOutFrame->nb_samples = channel_samples_count;
 av_channel_layout_copy(&pOutFrame->ch_layout, &pOutCodecCtx->ch_layout);
 pOutFrame->format = pOutCodecCtx->sample_fmt;
 pOutFrame->sample_rate = pOutCodecCtx->sample_rate;
 if ((av_frame_get_buffer(pOutFrame, 0)) < 0) {
 cout << "could not allocate output frame samples " << endl;
 av_frame_free(&pOutFrame);
 }
 
 //populate out frame buffer
 av_frame_make_writable(pOutFrame);
 for (int i{ 0 }; i < bSize; i++) {
 pOutFrame->data[0][i] = convertedSamplesBuffer[0][i];
 cout << pOutFrame->data[0][i];
 }
 if (pOutFrame) {
 pOutFrame->pts = pts;
 pts += pOutFrame->nb_samples;
 }
 int res = avcodec_send_frame(pOutCodecCtx, pOutFrame);
 if (res < 0) {
 cout << "error sending frame to encoder" << endl;
 cleanUp();
 return -1;
 }
 //int er = avformat_write_header(pOutFormatCtx,nullptr);
 AVPacket* pOutPacket = av_packet_alloc();
 pOutPacket->time_base.num = 1;
 pOutPacket->time_base.den = 8000;
 if (pOutPacket == nullptr) {
 cout << "unable to allocate packet" << endl;
 }
 while (res >= 0) {
 res = avcodec_receive_packet(pOutCodecCtx, pOutPacket);
 if (res == AVERROR(EAGAIN))
 continue;
 else if (ret == AVERROR_EOF)
 break;
 av_packet_rescale_ts(pOutPacket, pOutCodecCtx->time_base, pOutFormatCtx->streams[0]->time_base);
 //av_dump_format(pOutFormatCtx, 0, outFilename.c_str(), 1);
 if (av_write_frame(pOutFormatCtx, pOutPacket) < 0) {
 cout << "could not write frame" << endl;
 }
 }
 }
}
 av_frame_free(&pFrame);
 av_frame_free(&pOutFrame);
}
if (av_write_trailer(pOutFormatCtx) < 0) {
 cout << "could not write file trailer" << endl;
}
swr_free(&swr);
avcodec_free_context(&pOutCodecCtx);
av_packet_free(&packet);
}
</iostream>

Error/Exception


The exception is thrown when I call


if (av_write_frame(pOutFormatCtx, pOutPacket) < 0) { cout << "could not write frame" << endl; }
I also called this line

//int er = avformat_write_header(pOutFormatCtx,nullptr);

to see if I will get an exception but it did not throw any exception.


I have spent weeks on this issue with no success.
My goal is to take any audio from a file an be able to resample it if need be, and transcode it to PCM_ALAW.
I will appreciate any help I can get.


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