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Video d’abeille en portrait
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Mis à jour : Février 2012
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Type : Video
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FFMPEG and MSA1 Codec Issues when Converting WMV to MP4 Video
30 décembre 2023, par user136819I have a WMV video file (around 60 MB size, 01:23:00 length) and I'm running a Windows Server 2016 64-bit machine for testing.

As far as I know, this video file was recorded using Microsoft Live Meeting 2007 back in April 2010 and there is also an associated audio file (which I'm able to play).

I tried to play the video file on my machine using VLC player (latest update), however, I always see this weird MSA1 codec error :




So I tried to resolve this problem by downloading and installing the Combined-Community-Codec-Pack-64bit-2015-10-18, and using its MPC-HC 64-bit player to play the video...
I still couldn't get it to play ! Here are the logs shown by the MPC player :




WM ASF Reader::Raw Video 0

Media Type 0:
--------------------------
Video: MSA1 1440x900 5fps 164kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Video {73646976-0000-0010-8000-00AA00389B71}
subtype: Unknown GUID Name {3141534D-0000-0010-8000-00AA00389B71}
formattype: FORMAT_VideoInfo {05589F80-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 0
bTemporalCompression: 1
lSampleSize: 0
cbFormat: 88

VIDEOINFOHEADER:
rcSource: (0,0)-(1440,900)
rcTarget: (0,0)-(1440,900)
dwBitRate: 164794
dwBitErrorRate: 0
AvgTimePerFrame: 2000000

BITMAPINFOHEADER:
biSize: 40
biWidth: 1440
biHeight: 900
biPlanes: 1
biBitCount: 16
biCompression: MSA1
biSizeImage: 66101
biXPelsPerMeter: 0
biYPelsPerMeter: 0
biClrUsed: 0
biClrImportant: 0

pbFormat:
0000: 00 00 00 00 00 00 00 00 a0 05 00 00 84 03 00 00 ........ ...„...
0010: 00 00 00 00 00 00 00 00 a0 05 00 00 84 03 00 00 ........ ...„...
0020: ba 83 02 00 00 00 00 00 80 84 1e 00 00 00 00 00 ºƒ......€„......
0030: 28 00 00 00 a0 05 00 00 84 03 00 00 01 00 10 00 (... ...„.......
0040: 4d 53 41 31 35 02 01 00 00 00 00 00 00 00 00 00 MSA15...........
0050: 00 00 00 00 00 00 00 00 ........



Since that didn't work, I tried installing the Microsoft MPEG-4 V1/2/3 VKI Codec for ASF files and still couldn't get the players to play the video and saw the same errors as above !


As a final attempt, I tried to convert this video to MP4 format using FFMPEG (version from 2020). Even this failed. Here are the logs :


D:\ffmpeg-20200403-52523b6-win64-static\bin>ffmpeg.exe -i input-video.wmv -c:v copy -c:a aac -q:a 100 output.mp4
ffmpeg version git-2020-04-03-52523b6 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200328
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 42.102 / 56. 42.102
 libavcodec 58. 77.101 / 58. 77.101
 libavformat 58. 42.100 / 58. 42.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 77.101 / 7. 77.101
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
[msa1 @ 000002505bb847c0] Image dimensions should be a multiple of 16.
[asf @ 000002505bb4b600] Failed to open codec in avformat_find_stream_info
[msa1 @ 000002505bb847c0] Image dimensions should be a multiple of 16.
Input #0, asf, from 'input-video.wmv':
 Metadata:
 WMFSDKNeeded : 0.0.0.0000
 WMFSDKVersion : 10.00.00.4072
 IsVBR : 0
 Duration: 01:22:49.28, start: 0.000000, bitrate: 96 kb/s
 Stream #0:0(ger): Video: msa1 (MSA1 / 0x3141534D), none, 1440x900, 164 kb/s, 5 tbr, 1k tbn, 1k tbc
[mp4 @ 000002505bb4ce40] Could not find tag for codec msa1 in stream #0, codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
 Last message repeated 1 times

D:\ffmpeg-20200403-52523b6-win64-static\bin>



I wish to know, is there any other way I can convert this video to a more modern format so that I can play it on a modern machine and modern video player ?

What other codecs am I missing here ? (if so, where can I get them ?)

FYI, as mentioned earlier at the top, I'm only able to play the associated audio file...

I'm fairly new to all this stuff, so any help is appreciated.


Thanks in advance ! :)


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lavc/aarch64 : h264qpel, add 10-bit lowpass_8_10 based functions
4 décembre 2023, par Mikhail Nitenkolavc/aarch64 : h264qpel, add 10-bit lowpass_8_10 based functions
Benchmarks A53 A55 A72 A76
avg_h264_qpel_8_mc01_10_c : 936.5 924.0 656.0 504.7
avg_h264_qpel_8_mc01_10_neon : 234.7 202.0 120.7 63.2
avg_h264_qpel_8_mc02_10_c : 921.0 920.0 669.2 493.7
avg_h264_qpel_8_mc02_10_neon : 202.0 173.2 102.7 58.5
avg_h264_qpel_8_mc03_10_c : 936.5 924.0 656.0 509.5
avg_h264_qpel_8_mc03_10_neon : 236.2 203.7 120.0 63.2
avg_h264_qpel_8_mc10_10_c : 1441.0 1437.7 806.7 478.5
avg_h264_qpel_8_mc10_10_neon : 325.7 324.0 153.7 94.2
avg_h264_qpel_8_mc11_10_c : 2160.7 2148.2 1366.7 906.7
avg_h264_qpel_8_mc11_10_neon : 492.0 464.0 242.5 134.5
avg_h264_qpel_8_mc13_10_c : 2157.0 2138.2 1357.0 908.2
avg_h264_qpel_8_mc13_10_neon : 494.0 467.2 242.0 140.0
avg_h264_qpel_8_mc20_10_c : 1433.5 1410.0 785.2 486.0
avg_h264_qpel_8_mc20_10_neon : 293.7 289.7 138.0 91.5
avg_h264_qpel_8_mc30_10_c : 1458.5 1461.7 813.7 483.2
avg_h264_qpel_8_mc30_10_neon : 341.7 339.2 154.0 95.2
avg_h264_qpel_8_mc31_10_c : 2194.7 2197.2 1358.7 928.0
avg_h264_qpel_8_mc31_10_neon : 520.0 495.0 245.5 142.5
avg_h264_qpel_8_mc33_10_c : 2188.0 2205.5 1356.7 910.7
avg_h264_qpel_8_mc33_10_neon : 521.0 494.5 245.7 145.7
avg_h264_qpel_16_mc01_10_c : 3717.2 3595.0 2610.0 2012.0
avg_h264_qpel_16_mc01_10_neon : 920.5 791.5 483.2 240.5
avg_h264_qpel_16_mc02_10_c : 3684.0 3633.0 2659.0 1919.7
avg_h264_qpel_16_mc02_10_neon : 790.7 678.2 409.2 217.0
avg_h264_qpel_16_mc03_10_c : 3726.5 3596.0 2606.7 2010.0
avg_h264_qpel_16_mc03_10_neon : 922.0 792.5 483.2 239.7
avg_h264_qpel_16_mc10_10_c : 5912.0 5803.2 3241.5 1916.7
avg_h264_qpel_16_mc10_10_neon : 1267.5 1277.2 616.5 365.0
avg_h264_qpel_16_mc11_10_c : 8599.2 8482.5 5338.0 3616.2
avg_h264_qpel_16_mc11_10_neon : 1913.0 1827.0 956.2 542.2
avg_h264_qpel_16_mc13_10_c : 8643.7 8488.5 5388.0 3628.5
avg_h264_qpel_16_mc13_10_neon : 1914.7 1828.7 969.2 530.5
avg_h264_qpel_16_mc20_10_c : 5719.5 5641.0 3147.0 1946.2
avg_h264_qpel_16_mc20_10_neon : 1139.5 1150.0 539.5 344.0
avg_h264_qpel_16_mc30_10_c : 5930.0 5872.5 3267.5 1918.0
avg_h264_qpel_16_mc30_10_neon : 1331.5 1341.2 616.5 369.5
avg_h264_qpel_16_mc31_10_c : 8758.7 8697.7 5353.0 3630.7
avg_h264_qpel_16_mc31_10_neon : 2018.7 1941.7 982.2 574.7
avg_h264_qpel_16_mc33_10_c : 8683.2 8675.2 5339.2 3634.7
avg_h264_qpel_16_mc33_10_neon : 2019.7 1940.2 994.5 566.0
put_h264_qpel_8_mc01_10_c : 854.2 843.0 599.2 478.0
put_h264_qpel_8_mc01_10_neon : 192.7 168.0 101.7 56.7
put_h264_qpel_8_mc02_10_c : 766.5 760.0 550.2 441.0
put_h264_qpel_8_mc02_10_neon : 160.0 139.2 88.7 53.0
put_h264_qpel_8_mc03_10_c : 854.2 843.0 599.2 479.0
put_h264_qpel_8_mc03_10_neon : 194.2 169.7 102.0 56.2
put_h264_qpel_8_mc10_10_c : 1352.7 1353.7 749.7 446.7
put_h264_qpel_8_mc10_10_neon : 289.7 294.2 135.5 88.5
put_h264_qpel_8_mc11_10_c : 2080.0 2066.2 1309.5 876.7
put_h264_qpel_8_mc11_10_neon : 450.0 429.7 229.7 131.2
put_h264_qpel_8_mc13_10_c : 2074.7 2060.2 1294.5 870.5
put_h264_qpel_8_mc13_10_neon : 452.5 434.5 226.5 130.0
put_h264_qpel_8_mc20_10_c : 1221.5 1216.0 684.5 399.7
put_h264_qpel_8_mc20_10_neon : 257.7 262.5 121.2 78.7
put_h264_qpel_8_mc30_10_c : 1379.0 1374.7 757.2 449.5
put_h264_qpel_8_mc30_10_neon : 305.7 310.2 135.5 86.5
put_h264_qpel_8_mc31_10_c : 2109.2 2119.7 1299.5 878.0
put_h264_qpel_8_mc31_10_neon : 478.0 458.5 226.0 137.2
put_h264_qpel_8_mc33_10_c : 2101.5 2115.2 1306.5 887.0
put_h264_qpel_8_mc33_10_neon : 479.0 458.7 229.7 141.7
put_h264_qpel_16_mc01_10_c : 3485.7 3396.7 2460.5 1914.5
put_h264_qpel_16_mc01_10_neon : 752.5 665.5 397.0 213.2
put_h264_qpel_16_mc02_10_c : 3103.5 3023.2 2154.7 1720.7
put_h264_qpel_16_mc02_10_neon : 622.7 551.2 347.7 196.2
put_h264_qpel_16_mc03_10_c : 3486.2 3394.0 2436.5 1917.7
put_h264_qpel_16_mc03_10_neon : 754.0 666.5 397.0 215.7
put_h264_qpel_16_mc10_10_c : 5533.0 5488.5 2989.0 1783.0
put_h264_qpel_16_mc10_10_neon : 1123.5 1165.2 535.2 334.7
put_h264_qpel_16_mc11_10_c : 8437.7 8281.2 5209.0 3510.7
put_h264_qpel_16_mc11_10_neon : 1745.0 1697.0 878.5 513.5
put_h264_qpel_16_mc13_10_c : 8567.7 8468.0 5221.5 3528.0
put_h264_qpel_16_mc13_10_neon : 1751.7 1698.2 889.2 507.0
put_h264_qpel_16_mc20_10_c : 4907.5 4885.0 2786.2 1607.5
put_h264_qpel_16_mc20_10_neon : 995.5 1034.5 475.5 307.0
put_h264_qpel_16_mc30_10_c : 5579.7 5537.7 3045.2 1789.5
put_h264_qpel_16_mc30_10_neon : 1187.5 1231.2 532.5 334.5
put_h264_qpel_16_mc31_10_c : 8677.2 8672.5 5204.2 3516.0
put_h264_qpel_16_mc31_10_neon : 1850.7 1813.2 893.0 545.2
put_h264_qpel_16_mc33_10_c : 8688.7 8671.2 5223.2 3512.0
put_h264_qpel_16_mc33_10_neon : 1851.7 1814.2 908.5 535.2Signed-off-by : Mikhail Nitenko <mnitenko@gmail.com>
Signed-off-by : Martin Storsjö <martin@martin.st> -
Encode audio to aac with libavcodec
11 juin 2024, par ryanI'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac
(libfaac support enabled). I do this by calling avcodec_encode_audio
repeatedly with codec_context->frame_size samples each time. The first four
calls return successfully, but the fifth call never returns. When I use gdb
to break, the stack is corrupt.



If I use audacity to export the pcm data to a .wav file, then I can use
command-line ffmpeg to convert to aac without any issues, so I'm sure it's
something I'm doing wrong.



I've written a small test program that duplicates my problem. It reads the
test data from a file, which is available here :
http://birdie.protoven.com/audio.pcm ( 2 seconds of signed 16 bit LE pcm)



I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4.



ffmpeg version info :



FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers
 built on Mar 3 2010 15:40:46 with gcc 4.4.1
 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared
 libavutil 50.10. 0 / 50.10. 0
 libavcodec 52.55. 0 / 52.55. 0
 libavformat 52.54. 0 / 52.54. 0
 libavdevice 52. 2. 0 / 52. 2. 0
 libswscale 0.10. 0 / 0.10. 0
 libpostproc 51. 2. 0 / 51. 2. 0




Is there something I'm not setting, or setting incorrectly in my codec
context, maybe ? Any help is greatly appreciated !



Here is my test code :



#include 
#include <libavcodec></libavcodec>avcodec.h>

void EncodeTest(int sampleRate, int channels, int audioBitrate,
 uint8_t *audioData, size_t audioSize)
{
 AVCodecContext *audioCodec;
 AVCodec *codec;
 uint8_t *buf;
 int bufSize, frameBytes;

 avcodec_register_all();

 //Set up audio encoder
 codec = avcodec_find_encoder(CODEC_ID_AAC);
 if (codec == NULL) return;
 audioCodec = avcodec_alloc_context();
 audioCodec->bit_rate = audioBitrate;
 audioCodec->sample_fmt = SAMPLE_FMT_S16;
 audioCodec->sample_rate = sampleRate;
 audioCodec->channels = channels;
 audioCodec->profile = FF_PROFILE_AAC_MAIN;
 audioCodec->time_base = (AVRational){1, sampleRate};
 audioCodec->codec_type = CODEC_TYPE_AUDIO;
 if (avcodec_open(audioCodec, codec) < 0) return;

 bufSize = FF_MIN_BUFFER_SIZE * 10;
 buf = (uint8_t *)malloc(bufSize);
 if (buf == NULL) return;

 frameBytes = audioCodec->frame_size * audioCodec->channels * 2;
 while (audioSize >= frameBytes)
 {
 int packetSize;

 packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData);
 printf("encoder returned %d bytes of data\n", packetSize);
 audioData += frameBytes;
 audioSize -= frameBytes;
 }
}

int main()
{
 FILE *stream = fopen("audio.pcm", "rb");
 size_t size;
 uint8_t *buf;

 if (stream == NULL)
 {
 printf("Unable to open file\n");
 return 1;
 }

 fseek(stream, 0, SEEK_END);
 size = ftell(stream);
 fseek(stream, 0, SEEK_SET);
 buf = (uint8_t *)malloc(size);
 fread(buf, sizeof(uint8_t), size, stream);
 fclose(stream);

 EncodeTest(32000, 2, 448000, buf, size);
}