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• ffmpeg error on decode

  25 octobre 2013, par ademar111190

  I'm developing an android app with the libav and I'm trying decode a 3gp with code below :

  #define simbiLog(...) __android_log_print(ANDROID_LOG_DEBUG, "simbiose", __VA_ARGS__)

  ...

  AVCodec *codec;
  
  AVCodecContext *c = NULL;
  
  int len;
  
  FILE *infile, *outfile;
  
  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
  AVPacket avpkt;
  
  AVFrame *decoded_frame = NULL;
  
  
  
  simbiLog("inbuf size: %d", sizeof(inbuf) / sizeof(inbuf[0]));
  
  
  
  av_register_all();
  
  av_init_packet(&avpkt);
  
  
  
  codec = avcodec_find_decoder(AV_CODEC_ID_AMR_NB);
  
  if (!codec) {
  
      simbiLog("codec not found");
  
      return ERROR;
  
  }
  
  
  
  c = avcodec_alloc_context3(codec);
  
  if (!c) {
  
      simbiLog("Could not allocate audio codec context");
  
      return ERROR;
  
  }
  
  
  
  int open = avcodec_open2(c, codec, NULL);
  
  if (open < 0) {
  
      simbiLog("could not open codec %d", open);
  
      return ERROR;
  
  }
  
  
  
  infile = fopen(inputPath, "rb");
  
  if (!infile) {
  
      simbiLog("could not open %s", inputPath);
  
      return ERROR;
  
  }
  
  
  
  outfile = fopen(outputPath, "wb");
  
  if (!outfile) {
  
      simbiLog("could not open %s", outputPath);
  
      return ERROR;
  
  }
  
  
  
  avpkt.data = inbuf;
  
  avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, infile);
  
  int iterations = 0;
  
  
  
  while (avpkt.size > 0) {
  
      simbiLog("iteration %d", (++iterations));
  
      simbiLog("avpkt.size %d avpkt.data %X", avpkt.size, avpkt.data);
  
      int got_frame = 0;
  
  
  
      if (!decoded_frame) {
  
          if (!(decoded_frame = avcodec_alloc_frame())) {
  
              simbiLog("out of memory");
  
              return ERROR;
  
          }
  
      } else {
  
          avcodec_get_frame_defaults(decoded_frame);
  
      }
  
  
  
      //below the error, but it isn't occur on first time, only in 4th loop interation
  
      len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); 
  
      if (len < 0) {
  
          simbiLog("Error while decoding error %d frame %d duration %d", len, got_frame, avpkt.duration);
  
          return ERROR;
  
      } else {
  
          simbiLog("Decoding length %d frame %d duration %d", len, got_frame, avpkt.duration);
  
      }
  
  
  
      if (got_frame) {
  
          int data_size = av_samples_get_buffer_size(NULL, c->channels, decoded_frame->nb_samples, c->sample_fmt, 1);
  
          size_t* fwrite_size = fwrite(decoded_frame->data[0], 1, data_size, outfile);
  
          simbiLog("fwrite returned %d", fwrite_size);
  
      }
  
      avpkt.size -= len;
  
      avpkt.data += len;
  
      if (avpkt.size < AUDIO_REFILL_THRESH) {
  
          memmove(inbuf, avpkt.data, avpkt.size);
  
          avpkt.data = inbuf;
  
          len = fread(avpkt.data + avpkt.size, 1, AUDIO_INBUF_SIZE - avpkt.size, infile);
  
          if (len > 0)
  
              avpkt.size += len;
  
          simbiLog("fread returned %d", len);
  
      }
  
  }
  
  
  
  fclose(outfile);
  
  fclose(infile);
  
  
  
  avcodec_close(c);
  
  av_free(c);
  
  av_free(decoded_frame);

  but I'm getting the follow log and error :

  inbuf size: 20488
  
  iteration 1
  
  avpkt.size 3305 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 2
  
  avpkt.size 3292 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 3
  
  avpkt.size 3279 avpkt.data BEEED40C
  
  Decoding length 14 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 4
  
  avpkt.size 3265 avpkt.data BEEED40C
  
  Error while decoding error -1052488119 frame 0 duration 0

  the audio file I'm trying decode :

  $ avprobe blue.3gp 
  
  avprobe version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2007-2013 the Libav developers
  
    built on Mar 30 2013 22:23:21 with gcc 4.7.2
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'blue.3gp':
  
    Metadata:
  
      major_brand     : 3gp4
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2013-09-19 18:53:38
  
    Duration: 00:00:01.52, start: 0.000000, bitrate: 17 kb/s
  
      Stream #0.0(eng): Audio: amrnb, 8000 Hz, 1 channels, flt, 12 kb/s
  
      Metadata:
  
        creation_time   : 2013-09-19 18:53:38

  thanks a lot !

  ---

  EDITED

  I read on ffmper documentation about the method avcodec_decode_audio4 the follow :

  @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE larger than the actual read bytes because some optimized bitstream readers read 32 or 64 bits at once and could read over the end.
  
  @note You might have to align the input buffer. The alignment requirements depend on the CPU and the decoder.

  and I see here a solution using posix_memalign, to android i founded a similar method called memalign, so i did the change :

  removed :

  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];

  inserted :

  int inbufSize = sizeof(uint8_t) * (AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
  
  uint8_t *inbuf = memalign(FF_INPUT_BUFFER_PADDING_SIZE, inbufSize);
  
  simbiLog("inbuf size: %d", inbufSize);
  
  for (; inbufSize >= 0; inbufSize--)
  
      simbiLog("inbuf position: %d index: %p", inbufSize, &inbuf[inbufSize]);

  I'm getting the correct memory sequence position, but the error not changed.

  A piece of outpout :

  inbuf position: 37 index: 0x4e43d745
  
  inbuf position: 36 index: 0x4e43d744
  
  inbuf position: 35 index: 0x4e43d743
  
  inbuf position: 34 index: 0x4e43d742
  
  inbuf position: 33 index: 0x4e43d741
  
  inbuf position: 32 index: 0x4e43d740
  
  inbuf position: 31 index: 0x4e43d73f
  
  inbuf position: 30 index: 0x4e43d73e
  
  inbuf position: 29 index: 0x4e43d73d
  
  inbuf position: 28 index: 0x4e43d73c
  
  inbuf position: 27 index: 0x4e43d73b
  
  inbuf position: 26 index: 0x4e43d73a
  
  inbuf position: 25 index: 0x4e43d739
  
  inbuf position: 24 index: 0x4e43d738
  
  inbuf position: 23 index: 0x4e43d737
  
  inbuf position: 22 index: 0x4e43d736
  
  inbuf position: 21 index: 0x4e43d735
  
  inbuf position: 20 index: 0x4e43d734
  
  inbuf position: 19 index: 0x4e43d733

• avformat/movenc : Fix flush fragment

  22 mai, par Zhao Zhili

  avformat/movenc : Fix flush fragment
  The follow cmd output corrupted file before the patch :
  ffmpeg -f lavfi -i color=blue,trim=duration=0.04 \
  
  	-f lavfi -i anullsrc,atrim=duration=2 \
  
  	-movflags +empty_moov+hybrid_fragmented \
  
  	-frag_duration 1000000 \
  
  	-frag_interleave 1 \
  
  	output.mp4
  1. first_track is the first track with track->entry != 0. As in the
  
  command above, video track (track index 0) has a single frame. When
  
  flush the second fragment, first_track is 1, the audio track.
  2. write_moof = i == first_track, so write_moof is false for i = 0.
  3. When mov->frag_interleave != 0, mov->mdat_buf != NULL, because
  
  it contains audio data. So avio_write is called before write_moof,
  
  that is, the data write before moof, and mov_finish_fragment
  
  executed with wrong mdat_start.
  4. With normal fmp4 output, the error isn't obvious. With
  
  hybrid_fragmented, ffplay output.mp4 shows a lot of error messages.
  Signed-off-by : Zhao Zhili <zhilizhao@tencent.com>
  

  • [DH] libavformat/movenc.c
  • [DH] tests/fate/mov.mak

• Hello ! I have a little problem with my discord.py music bot

  5 mars 2021, par Luca M. Schmidt

  i'm rather new/unexpierience with youtube-dl and python. The thing i'm triying to do is to add a queue system to my music cog. When possible it should be able to add songs and start the next song after the first one ended as well. You dont need to provide the complete code for it, tryingt to explain how it should work and giving some tipps should be enought. If more information is needed, feel free to ask. Thx for helping.

  


  My Code (Stripped down a bit) :

  


  # IMPORT


import discord
from discord.ext import commands
import json
import asyncio
import youtube_dl



# LOKALE_VARIABLEN


ytdl_format_options = {
    'format': 'bestaudio/best',
    'outtmpl': '%(extractor)s-%(id)s-%(title)s.%(ext)s',
    'restrictfilenames': True,
    'noplaylist': True,
    'nocheckcertificate': True,
    'ignoreerrors': False,
    'logtostderr': False,
    'quiet': True,
    'no_warnings': True,
    'default_search': 'auto',
    'source_address': '0.0.0.0'
}

ffmpeg_options = {
    'options': '-vn'
}

ytdl = youtube_dl.YoutubeDL(ytdl_format_options)

songs = asyncio.Queue()
play_next_song = asyncio.Event()


# ----

class YTDLSource(discord.PCMVolumeTransformer):
    def __init__(self, source, *, data, volume=0.5):
        super().__init__(source, volume)

        self.data = data

        self.title = data.get('title')
        self.url = data.get('url')
        self.thumbnail = data.get('thumbnail')


    @classmethod
    async def from_url(cls, url, *, loop=None, stream=False):
        loop = loop or asyncio.get_event_loop()
        data = await loop.run_in_executor(None, lambda: ytdl.extract_info(url, download=not stream))

        if 'entries' in data:
            data = data['entries'][0]

        filename = data['url'] if stream else ytdl.prepare_filename(data)
        return cls(discord.FFmpegPCMAudio(filename, **ffmpeg_options), data=data)


# COG_SETUP(START)


class Music(commands.Cog):

    def __init__(self, client):
        self.client = client


    @commands.command()
    async def join(self, ctx, *, channel: discord.VoiceChannel):

        if ctx.voice_client is not None:
            return await ctx.voice_client.move_to(channel)

        await channel.connect()

    @commands.command()
    @commands.cooldown(1, 10, commands.BucketType.user)
    async def play(self, ctx, *, url):

        try:

            async with ctx.typing():
                player = await YTDLSource.from_url(url, loop=self.client.loop, stream=True)
            ctx.voice_client.play(player, after=lambda e: print('Player error: %s' % e) if e else None)

        except:
            embed = discord.Embed(
                title='Fehler!',
                colour=discord.Colour.red(),
                description='Dies ist eine nicht Unterstützte URL!'
            )

            return await ctx.send(embed=embed)

        embed = discord.Embed(
            title='',
            colour=discord.Colour.blue(),
            description=f'[{format(player.title)}]({player.url})'
        )
        embed.set_author(name='Spielt gerade:')
        embed.set_image(url=player.thumbnail)
        embed.set_footer(text=f'Hinzugefügt von: {ctx.author.name}', icon_url=ctx.author.avatar_url)


        await ctx.send(embed=embed)




    @commands.command()
    async def volume(self, ctx, volume: int):

        if ctx.voice_client is None:

            embed = discord.Embed(
                title='Fehler!',
                colour=discord.Colour.red(),
                description='Ich bin mit keinem Sprachkanal verbunden!'
            )

            return await ctx.send(embed=embed)

        elif ctx.voice_client is not None:

            if volume in range(0, 201):
                try:
                    ctx.voice_client.source.volume = volume / 100

                    embed = discord.Embed(
                        title='Lautstärke',
                        colour=discord.Colour.blue(),
                        description=f'Lautstärke auf **{format(volume)}**% gestellt.'
                    )
                    embed.set_footer(text=f"Angepasst von: {ctx.author.name}", icon_url=ctx.author.avatar_url)

                    return await ctx.send(embed=embed)
                except:
                    pass

            else:

                embed = discord.Embed(
                    title='Fehler!',
                    colour=discord.Colour.red(),
                    description='Das ist Lauter, als die Musik geht!'
                )

                return await ctx.send(embed=embed)

    @commands.command()
    async def stop(self, ctx):
        try:

            await ctx.voice_client.disconnect()
            await ctx.message.delete()

        except:
            pass

    @commands.command()
    async def pause(self, ctx):

        if ctx.voice_client.is_playing():

            ctx.voice_client.pause()
            await ctx.message.delete()
            return

        else:

            embed = discord.Embed(
                title='Fehler!',
                colour=discord.Colour.red(),
                description='Es spielt keine Musik!'
            )

            return await ctx.send(embed=embed)

    @commands.command()
    async def resume(self, ctx):

        if ctx.voice_client.is_paused():

            ctx.voice_client.resume()
            await ctx.message.delete()
            return

        else:

            embed = discord.Embed(
                title='Fehler!',
                colour=discord.Colour.red(),
                description='Es wurde keine Musik pausiert, darum kann ich auch nichts fortsetzen!'
            )

            return await ctx.send(embed=embed)

    @resume.before_invoke
    @play.before_invoke
    async def ensure_voice(self, ctx):
        if ctx.voice_client is None:
            if ctx.author.voice:
                try:

                    await ctx.author.voice.channel.connect()

                except commands.CommandError:
                    embed = discord.Embed(
                        title='Fehler!',
                        colour=discord.Colour.red(),
                        description='Du bist nicht mit einem Sprachkanal verbunden!'
                    )

                    await ctx.send(embed=embed)

        elif ctx.voice_client.is_playing():
            ctx.voice_client.stop()


# COG_SETUP(END)


def setup(client):
    client.add_cog(Music(client))