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• Cortex-A7 instruction cycle timings

  15 mai 2014, par Mans — ARM

  The Cortex-A7 ARM core is a popular choice in low-power and low-cost designs. Unfortunately, the public TRM does not include instruction timing information. It does reveal that execution is in-order which makes measuring the throughput and latency for individual instructions relatively straight-forward. The table below lists the measured issue cycles … Continue reading →

• Correct command to transmit audio to ip camera using ffmpeg ?

  4 novembre 2016, par the_naive

  So I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.

  The command I’m using is the following :

  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
  
  ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -report

  The ouput I get following this command is the following :

  ffmpeg started on 2016-11-04 at 17:32:13
  
  Report written to "ffmpeg-20161104-173213.log"
  
  Command line:
  
  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
  
  ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.4.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 35.100 / 55. 35.100
  
    libavcodec     57. 66.101 / 57. 66.101
  
    libavformat    57. 57.100 / 57. 57.100
  
    libavdevice    57.  2.100 / 57.  2.100
  
    libavfilter     6. 66.100 /  6. 66.100
  
    libswscale      4.  3.100 /  4.  3.100
  
    libswresample   2.  4.100 /  2.  4.100
  
    libpostproc    54.  2.100 / 54.  2.100
  
  Splitting the commandline.
  
  Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
  
  Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
  
  Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
  
  Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
  
  Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
  
  Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
  
  Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
  
  Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
  
  Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
  
  Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
  
  Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
  
  Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
  
  Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
  
  Trailing options were found on the commandline.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option v (set logging level) with argument debug.
  
  Applying option y (overwrite output files) with argument 1.
  
  Applying option report (generate a report) with argument 1.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
  
  Applying option re (read input at native frame rate) with argument 1.
  
  Applying option f (force format) with argument dshow.
  
  Successfully parsed a group of options.
  
  Opening an input file: audio=Microphone (2- High Definition Audio Device).
  
  [dshow @ 00000000000279e0] Selecting pin Capture on audio only
  
  dshow passing through packet of type audio size    88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
  
  [dshow @ 00000000000279e0] All info found
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
  
    Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
  
      Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  Applying option c:a (codec name) with argument pcm_mulaw.
  
  Applying option ac (set number of audio channels) with argument 1.
  
  Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
  
  Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
  
  Applying option f (force format) with argument flv.
  
  Successfully parsed a group of options.
  
  Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  [http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Expect: 100-continue
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  
  
  
  
  
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001
  
  
  
  
  
  
  
  
  
  Successfully opened the file.
  
  detected 8 logical cores
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
  
  [AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000 
  
  [auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
  
  Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
  
    Metadata:
  
      encoder         : Lavf57.57.100
  
      Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc57.66.101 pcm_mulaw
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
  
  Press [q] to stop, [?] for help
  
  cur_dts is invalid (this is harmless if it occurs once at the start per stream)
  
  av_interleaved_write_frame(): Unknown error
  
  No more output streams to write to, finishing.
  
  Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize=       8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x    
  
  video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
  
  Input file #0 (audio=Microphone (2- High Definition Audio Device)):
  
    Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples); 
  
    Total: 1 packets (88200 bytes) demuxed
  
  Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
  
    Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes); 
  
    Total: 1 packets (7984 bytes) muxed
  
  1 frames successfully decoded, 0 decoding errors
  
  [AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
  
  dshow passing through packet of type audio size    12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
  
  Conversion failed!

  For some reason, despite setting multiple_requests, reconnect_eof, reconnect_streamed all to true, connection becomes closed.

  Could you please tell me what I’m doing wrong ?

• Decoding audio w/ ffmpeg error on Android

  14 août 2012, par stranded

  Well, I knew I was going out of my comfort zone when I decided to try and decode audio using ffmpeg on Android but now I will have to admit that I'm stranded.
  It took me many days to just build ffmpeg for Android. Roman's10 guide did not work for me but finally things started looking up, thanks to this tutorial. So because of Dmitry's help I managed to build the armeabi version (not armeabi-v7) for my phone (LG P500) and everything basic works.

  But when I try to use avcodec_decode_audio3() things go downhill :( Never before have I felt so close to making things work (after all it seems to be only one line that is troublesome)
  but unable to though. I've read many questions here on SO that have brought me closer to the goal. Googling, on the other hand, has had limited results - making questions here the only fruit.

  Yes, I know ! I ramble. But I can't help it, I'm only trying to explain in detail where I'm stuck and how I got there. So without further ado I bring you the code :

  NATIVE CODE :

  #include 
  
  #include <android></android>log.h>
  
  
  
  #include "libavcodec/avcodec.h"
  
  #include "libavformat/avformat.h"
  
  
  
  #define LOG_TAG "mylib"
  
  #define LOGI(...)  __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
  
  #define LOGE(...)  __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
  
  
  
  #define INBUFF_SIZE 4096
  
  #define AUDIO_INBUFF 20480
  
  #define AUDIO_REFILL_SIZE 4096
  
  
  
  jint Java_com_nothingworks_for_me_MainActivity_decode(JNIEnv * env, jobject this, jstring jfilename){
  
      const char *filename = (*env)->GetStringUTFChars(env, jfilename, NULL);
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int audioStream;
  
      int out_size, len, i;
  
      FILE *f, *outfile;
  
      uint8_t *outbuf;
  
      uint8_t inbuf[AUDIO_INBUFF + FF_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFormatContext *pFormatCtx;
  
  
  
      av_register_all();
  
      avcodec_init();
  
      av_init_packet(&amp;avpkt);
  
  
  
      if(av_open_input_file(&amp;pFormatCtx, filename, NULL, 0, NULL)!=0)
  
      {
  
          LOGE("Can&#39;t open file &#39;%s&#39;\n", filename);
  
          return 1;
  
      }
  
      else
  
      {
  
          LOGI("File was opened\n");
  
          LOGI("File &#39;%s&#39;, Codec %s",
  
                  pFormatCtx->filename,
  
                  pFormatCtx->iformat->name
  
          );
  
      }
  
  
  
      if (av_find_stream_info(pFormatCtx) &lt; 0){
  
          LOGE("Can&#39;t find stream info");
  
      }
  
      audioStream = -1;
  
      for (i = 0; i &lt; pFormatCtx->nb_streams; i++) {
  
          if (pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) {
  
              audioStream = i;
  
              break;
  
          }
  
      }
  
  
  
      if (audioStream == -1) {
  
          LOGE("Didn&#39;t find stream!");
  
      }
  
      c = pFormatCtx->streams[audioStream]->codec;
  
  
  
      codec = avcodec_find_decoder(c->codec_id);
  
      if (!codec) {
  
          LOGE("Unsupported Codec!");
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      /* open it */
  
      if (avcodec_open(c, codec) &lt; 0) {
  
          LOGE("Can&#39;t open codec");
  
          exit(1);
  
      }
  
  
  
      outbuf = av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE * 2);
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          LOGE("Can&#39;t open file");
  
          exit(1);
  
      }
  
  
  
      /* decode until eof */
  
      avpkt.data = inbuf;
  
      avpkt.size = fread(inbuf, 1, AUDIO_INBUFF, f);
  
      LOGI("avpkt.size %d", avpkt.size);
  
  
  
      while (avpkt.size > 0) {
  
          out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 2; 

  THINGS GO WRONG HERE ! avcodec_decode_audio3() The code continues from ▲ to ▼ :

          len = avcodec_decode_audio3(c, (int16_t *)outbuf, &out_size, &avpkt);
  
          LOGI("data_size %d len %d", out_size, len);
  
          if (len < 0) {
  
              LOGE("Error while decoding");
  
              exit(1);
  
          }
  
          if (out_size > 0) {
  
  
  
          }
  
          avpkt.size -= len;
  
          avpkt.data += len;
  
          if (avpkt.size < AUDIO_REFILL_SIZE) {
  
              /* Refill the input buffer, to avoid trying to decode
  
               * incomplete frames. Instead of this, one could also use
  
               * a parser, or use a proper container format through
  
               * libavformat. */
  
              memmove(inbuf, avpkt.data, avpkt.size);
  
              avpkt.data = inbuf;
  
              len = fread(avpkt.data + avpkt.size, 1,
  
                      AUDIO_INBUFF - avpkt.size, f);
  
              if (len > 0)
  
                  avpkt.size += len;
  
          }
  
      }
  
      fclose(f);
  
      free(outbuf);
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
      return 0;
  
  }

  ---

  What happens is that avcodec_decode_audio3() returns -1 and that's pretty much it :(
  I have no idea what to do next. I can't find much info about this and I only started fiddling with C less than two weeks ago so your guidance is my only hope now [play dramatic sound]. Hope someone can shed a little light on this mystery.

  Ohh ! And the native code is some kind of a hybrid between what I have found here on SO, like this and this, and the ffmpeg example. On the java side I only have a call to this native method and pass it string which is the path to a MP3 song on my droid. I don't use AudioTrack or anything else in my java code yet 'cause I'm only trying to get the decoding to work for now.

  -Drama Queen OUT !