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• Iterate through URLs in bash

  16 février 2016, par cclloyd

  If I wanted to use ffmpeg to download a bunch of .ts files from a website, and the url format is

  http://example.com/video-1080Pxxxxx.ts

  Where the ’x’ is a number from 00000 to 99999 (required 5 zeros), how would I iterate through that in bash so that it tries every integer starting at 00000, 00001, 00002, etc.?

• Simply beyond ridiculous

  7 mai 2010, par Dark Shikari — H.265, speed

  For the past few years, various improvements on H.264 have been periodically proposed, ranging from larger transforms to better intra prediction. These finally came together in the JCT-VC meeting this past April, where over two dozen proposals were made for a next-generation video coding standard. Of course, all of these were in very rough-draft form ; it will likely take years to filter it down into a usable standard. In the process, they’ll pick the most useful features (hopefully) from each proposal and combine them into something a bit more sane. But, of course, it all has to start somewhere.

  A number of features were common : larger block sizes, larger transform sizes, fancier interpolation filters, improved intra prediction schemes, improved motion vector prediction, increased internal bit depth, new entropy coding schemes, and so forth. A lot of these are potentially quite promising and resolve a lot of complaints I’ve had about H.264, so I decided to try out the proposal that appeared the most interesting : the Samsung+BBC proposal (A124), which claims compression improvements of around 40%.

  The proposal combines a bouillabaisse of new features, ranging from a 12-tap interpolation filter to 12thpel motion compensation and transforms as large as 64×64. Overall, I would say it’s a good proposal and I don’t doubt their results given the sheer volume of useful features they’ve dumped into it. I was a bit worried about complexity, however, as 12-tap interpolation filters don’t exactly scream “fast”.

  I prepared myself for the slowness of an unoptimized encoder implementation, compiled their tool, and started a test encode with their recommended settings.

  I waited. The first frame, an I-frame, completed.

  I took a nap.

  I waited. The second frame, a P-frame, was done.

  I played a game of Settlers.

  I waited. The third frame, a B-frame, was done.

  I worked on a term paper.

  I waited. The fourth frame, a B-frame, was done.

  After a full 6 hours, 8 frames had encoded. Yes, at this rate, it would take a full two weeks to encode 10 seconds of HD video. On a Core i7. This is not merely slow ; this is over 1000 times slower than x264 on “placebo” mode. This is so slow that it is not merely impractical ; it is impossible to even test. This encoder is apparently designed for some sort of hypothetical future computer from space. And word from other developers is that the Intel proposal is even slower.

  This has led me to suspect that there is a great deal of cheating going on in the H.265 proposals. The goal of the proposals, of course, is to pick the best feature set for the next generation video compression standard. But there is an extra motivation : organizations whose features get accepted get patents on the resulting standard, and thus income. With such large sums of money in the picture, dishonesty becomes all the more profitable.

  There is a set of rules, of course, to limit how the proposals can optimize their encoders. If different encoders use different optimization techniques, the results will no longer be comparable — remember, they are trying to compare compression features, not methods of optimizing encoder-side decisions. Thus all encoders are required to use a constant quantizer, specified frame types, and so forth. But there are no limits on how slow an encoder can be or what algorithms it can use.

  It would be one thing if the proposed encoder was a mere 10 times slower than the current reference ; that would be reasonable, given the low level of optimization and higher complexity of the new standard. But this is beyond ridiculous. With the prize given to whoever can eke out the most PSNR at a given quantizer at the lowest bitrate (with no limits on speed), we’re just going to get an arms race of slow encoders, with every company trying to use the most ridiculous optimizations possible, even if they involve encoding the frame 100,000 times over to choose the optimal parameters. And the end result will be as I encountered here : encoders so slow that they are simply impossible to even test.

  Such an arms race certainly does little good in optimizing for reality where we don’t have 30 years to encode an HD movie : a feature that gives great compression improvements is useless if it’s impossible to optimize for in a reasonable amount of time. Certainly once the standard is finalized practical encoders will be written — but it makes no sense to optimize the standard for a use-case that doesn’t exist. And even attempting to “optimize” anything is difficult when encoding a few seconds of video takes weeks.

  Update : The people involved have contacted me and insist that there was in fact no cheating going on. This is probably correct ; the problem appears to be that the rules that were set out were simply not strict enough, making many changes that I would intuitively consider “cheating” to be perfectly allowed, and thus everyone can do it.

  I would like to apologize if I implied that the results weren’t valid ; they are — the Samsung-BBC proposal is definitely one of the best, which is why I picked it to test with. It’s just that I think any situation in which it’s impossible to test your own software is unreasonable, and thus the entire situation is an inherently broken one, given the lax rules, slow baseline encoder, and no restrictions on compute time.

• Decoding audio via Android using FFMpeg

  31 octobre 2013, par Rob Haupt

  I can play Wav files using the below code without issues. When trying to play the exact same media in Mp3 format I only get garbled junk. I believe I am fundamentally misunderstanding how the avcodec_decode_audio3 function works.

  Since the Wav file contains PCM data when it is decoded it can go straight to the AudioTrack.write function. There must be some additional step to get Mp3s to work like this. I don't know what I'm missing, but I've been pulling my hair out for a week now.

  Java Code

  package com.rohaupt.RRD2;
  
  
  
  import java.io.FileNotFoundException;
  
  import java.io.FileOutputStream;
  
  import java.io.IOException;
  
  
  
  import android.app.Activity;
  
  import android.media.AudioFormat;
  
  import android.media.AudioManager;
  
  import android.media.AudioTrack;
  
  import android.media.MediaPlayer;
  
  import android.os.Bundle;
  
  import android.os.SystemClock;
  
  
  
  public class player extends Activity
  
  {
  
      private AudioTrack track;
  
      private FileOutputStream os;
  
      /** Called when the activity is first created. */
  
      @Override
  
      public void onCreate(Bundle savedInstanceState)
  
      {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.main);
  
          createEngine();
  
  
  
          MediaPlayer mp = new MediaPlayer();
  
          mp.start();
  
  
  
          int bufSize = AudioTrack.getMinBufferSize(32000,
  
                                                    AudioFormat.CHANNEL_CONFIGURATION_STEREO, 
  
                                                    AudioFormat.ENCODING_PCM_16BIT);
  
  
  
  
  
          track = new AudioTrack(AudioManager.STREAM_MUSIC, 
  
                                 32000, 
  
                                 AudioFormat.CHANNEL_CONFIGURATION_STEREO, 
  
                                 AudioFormat.ENCODING_PCM_16BIT, 
  
                                 bufSize,
  
                                 AudioTrack.MODE_STREAM);
  
  
  
          byte[] bytes = new byte[bufSize];
  
  
  
          try {
  
              os = new FileOutputStream("/sdcard/a.out",false);
  
          } catch (FileNotFoundException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
  
  
          String result = loadFile("/sdcard/a.mp3",bytes);
  
  
  
          try {
  
              os.close();
  
          } catch (IOException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
      }
  
  
  
      void playSound(byte[] buf, int size) {  
  
          //android.util.Log.v("ROHAUPT", "RAH Playing");
  
          if(track.getPlayState()!=AudioTrack.PLAYSTATE_PLAYING)
  
              track.play();
  
          track.write(buf, 0, size);
  
  
  
          try {
  
              os.write(buf,0,size);
  
          } catch (IOException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
      }
  
  
  
  
  
      private native void createEngine();
  
      private native String loadFile(String file, byte[] array);
  
  
  
      /** Load jni .so on initialization*/ 
  
      static {
  
           System.loadLibrary("avutil"); 
  
           System.loadLibrary("avcore"); 
  
           System.loadLibrary("avcodec");
  
           System.loadLibrary("avformat");
  
           System.loadLibrary("avdevice");
  
           System.loadLibrary("swscale");
  
           System.loadLibrary("avfilter");
  
           System.loadLibrary("ffmpeg");
  
      }
  
  }

  C Code

  #include 
  
  #include 
  
  #include 
  
  #include <android></android>log.h>
  
  
  
  #include "libavcodec/avcodec.h"
  
  #include "libavformat/avformat.h"
  
  
  
  #define DEBUG_TAG "ROHAUPT"  
  
  
  
  void Java_com_rohaupt_RRD2_player_createEngine(JNIEnv* env, jclass clazz)
  
      {
  
          avcodec_init();
  
  
  
          av_register_all();
  
  
  
  
  
      }
  
  
  
      jstring Java_com_rohaupt_RRD2_player_loadFile(JNIEnv* env, jobject obj,jstring file,jbyteArray array)
  
      {   
  
          jboolean            isCopy;  
  
          int                 i;
  
          int                 audioStream=-1;
  
          int                 res;
  
          int                 decoded = 0;
  
          int                 out_size;
  
          AVFormatContext     *pFormatCtx;
  
          AVCodecContext      *aCodecCtx;
  
          AVCodecContext      *c= NULL;
  
          AVCodec             *aCodec;
  
          AVPacket            packet;
  
          jclass              cls = (*env)->GetObjectClass(env, obj);
  
          jmethodID           play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");//At the begining of your main function
  
          const char *        szfile = (*env)->GetStringUTFChars(env, file, &amp;isCopy);
  
          int16_t *           pAudioBuffer = (int16_t *) av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE);
  
          int16_t *           outBuffer = (int16_t *) av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE);
  
  
  
  
  
          __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG, "RAH28 Starting");
  
          res = av_open_input_file(&amp;pFormatCtx, szfile, NULL, 0, NULL);
  
          if(res!=0)
  
          {
  
              __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH opening input failed with result: [%d]", res);
  
              return file;
  
          }
  
  
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting stream info");
  
          res = av_find_stream_info(pFormatCtx);
  
          if(res&lt;0)
  
          {
  
              __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting stream info failed with result: [%d]", res);
  
              return file;
  
          }
  
  
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting audio stream");
  
          for(i=0; i &lt; pFormatCtx->nb_streams; i++) {
  
            if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &amp;&amp;
  
               audioStream &lt; 0) {
  
              audioStream=i;
  
            }
  
          }
  
  
  
  
  
          if(audioStream==-1)
  
          {
  
              __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH couldn&#39;t find audio stream");
  
              return file;
  
          }
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio stream found with result: [%d]", res);
  
  
  
  
  
          aCodecCtx=pFormatCtx->streams[audioStream]->codec;
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info loaded");
  
  
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info [%d]", aCodecCtx->codec_id);
  
  
  
          aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
  
          if(!aCodec) {
  
             __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec unsupported");
  
             return file;
  
          }
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info found");
  
  
  
  
  
          res = avcodec_open(aCodecCtx, aCodec);
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec loaded [%d] [%d]",aCodecCtx->sample_fmt,res);
  
  
  
          //c=avcodec_alloc_context();
  
          av_init_packet(&amp;packet);
  
  
  
  
  
          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH channels [%d] sample rate [%d] sample format [%d]",aCodecCtx->channels,aCodecCtx->sample_rate,aCodecCtx->sample_fmt);
  
  
  
  
  
          int x,y;
  
          x=0;y=0;
  
          while (av_read_frame(pFormatCtx, &amp;packet)>= 0) {
  
              __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH frame read: [%d] [%d]",x++,y);
  
  
  
              if (aCodecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio ready");
  
                          int data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE;
  
                          int size=packet.size;
  
                          y=0;
  
                          decoded = 0;
  
                          __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH packet size: [%d]", size);
  
                          while(size > 0) {
  
  
  
                                  __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH decoding: [%d] [%d]",x,y++);
  
                                  int len = avcodec_decode_audio3(aCodecCtx, pAudioBuffer, &amp;data_size, &amp;packet);
  
  
  
  
  
  
  
                                  __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 1 size [%d] len [%d] data_size [%d] out_size [%d]",size,len,data_size,out_size);
  
                                  jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
  
  
                                  memcpy(bytes + decoded, pAudioBuffer, len); //
  
  
  
  
  
                                  __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 2");
  
  
  
                                  (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                                  __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 3");
  
  
  
                                  (*env)->CallVoidMethod(env, obj, play, array, len);
  
  
  
                                  __android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 4");
  
  
  
  
  
                                  size -= len;
  
                                  decoded += len;
  
  
  
                         }
  
                         av_free_packet(&amp;packet);
  
              }
  
  
  
       }
  
  
  
            // Close the video file
  
          av_close_input_file(pFormatCtx);
  
  
  
          //__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH Finished Running result: [%d]", res);
  
          (*env)->ReleaseStringUTFChars(env, file, szfile);   
  
          return file;  
  
      }

  To add some detail. When Calling this function with a Wav File I get the following Log Data

  I/ROHAUPT (  227): RAH28 Starting
  
  D/ROHAUPT (  227): RAH getting stream info
  
  D/ROHAUPT (  227): RAH getting audio stream
  
  D/ROHAUPT (  227): RAH audio stream found with result: [0]
  
  D/ROHAUPT (  227): RAH audio codec info loaded
  
  D/ROHAUPT (  227): RAH audio codec info [65536]
  
  D/ROHAUPT (  227): RAH audio codec info found
  
  D/ROHAUPT (  227): RAH audio codec loaded [1] [0]
  
  D/ROHAUPT (  227): RAH channels [2] sample rate [32000] sample format [1]
  
  D/ROHAUPT (  227): RAH frame read: [0] [0]
  
  D/ROHAUPT (  227): RAH audio ready
  
  D/ROHAUPT (  227): RAH packet size: [4096]
  
  D/ROHAUPT (  227): RAH decoding: [1] [0]
  
  D/ROHAUPT (  227): RAH 1 size [4096] len [4096] data_size [4096] out_size [0]
  
  D/ROHAUPT (  227): RAH 2
  
  D/ROHAUPT (  227): RAH 3
  
  D/ROHAUPT (  227): RAH 4
  
  D/ROHAUPT (  227): RAH frame read: [1] [1]
  
  D/ROHAUPT (  227): RAH audio ready
  
  ...
  
  D/ROHAUPT (  227): RAH frame read: [924] [1]
  
  D/ROHAUPT (  227): RAH audio ready
  
  D/ROHAUPT (  227): RAH packet size: [4096]
  
  D/ROHAUPT (  227): RAH decoding: [925] [0]
  
  D/ROHAUPT (  227): RAH 1 size [4096] len [4096] data_size [4096] out_size [0]
  
  D/ROHAUPT (  227): RAH 2
  
  D/ROHAUPT (  227): RAH 3
  
  D/ROHAUPT (  227): RAH 4
  
  D/ROHAUPT (  227): RAH frame read: [925] [1]
  
  D/ROHAUPT (  227): RAH audio ready
  
  D/ROHAUPT (  227): RAH packet size: [3584]
  
  D/ROHAUPT (  227): RAH decoding: [926] [0]
  
  D/ROHAUPT (  227): RAH 1 size [3584] len [3584] data_size [3584] out_size [0]
  
  D/ROHAUPT (  227): RAH 2
  
  D/ROHAUPT (  227): RAH 3
  
  D/ROHAUPT (  227): RAH 4

  When calling with an Mp3 file I get the following

  I/ROHAUPT (  280): RAH28 Starting
  
  D/ROHAUPT (  280): RAH getting stream info
  
  D/ROHAUPT (  280): RAH getting audio stream
  
  D/ROHAUPT (  280): RAH audio stream found with result: [0]
  
  D/ROHAUPT (  280): RAH audio codec info loaded
  
  D/ROHAUPT (  280): RAH audio codec info [86017]
  
  D/ROHAUPT (  280): RAH audio codec info found
  
  D/ROHAUPT (  280): RAH audio codec loaded [1] [0]
  
  D/ROHAUPT (  280): RAH channels [2] sample rate [32000] sample format [1]
  
  D/ROHAUPT (  280): RAH frame read: [0] [0]
  
  D/ROHAUPT (  280): RAH audio ready
  
  D/ROHAUPT (  280): RAH packet size: [432]
  
  D/ROHAUPT (  280): RAH decoding: [1] [0]
  
  D/ROHAUPT (  280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
  
  D/ROHAUPT (  280): RAH 2
  
  ...
  
  D/ROHAUPT (  280): RAH frame read: [822] [1]
  
  D/ROHAUPT (  280): RAH audio ready
  
  D/ROHAUPT (  280): RAH packet size: [432]
  
  D/ROHAUPT (  280): RAH decoding: [823] [0]
  
  D/ROHAUPT (  280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
  
  D/ROHAUPT (  280): RAH 2
  
  D/ROHAUPT (  280): RAH 3
  
  D/ROHAUPT (  280): RAH 4
  
  D/ROHAUPT (  280): RAH frame read: [823] [1]
  
  D/ROHAUPT (  280): RAH audio ready
  
  D/ROHAUPT (  280): RAH packet size: [432]
  
  D/ROHAUPT (  280): RAH decoding: [824] [0]
  
  D/ROHAUPT (  280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
  
  D/ROHAUPT (  280): RAH 2
  
  D/ROHAUPT (  280): RAH 3
  
  D/ROHAUPT (  280): RAH 4