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• continuous record/recognize audio with pocketsphinx/ffmpeg

  10 novembre 2017, par Michael

  as the title already says, I want to continuous record raw audio through my microphone.
  So the idea was running a simple C program in the background as service that would create chunks of audio and send those files through the sphinx speech recognition.

  After that I can do some processing with the recognized words.

  The problem is the (continuous) recognition. I can’t just record audio chunks containing 10 seconds what i’ve said, because maybe chunk[33] -> chunk[34] belong together and then sphinx would output something like :

  recognized chunk[33] -> ["enable light"]
  
  recognized chunk[34] -> ["5 with 50 percent"]

  Another approach would be to continuous record audio but then I can’t process big audio files with sphinx.

  I’m using the basic example from pocketsphinx :

  #include 
  
  
  
  int main(int argc, char *argv[])
  
  {
  
  ps_decoder_t *ps;
  
  cmd_ln_t *config;
  
  FILE *fh;
  
  char const *hyp, *uttid;
  
  int16 buf[512];
  
  int rv;
  
  int32 score;
  
  
  
  config = cmd_ln_init(NULL, ps_args(), TRUE,
  
               "-hmm", MODELDIR "/en-us/en-us",
  
               "-lm", MODELDIR "/en-us/en-us.lm.bin",
  
               "-dict", MODELDIR "/en-us/cmudict-en-us.dict",
  
               NULL);
  
  if (config == NULL) {
  
  fprintf(stderr, "Failed to create config object, see log for details\n");
  
  return -1;
  
  }
  
  
  
  ps = ps_init(config);
  
  if (ps == NULL) {
  
  fprintf(stderr, "Failed to create recognizer, see log for details\n");
  
  return -1;
  
  }
  
  
  
  fh = fopen("audiochunk_33.raw", "rb");
  
  if (fh == NULL) {
  
  fprintf(stderr, "Unable to open input file goforward.raw\n");
  
  return -1;
  
  }
  
  
  
  rv = ps_start_utt(ps);
  
  
  
  while (!feof(fh)) {
  
  size_t nsamp;
  
  nsamp = fread(buf, 2, 512, fh);
  
  rv = ps_process_raw(ps, buf, nsamp, FALSE, FALSE);
  
  }
  
  
  
  rv = ps_end_utt(ps);
  
  hyp = ps_get_hyp(ps, &score);
  
  printf("Recognized: %s\n", hyp);
  
  
  
  fclose(fh);
  
  ps_free(ps);
  
  cmd_ln_free_r(config);
  
  
  
  return 0;

  }

  And here is a basic example using ffmpeg to create a simple audio file/chunk :

  #include 
  
  #include 
  
  #include 
  
  
  
  #define N 44100
  
  
  
  void main()
  
  {
  
  // Create audio buffer
  
  int16_t buf[N] = {0}; // buffer
  
  int n;                // buffer index
  
  double Fs = 44100.0;  // sampling frequency
  
  
  
  // Generate 1 second of audio data - it's just a 1 kHz sine wave
  
  for (n=0 ; n<n></n>Fs);
  
  
  
  // Pipe the audio data to ffmpeg, which writes it to a wav file
  
  FILE *pipeout;
  
  pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i - beep.wav", "w");
  
  fwrite(buf, 2, N, pipeout);
  
  pclose(pipeout);
  
  }

  BR
  Michael

• How to force specific AVInputFormat in code (FFMPEG) ?

  18 février 2020, par kugipark

  ++Plaease understand that maybe some words or sentences could not be correct English.++

  I’m novice programmer and developing a video player app for Android which can play 5ch .avi video from http. It based on ffplay.c in FFMPEG libarary.

  Currently I have 2 problems for this below. It has occured only in android devices.

  ---

  1) Too much time taken for detecting format.

  • opening and finding stream info takes more than a minute when I trying to open 5ch video source whereas normal .mp4 (h264) source open almost immediately.

  2) Demuxing is too slow when it comes to large size video.

  • If the resolution of video getting larger, then displaying frame rate getting slower even though there are enough memory, network and CPU resources physically.

  To resolve my problem, I tried to force the format and decoders but I couldn’t found some information for specifying the input source in code level.
  The official doc site only refers to a sentence about those parameters like "If non-NULL, this parameter forces a specific input format. Otherwise the format is autodetected.". So I don’t have any clues how to set the number of streams, decoders, decoders’ private options, and etc. (And also which parameters should I manage.) If someone knows how to set the options(like AVDictionary), and pass to the av functions, please let me know an exmple. The source contains 2 video streams, 1 audio stream, and 2 more extra streams (for custom data like gps). The stream information of video is below. I printed it manually, and these are auto-detected information.

      ---------- File format information ----------
  
  flags=2097152
  
  video_codec_id=0 (NONE)
  
  audio_codec_id=0 (NONE)
  
  ctx_flags=0
  
  data_codec_id=0 (NONE)
  
  format_whitelist=(null)
  
  iformat=0xa19d0d2c
  
  ---------- Stream information ----------
  
  stream 1 of 5:
  
  ----- common ----------
  
  bit_rate: 11383235
  
  bits_per_coded_sample: 24
  
  bits_per_raw_sample: 0
  
  codec_id: 0x1C (H264)
  
  codec_tag: 875967048
  
  extradata_size: 0
  
  level: -99
  
  profile: -99
  
  sample_rate: 0
  
  ----- Video Stream ----------
  
  chroma_location: 0
  
  color_primaries: 2
  
  color_space: 2
  
  color_trc: 2
  
  field_order: 0
  
  format: -1 (NONE)
  
  height: 1080
  
  width: 1920
  
  sample_aspect_ratio.den: 1
  
  sample_aspect_ratio.num: 0
  
  video_delay: 0
  
  ----------------------------------------
  
  stream 2 of 5:
  
  ----- common ----------
  
  bit_rate: 6185438
  
  bits_per_coded_sample: 24
  
  bits_per_raw_sample: 0
  
  codec_id: 0x1C (H264)
  
  codec_tag: 875967048
  
  extradata_size: 0
  
  level: -99
  
  profile: -99
  
  sample_rate: 0
  
  ----- Video Stream ----------
  
  chroma_location: 0
  
  color_primaries: 2
  
  color_space: 2
  
  color_trc: 2
  
  field_order: 0
  
  format: -1 (NONE)
  
  height: 720
  
  width: 1280
  
  sample_aspect_ratio.den: 1
  
  sample_aspect_ratio.num: 0
  
  video_delay: 0
  
  ----------------------------------------
  
  stream 3 of 5:
  
  ----- common ----------
  
  bit_rate: 352800
  
  bits_per_coded_sample: 16
  
  bits_per_raw_sample: 0
  
  codec_id: 0x10000 (PCM_S16LE)
  
  codec_tag: 1
  
  extradata_size: 0
  
  level: -99
  
  profile: -99
  
  sample_rate: 22050
  
  ----- Audio Stream ----------
  
  block_align: 2
  
  channels: 1
  
  channel_layout: 0
  
  color_range: 0
  
  frame_size: 0
  
  initial_padding: 0
  
  seek_preroll: 0
  
  trailing_padding: 0
  
  ----------------------------------------
  
  stream 4 of 5:
  
  ----- common ----------
  
  bit_rate: 15625
  
  bits_per_coded_sample: 0
  
  bits_per_raw_sample: 0
  
  codec_id: 0x0 (NONE)
  
  codec_tag: 0
  
  extradata_size: 0
  
  level: -99
  
  profile: -99
  
  sample_rate: 0
  
  ----- Subtitle Stream ----------
  
  ----------------------------------------
  
  stream 5 of 5:
  
  ----- common ----------
  
  bit_rate: 33862
  
  bits_per_coded_sample: 0
  
  bits_per_raw_sample: 0
  
  codec_id: 0x0 (NONE)
  
  codec_tag: 0
  
  extradata_size: 0
  
  level: -99
  
  profile: -99
  
  sample_rate: 0
  
  ----- Subtitle Stream ----------

• I want to convert images and sounds into video using php [closed]

  6 janvier 2023, par hossein ahmadi

  I have 5 pictures that contain 5 English words
Also, I have five audio files of pronunciation of the words above.

  


  I want to write a program with php that puts five pictures together with intervals of five seconds and plays the audio file of each according to the words.

  


  5s(word1.jpg)   5s(word2.jpg)   5s(word3.jpg)   ...
(audio1.mp3)    (audio2.mp3)    (audio3.mp3) ...
output.mp4


  


  I researched ffmpeg but still haven't found the answer to what I want to do