shithub: opus-tools

--- a/src/opusdec.c

+++ b/src/opusdec.c

@@ -187,7 +187,7 @@

    char *c=comments;

    int len, i, nb_fields;

    char *end;

    if (strncmp(c, "OpusTags", 8) != 0)

       fprintf (stderr, "Invalid/corrupted comments\n");

@@ -250,7 +250,7 @@

       if (audio_fd<0)

          perror("Cannot open /dev/dsp");

-         exit(1);

+         exit(1);

       format=AFMT_S16_NE;

@@ -287,7 +287,7 @@

 #elif defined HAVE_SYS_AUDIOIO_H

       audio_info_t info;

       int audio_fd;

       audio_fd = open("/dev/audio", O_WRONLY);

       if (audio_fd<0)

@@ -303,7 +303,7 @@

       info.play.precision = 16;

       info.play.input_sample_rate = rate;

       info.play.channels = *channels;

       if (ioctl(audio_fd, AUDIO_SETINFO, &info) < 0)

          perror ("AUDIO_SETINFO");

@@ -331,7 +331,7 @@

 #endif

          fout=stdout;

-      else

+      else

          fout = fopen(outFile, "wb");

          if (!fout)

@@ -355,13 +355,13 @@

    printf ("input_file can be:\n");

    printf ("  filename.oga         regular Opus file\n");

    printf ("  -                    stdin\n");

-   printf ("\n");

+   printf ("\n");

    printf ("output_file can be:\n");

    printf ("  filename.wav         Wav file\n");

    printf ("  filename.*           Raw PCM file (any extension other that .wav)\n");

    printf ("  -                    stdout\n");

    printf ("  (nothing)            Will be played to soundcard\n");

-   printf ("\n");

+   printf ("\n");

    printf ("Options:\n");

    printf (" --mono                Force decoding in mono\n");

    printf (" --stereo              Force decoding in stereo\n");

@@ -368,7 +368,7 @@

    printf (" --rate n              Force decoding at sampling rate n Hz\n");

    printf (" --no-dither           Do not dither 16-bit output\n");

    printf (" --packet-loss n       Simulate n %% random packet loss\n");

-   printf (" -V                    Verbose mode (show bit-rate)\n");

+   printf (" -V                    Verbose mode (show bit-rate)\n");

    printf (" -h, --help            This help\n");

    printf (" -v, --version         Version information\n");

    printf ("\n");

@@ -388,6 +388,7 @@

 static OpusMSDecoder *process_header(ogg_packet *op, opus_int32 *rate, int *channels, int *preskip, float *gain, int quiet)

+   int error;

    OpusMSDecoder *st;

    OpusHeader header;

    unsigned char mapping[256] = {0,1};

@@ -397,7 +398,7 @@

       fprintf(stderr, "Cannot parse header\n");

       return NULL;

    if (header.channels>2 || header.channels<1)

       fprintf (stderr, "Unsupported number of channels: %d\n", header.channels);

@@ -409,10 +410,10 @@

    if (!*rate)

       *rate = header.input_sample_rate;

    *preskip = header.preskip;

-   st = opus_multistream_decoder_create(48000, header.channels, 1, header.channels==2 ? 1 : 0, mapping);

+   st = opus_multistream_decoder_create(48000, header.channels, 1, header.channels==2 ? 1 : 0, mapping, &error);

    if (!st)

-      fprintf (stderr, "Decoder initialization failed.\n");

+      fprintf (stderr, "Decoder initialization failed: %s\n", opus_strerror(error));

       return NULL;

@@ -530,7 +531,7 @@

    shapemem.b_buf=0;

    shapemem.mute=960;

    shapemem.fs=0;

    /*Process options*/

    while(1)

@@ -538,7 +539,7 @@

                        long_options, &option_index);

       if (c==-1)

          break;

       switch(c)

       case 0:

@@ -613,7 +614,7 @@

 #endif

       fin=stdin;

-   else

+   else

       fin = fopen(inFile, "rb");

       if (!fin)

@@ -627,9 +628,9 @@

    /*Init Ogg data struct*/

    ogg_sync_init(&oy);

    /*Main decoding loop*/

    while (1)

       char *data;

@@ -637,7 +638,7 @@

       /*Get the ogg buffer for writing*/

       data = ogg_sync_buffer(&oy, 200);

       /*Read bitstream from input file*/

-      nb_read = fread(data, sizeof(char), 200, fin);

+      nb_read = fread(data, sizeof(char), 200, fin);

       ogg_sync_wrote(&oy, nb_read);

       /*Loop for all complete pages we got (most likely only one)*/

@@ -751,7 +752,7 @@

          break;

    /* Drain the resampler */

    if (resampler)

@@ -758,7 +759,7 @@

       int i;

       float zeros[200];

       int drain;

       for (i=0;i<200;i++)

          zeros[i] = 200;

       drain = speex_resampler_get_input_latency(resampler);

@@ -770,7 +771,7 @@

          drain -= tmp;

       } while (drain>0);

    if (fout && wav_format)

       if (fseek(fout,4,SEEK_SET)==0)

@@ -809,7 +810,7 @@

    if (close_in)

       fclose(fin);

    if (fout != NULL)

-      fclose(fout);

+      fclose(fout);

    return 0;

--- a/src/opusenc.c

+++ b/src/opusenc.c

@@ -67,7 +67,7 @@

    int written;

    written = fwrite(page->header,1,page->header_len, fp);

    written += fwrite(page->body,1,page->body_len, fp);

    return written;

@@ -77,7 +77,7 @@

 #define IMAX(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum int value.   */

 /* Convert input audio bits, endians and channels */

-static int read_samples_pcm(FILE *fin,int frame_size, int bits, int channels,

+static int read_samples_pcm(FILE *fin,int frame_size, int bits, int channels,

                             int lsb, float * input, char *buff, opus_int32 *size,

                             int *extra_samples)

@@ -127,8 +127,8 @@

       /* convert to our endian format */

       for(i=0;i<frame_size*channels;i++)

-         if(lsb)

-            s[i]=le_short(s[i]);

+         if(lsb)

+            s[i]=le_short(s[i]);

          else

             s[i]=be_short(s[i]);

@@ -157,7 +157,7 @@

    return nb_read;

-static int read_samples(FILE *fin,int frame_size, int bits, int channels,

+static int read_samples(FILE *fin,int frame_size, int bits, int channels,

                         int lsb, float * input, char *buff, opus_int32 *size,

                         SpeexResamplerState *resampler, int *extra_samples)

@@ -217,15 +217,15 @@

    printf ("  filename.wav      wav file\n");

    printf ("  filename.*        Raw PCM file (any extension other than .wav)\n");

    printf ("  -                 stdin\n");

-   printf ("\n");

+   printf ("\n");

    printf ("output_file can be:\n");

    printf ("  filename.oga      compressed file\n");

    printf ("  -                 stdout\n");

-   printf ("\n");

+   printf ("\n");

    printf ("Options:\n");

-   printf (" --speech           Optimize for speech\n");

-   printf (" --music            Optimize for music\n");

-   printf (" --bitrate n        Encoding bit-rate in kbit/sec\n");

+   printf (" --speech           Optimize for speech\n");

+   printf (" --music            Optimize for music\n");

+   printf (" --bitrate n        Encoding bit-rate in kbit/sec\n");

    printf (" --cbr              Use constant bitrate encoding\n");

    printf (" --comp n           Encoding complexity (0-10)\n");

    printf (" --framesize n      Frame size (Default: 960)\n");

@@ -235,17 +235,17 @@

    printf ("                     used multiple times\n");

    printf (" --author           Author of this track\n");

    printf (" --title            Title for this track\n");

-   printf (" -h, --help         This help\n");

-   printf (" -v, --version      Version information\n");

-   printf (" -V                 Verbose mode (show bit-rate)\n");

+   printf (" -h, --help         This help\n");

+   printf (" -v, --version      Version information\n");

+   printf (" -V                 Verbose mode (show bit-rate)\n");

    printf ("Raw input options:\n");

-   printf (" --rate n           Sampling rate for raw input\n");

-   printf (" --mono             Consider raw input as mono\n");

-   printf (" --stereo           Consider raw input as stereo\n");

-   printf (" --le               Raw input is little-endian\n");

-   printf (" --be               Raw input is big-endian\n");

-   printf (" --8bit             Raw input is 8-bit unsigned\n");

-   printf (" --16bit            Raw input is 16-bit signed\n");

+   printf (" --rate n           Sampling rate for raw input\n");

+   printf (" --mono             Consider raw input as mono\n");

+   printf (" --stereo           Consider raw input as stereo\n");

+   printf (" --le               Raw input is little-endian\n");

+   printf (" --be               Raw input is big-endian\n");

+   printf (" --8bit             Raw input is 8-bit unsigned\n");

+   printf (" --16bit            Raw input is 16-bit signed\n");

    printf ("Default raw PCM input is 48kHz, 16-bit, little-endian, stereo\n");

@@ -335,7 +335,7 @@

                        long_options, &option_index);

       if (c==-1)

          break;

       switch(c)

       case 0:

@@ -404,13 +404,13 @@

 	     fprintf (stderr, "Comments must be of the form name=value\n");

 	     exit(1);

-           comment_add(&comments, &comments_length, NULL, optarg);

+           comment_add(&comments, &comments_length, NULL, optarg);

          } else if (strcmp(long_options[option_index].name,"author")==0)

-           comment_add(&comments, &comments_length, "author=", optarg);

+           comment_add(&comments, &comments_length, "author=", optarg);

          } else if (strcmp(long_options[option_index].name,"title")==0)

-           comment_add(&comments, &comments_length, "title=", optarg);

+           comment_add(&comments, &comments_length, "title=", optarg);

          break;

@@ -456,7 +456,7 @@

 #endif

       fin=stdin;

-   else

+   else

       fin = fopen(inFile, "rb");

       if (!fin)

@@ -497,9 +497,14 @@

        bitrate=128.0;

    bytes_per_packet = MAX_FRAME_BYTES;

    /*Initialize OPUS encoder*/

-   st = opus_multistream_encoder_create(48000, chan, 1, chan==2, mapping, OPUS_APPLICATION_AUDIO);

+   st = opus_multistream_encoder_create(48000, chan, 1, chan==2, mapping, OPUS_APPLICATION_AUDIO, &ret);

+   if (!st)

+   {

+      fprintf (stderr, "Failed to create the encoder: %s\n", opus_strerror(ret));

+      exit(1);

+   }

    opus_multistream_encoder_ctl(st, OPUS_SET_SIGNAL(signal));

    header.channels = chan;

    opus_multistream_encoder_ctl(st, OPUS_GET_LOOKAHEAD(&lookahead));

@@ -512,7 +517,7 @@

    /* 0 dB gain is the recommended unless you know what you're doing */

    header.gain = 0;

    header.input_sample_rate = rate;

    /* Extra samples that need to be read to compensate for the pre-skip */

    extra_samples = (int)header.preskip * (rate/48000.);

@@ -570,7 +575,7 @@

 #endif

       fout=stdout;

-   else

+   else

       fout = fopen(outFile, "wb");

       if (!fout)

@@ -741,8 +746,8 @@

    return 0;

-/*

- Comments will be stored in the Vorbis style.

+/*

+ Comments will be stored in the Vorbis style.

  It is describled in the "Structure" section of

     http://www.xiph.org/ogg/vorbis/doc/v-comment.html