Merge commit '1c9e8616c535ef496e7ee8a5cbc5e9e972a6977d'

[ffmpeg] / libavformat / dv.c

diff --git a/libavformat/dv.c b/libavformat/dv.c
index c689d9b100ff650b8f7dfbc28441af3cbe325a3e..89a9e10c14a258fcb4242e53057146f40fa4b1ae 100644 (file)
--- a/libavformat/dv.c
+++ b/libavformat/dv.c
@@ -110,11 +110,11 @@ static const int dv_audio_frequency[3] = {
 
 /*
  * There's a couple of assumptions being made here:
- * 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
+ * 1. By default we silence erroneous (0x8000/16-bit 0x800/12-bit) audio samples.
  *    We can pass them upwards when libavcodec will be ready to deal with them.
  * 2. We don't do software emphasis.
- * 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
- *    are converted into 16bit linear ones.
+ * 3. Audio is always returned as 16-bit linear samples: 12-bit nonlinear samples
+ *    are converted into 16-bit linear ones.
  */
 static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
                             const AVDVProfile *sys)
@@ -130,7 +130,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
 
     smpls = as_pack[1]      & 0x3f; /* samples in this frame - min. samples */
     freq  = as_pack[4] >> 3 & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
-    quant = as_pack[4]      & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
+    quant = as_pack[4]      & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */
 
     if (quant > 1)
         return -1;  /* unsupported quantization */
@@ -161,7 +161,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
         for (i = 0; i < sys->difseg_size; i++) {
             frame += 6 * 80; /* skip DIF segment header */
             if (quant == 1 && i == half_ch) {
-                /* next stereo channel (12bit mode only) */
+                /* next stereo channel (12-bit mode only) */
                 av_assert0(ipcm<4);
                 pcm = ppcm[ipcm++];
                 if (!pcm)
@@ -171,7 +171,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
             /* for each AV sequence */
             for (j = 0; j < 9; j++) {
                 for (d = 8; d < 80; d += 2) {
-                    if (quant == 0) {  /* 16bit quantization */
+                    if (quant == 0) {  /* 16-bit quantization */
                         of = sys->audio_shuffle[i][j] +
                              (d - 8) / 2 * sys->audio_stride;
                         if (of * 2 >= size)
@@ -184,7 +184,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
 
                         if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)
                             pcm[of * 2 + 1] = 0;
-                    } else {           /* 12bit quantization */
+                    } else {           /* 12-bit quantization */
                         lc = ((uint16_t)frame[d]     << 4) |
                              ((uint16_t)frame[d + 2] >> 4);
                         rc = ((uint16_t)frame[d + 1] << 4) |
@@ -233,7 +233,7 @@ static int dv_extract_audio_info(DVDemuxContext *c, const uint8_t *frame)
     smpls = as_pack[1]      & 0x3f; /* samples in this frame - min. samples */
     freq  = as_pack[4] >> 3 & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
     stype = as_pack[3]      & 0x1f; /* 0 - 2CH, 2 - 4CH, 3 - 8CH */
-    quant = as_pack[4]      & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
+    quant = as_pack[4]      & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */
 
     if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency)) {
         av_log(c->fctx, AV_LOG_ERROR,