avio: avio_ prefixes for get_* functions

In the name of consistency:
get_byte           -> avio_r8
get_<type>         -> avio_r<type>
get_buffer         -> avio_read

get_partial_buffer will be made private later

get_strz is left out becase I want to change it later to return
something useful.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

1 files changed, 15 insertions, 15 deletions

diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c
index 017a8a6050..40d0e6296a 100644
--- a/libavformat/bethsoftvid.c
+++ b/libavformat/bethsoftvid.c
@@ -68,7 +68,7 @@ static int vid_read_header(AVFormatContext *s,
     *    int16s: always_512, nframes, width, height, delay, always_14
     */
     url_fseek(pb, 5, SEEK_CUR);
-    vid->nframes = get_le16(pb);
+    vid->nframes = avio_rl16(pb);
 
     stream = av_new_stream(s, 0);
     if (!stream)
@@ -76,11 +76,11 @@ static int vid_read_header(AVFormatContext *s,
     av_set_pts_info(stream, 32, 1, 60);     // 16 ms increments, i.e. 60 fps
     stream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
     stream->codec->codec_id = CODEC_ID_BETHSOFTVID;
-    stream->codec->width = get_le16(pb);
-    stream->codec->height = get_le16(pb);
+    stream->codec->width = avio_rl16(pb);
+    stream->codec->height = avio_rl16(pb);
     stream->codec->pix_fmt = PIX_FMT_PAL8;
-    vid->bethsoft_global_delay = get_le16(pb);
-    get_le16(pb);
+    vid->bethsoft_global_delay = avio_rl16(pb);
+    avio_rl16(pb);
 
     // done with video codec, set up audio codec
     stream = av_new_stream(s, 0);
@@ -117,11 +117,11 @@ static int read_frame(BVID_DemuxContext *vid, AVIOContext *pb, AVPacket *pkt,
     vidbuf_start[vidbuf_nbytes++] = block_type;
 
     // get the video delay (next int16), and set the presentation time
-    vid->video_pts += vid->bethsoft_global_delay + get_le16(pb);
+    vid->video_pts += vid->bethsoft_global_delay + avio_rl16(pb);
 
     // set the y offset if it exists (decoder header data should be in data section)
     if(block_type == VIDEO_YOFF_P_FRAME){
-        if(get_buffer(pb, &vidbuf_start[vidbuf_nbytes], 2) != 2)
+        if(avio_read(pb, &vidbuf_start[vidbuf_nbytes], 2) != 2)
             goto fail;
         vidbuf_nbytes += 2;
     }
@@ -131,21 +131,21 @@ static int read_frame(BVID_DemuxContext *vid, AVIOContext *pb, AVPacket *pkt,
         if(!vidbuf_start)
             return AVERROR(ENOMEM);
 
-        code = get_byte(pb);
+        code = avio_r8(pb);
         vidbuf_start[vidbuf_nbytes++] = code;
 
         if(code >= 0x80){ // rle sequence
             if(block_type == VIDEO_I_FRAME)
-                vidbuf_start[vidbuf_nbytes++] = get_byte(pb);
+                vidbuf_start[vidbuf_nbytes++] = avio_r8(pb);
         } else if(code){ // plain sequence
-            if(get_buffer(pb, &vidbuf_start[vidbuf_nbytes], code) != code)
+            if(avio_read(pb, &vidbuf_start[vidbuf_nbytes], code) != code)
                 goto fail;
             vidbuf_nbytes += code;
         }
         bytes_copied += code & 0x7F;
         if(bytes_copied == npixels){ // sometimes no stop character is given, need to keep track of bytes copied
             // may contain a 0 byte even if read all pixels
-            if(get_byte(pb))
+            if(avio_r8(pb))
                 url_fseek(pb, -1, SEEK_CUR);
             break;
         }
@@ -182,7 +182,7 @@ static int vid_read_packet(AVFormatContext *s,
     if(vid->is_finished || url_feof(pb))
         return AVERROR(EIO);
 
-    block_type = get_byte(pb);
+    block_type = avio_r8(pb);
     switch(block_type){
         case PALETTE_BLOCK:
             url_fseek(pb, -1, SEEK_CUR);     // include block type
@@ -195,12 +195,12 @@ static int vid_read_packet(AVFormatContext *s,
             return ret_value;
 
         case FIRST_AUDIO_BLOCK:
-            get_le16(pb);
+            avio_rl16(pb);
             // soundblaster DAC used for sample rate, as on specification page (link above)
-            s->streams[1]->codec->sample_rate = 1000000 / (256 - get_byte(pb));
+            s->streams[1]->codec->sample_rate = 1000000 / (256 - avio_r8(pb));
             s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_coded_sample;
         case AUDIO_BLOCK:
-            audio_length = get_le16(pb);
+            audio_length = avio_rl16(pb);
             ret_value = av_get_packet(pb, pkt, audio_length);
             pkt->stream_index = 1;
             return ret_value != audio_length ? AVERROR(EIO) : ret_value;