X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Fdss.c;h=ee3a099a950c417eb0b0eca72da20669b7414415;hb=2606c48391377681541111263de41c79adeffa49;hp=0585049130234d0ac4fbe05387a98f8fbbcffb93;hpb=aebf314ab8ca93c00c2ee055d23d7ba14274a483;p=ffmpeg

diff --git a/libavformat/dss.c b/libavformat/dss.c
index 05850491302..ee3a099a950 100644
--- a/libavformat/dss.c
+++ b/libavformat/dss.c
@@ -50,7 +50,6 @@ typedef struct DSSDemuxContext {
     int counter;
     int swap;
     int dss_sp_swap_byte;
-    int8_t dss_sp_buf[DSS_FRAME_SIZE + 1];
 
     int packet_size;
     int dss_header_size;
@@ -144,6 +143,8 @@ static int dss_read_header(AVFormatContext *s)
     if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
         st->codecpar->codec_id    = AV_CODEC_ID_DSS_SP;
         st->codecpar->sample_rate = 11025;
+        s->bit_rate = 8 * (DSS_FRAME_SIZE - 1) * st->codecpar->sample_rate
+                        * 512 / (506 * 264);
     } else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
         st->codecpar->codec_id    = AV_CODEC_ID_G723_1;
         st->codecpar->sample_rate = 8000;
@@ -180,33 +181,29 @@ static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt)
     ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE;
 }
 
-static void dss_sp_byte_swap(DSSDemuxContext *ctx,
-                             uint8_t *dst, const uint8_t *src)
+static void dss_sp_byte_swap(DSSDemuxContext *ctx, uint8_t *data)
 {
     int i;
 
     if (ctx->swap) {
-        for (i = 3; i < DSS_FRAME_SIZE; i += 2)
-            dst[i] = src[i];
-
         for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
-            dst[i] = src[i + 4];
+            data[i] = data[i + 4];
 
-        dst[1] = ctx->dss_sp_swap_byte;
+        /* Zero the padding. */
+        data[DSS_FRAME_SIZE] = 0;
+        data[1] = ctx->dss_sp_swap_byte;
     } else {
-        memcpy(dst, src, DSS_FRAME_SIZE);
-        ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2];
+        ctx->dss_sp_swap_byte = data[DSS_FRAME_SIZE - 2];
     }
 
     /* make sure byte 40 is always 0 */
-    dst[DSS_FRAME_SIZE - 2] = 0;
+    data[DSS_FRAME_SIZE - 2] = 0;
     ctx->swap             ^= 1;
 }
 
 static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     DSSDemuxContext *ctx = s->priv_data;
-    AVStream *st = s->streams[0];
     int read_size, ret, offset = 0, buff_offset = 0;
     int64_t pos = avio_tell(s->pb);
 
@@ -219,9 +216,6 @@ static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
     } else
         read_size = DSS_FRAME_SIZE;
 
-    ctx->counter -= read_size;
-    ctx->packet_size = DSS_FRAME_SIZE - 1;
-
     ret = av_new_packet(pkt, DSS_FRAME_SIZE);
     if (ret < 0)
         return ret;
@@ -229,32 +223,31 @@ static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
     pkt->duration     = 264;
     pkt->pos = pos;
     pkt->stream_index = 0;
-    s->bit_rate = 8LL * ctx->packet_size * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
 
-    if (ctx->counter < 0) {
-        int size2 = ctx->counter + read_size;
-
-        ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
-                        size2 - offset);
-        if (ret < size2 - offset)
+    if (ctx->counter < read_size) {
+        ret = avio_read(s->pb, pkt->data + buff_offset,
+                        ctx->counter);
+        if (ret < ctx->counter)
             goto error_eof;
 
+        offset = ctx->counter;
         dss_skip_audio_header(s, pkt);
-        offset = size2;
     }
+    ctx->counter -= read_size;
 
-    ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
+    /* This will write one byte into pkt's padding if buff_offset == 3 */
+    ret = avio_read(s->pb, pkt->data + offset + buff_offset,
                     read_size - offset);
     if (ret < read_size - offset)
         goto error_eof;
 
-    dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);
+    dss_sp_byte_swap(ctx, pkt->data);
 
     if (ctx->dss_sp_swap_byte < 0) {
         return AVERROR(EAGAIN);
     }
 
-    return pkt->size;
+    return 0;
 
 error_eof:
     return ret < 0 ? ret : AVERROR_EOF;
@@ -278,7 +271,7 @@ static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
     size = frame_size[byte & 3];
 
     ctx->packet_size = size;
-    ctx->counter -= size;
+    ctx->counter--;
 
     ret = av_new_packet(pkt, size);
     if (ret < 0)
@@ -288,29 +281,28 @@ static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
     pkt->data[0]  = byte;
     offset        = 1;
     pkt->duration = 240;
-    s->bit_rate = 8LL * size * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
+    s->bit_rate = 8LL * size-- * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
 
     pkt->stream_index = 0;
 
-    if (ctx->counter < 0) {
-        int size2 = ctx->counter + size;
-
+    if (ctx->counter < size) {
         ret = avio_read(s->pb, pkt->data + offset,
-                        size2 - offset);
-        if (ret < size2 - offset) {
+                        ctx->counter);
+        if (ret < ctx->counter)
             return ret < 0 ? ret : AVERROR_EOF;
-        }
 
+        offset += ctx->counter;
+        size   -= ctx->counter;
+        ctx->counter = 0;
         dss_skip_audio_header(s, pkt);
-        offset = size2;
     }
+    ctx->counter -= size;
 
-    ret = avio_read(s->pb, pkt->data + offset, size - offset);
-    if (ret < size - offset) {
+    ret = avio_read(s->pb, pkt->data + offset, size);
+    if (ret < size)
         return ret < 0 ? ret : AVERROR_EOF;
-    }
 
-    return pkt->size;
+    return 0;
 }
 
 static int dss_read_packet(AVFormatContext *s, AVPacket *pkt)
@@ -362,7 +354,7 @@ static int dss_read_seek(AVFormatContext *s, int stream_index,
 }
 
 
-AVInputFormat ff_dss_demuxer = {
+const AVInputFormat ff_dss_demuxer = {
     .name           = "dss",
     .long_name      = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
     .priv_data_size = sizeof(DSSDemuxContext),