Separating voice activity flag from audio level in RtpHeaderExtension.

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 294 matching lines @@
   // is zero.
   header.extension.hasTransmissionTimeOffset = false;
   header.extension.transmissionTimeOffset = 0;
 
   // May not be present in packet.
   header.extension.hasAbsoluteSendTime = false;
   header.extension.absoluteSendTime = 0;
 
   // May not be present in packet.
   header.extension.hasAudioLevel = false;
+  header.extension.voiceActivity = false;
   header.extension.audioLevel = 0;
 
   // May not be present in packet.
   header.extension.hasVideoRotation = false;
   header.extension.videoRotation = 0;
 
   if (X) {
     /* RTP header extension, RFC 3550.
      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ skipping 91 matching lines @@
           if (len != 0) {
             LOG(LS_WARNING) << "Incorrect audio level len: " << len;
             return;
           }
           //  0                   1
           //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           // |  ID   | len=0 |V|   level     |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //
-
-          // Parse out the fields but only use it for debugging for now.
-          // const uint8_t V = (*ptr & 0x80) >> 7;
-          // const uint8_t level = (*ptr & 0x7f);
-          // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
-          // level=%u", ID, len, V, level);
-
-          header.extension.audioLevel = ptr[0];
+          header.extension.audioLevel = ptr[0] & 0x7f;
+          header.extension.voiceActivity = (ptr[0] & 0x80) != 0;
           header.extension.hasAudioLevel = true;
           break;
         }
         case kRtpExtensionAbsoluteSendTime: {
           if (len != 2) {
             LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
             return;
           }
           //  0                   1                   2                   3
           //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ skipping 60 matching lines @@
       return num_zero_bytes;
     }
     ptr++;
     num_zero_bytes++;
   }
   return num_zero_bytes;
 }
 }  // namespace RtpUtility
 
 }  // namespace webrtc