| Index: webrtc/modules/rtp_rtcp/source/rtp_utility.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
|
| index 0d083bd92a5289f5c4e940c40a0eea096c01190e..2727e7b8bc168ead1d67b213e580a98c37b747b9 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
|
| @@ -312,6 +312,7 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
|
|
|
| // May not be present in packet.
|
| header.extension.hasAudioLevel = false;
|
| + header.extension.voiceActivity = false;
|
| header.extension.audioLevel = 0;
|
|
|
| // May not be present in packet.
|
| @@ -423,14 +424,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
|
| // | 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;
|
| }
|
|
|