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Separating voice activity flag from audio level in RtpHeaderExtension.

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VAD flag was embedded in RtpHeaderExtension.audioLevel, which is not easy to interpret. This CL tries to separate the flag with the actual audio level.

BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1272343003 .

Cr-Commit-Position: refs/heads/master@{#9691}
This commit is contained in:
Minyue 2015-08-10 15:08:36 +02:00
parent c2ee2c86f9
commit 4cee419e07
5 changed files with 14 additions and 15 deletions
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1
webrtc/common_types.cc
View File

@ -28,6 +28,7 @@ RTPHeaderExtension::RTPHeaderExtension()
hasTransportSequenceNumber(false),
transportSequenceNumber(0),
hasAudioLevel(false),
voiceActivity(false),
audioLevel(0),
hasVideoRotation(false),
videoRotation(0) {

1
webrtc/common_types.h
View File

@ -804,6 +804,7 @@ struct RTPHeaderExtension {
// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
bool hasAudioLevel;
bool voiceActivity;
uint8_t audioLevel;
// For Coordination of Video Orientation. See

5
webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
View File

@ -123,11 +123,10 @@ int main(int argc, char* argv[]) {
packet->header().markerBit,
packet->header().ssrc);
if (print_audio_level && packet->header().extension.hasAudioLevel) {
// |audioLevel| consists of one bit for "V" and then 7 bits level.
fprintf(out_file,
" %5u (%1i)",
packet->header().extension.audioLevel & 0x7F,
(packet->header().extension.audioLevel & 0x80) == 0 ? 0 : 1);
packet->header().extension.audioLevel,
packet->header().extension.voiceActivity);
}
if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
if (cycles == -1) {

11
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -320,6 +320,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
EXPECT_FALSE(rtp_header.extension.hasAudioLevel);
EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime);
EXPECT_FALSE(rtp_header.extension.voiceActivity);
EXPECT_EQ(0u, rtp_header.extension.audioLevel);
EXPECT_EQ(0u, rtp_header.extension.videoRotation);
}
@ -504,9 +505,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
// Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to
// UpdateAudioLevel(), above.
EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
EXPECT_TRUE(rtp_header.extension.voiceActivity);
EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
// Parse without map extension
webrtc::RTPHeader rtp_header2;
@ -516,6 +516,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
EXPECT_FALSE(rtp_header2.extension.voiceActivity);
EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
}
@ -566,7 +567,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
EXPECT_TRUE(rtp_header.extension.voiceActivity);
EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
EXPECT_EQ(kTransportSequenceNumber,
rtp_header.extension.transportSequenceNumber);
@ -584,6 +586,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
EXPECT_FALSE(rtp_header2.extension.voiceActivity);
EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
}

11
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
View File

@ -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;
}