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Fix for RTP extension audio level.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/339002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-01-04 17:04:51 +00:00
parent d77a6614fa
commit 6c1d41583a
8 changed files with 53 additions and 44 deletions
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5
src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
View File

@ -41,8 +41,9 @@ enum RTPAliveType
enum RTPExtensionType
{
NONE,
TRANSMISSION_TIME_OFFSET
kRtpExtensionNone,
kRtpExtensionTransmissionTimeOffset,
kRtpExtensionAudioLevel,
};
enum RTCPAppSubTypes

8
src/modules/rtp_rtcp/source/rtp_header_extension.cc
View File

@ -112,7 +112,7 @@ WebRtc_Word32 RtpHeaderExtensionMap::Size() const {
RTPExtensionType RtpHeaderExtensionMap::First() const {
MapItem* item = extensionMap_.First();
if (item == NULL) {
return NONE;
return kRtpExtensionNone;
}
HeaderExtension* extension = (HeaderExtension*)item->GetItem();
return extension->type;
@ -122,15 +122,15 @@ RTPExtensionType RtpHeaderExtensionMap::Next(RTPExtensionType type) const
{
WebRtc_UWord8 id;
if (GetId(type, &id) != 0) {
return NONE;
return kRtpExtensionNone;
}
MapItem* item = extensionMap_.Find(id);
if (item == NULL) {
return NONE;
return kRtpExtensionNone;
}
item = extensionMap_.Next(item);
if (item == NULL) {
return NONE;
return kRtpExtensionNone;
}
HeaderExtension* extension = (HeaderExtension*)item->GetItem();
return extension->type;

2
src/modules/rtp_rtcp/source/rtp_header_extension.h
View File

@ -28,7 +28,7 @@ struct HeaderExtension {
HeaderExtension(RTPExtensionType extension_type)
: type(extension_type),
length(0) {
if (type == TRANSMISSION_TIME_OFFSET) {
if (type == kRtpExtensionTransmissionTimeOffset) {
length = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
}
}

42
src/modules/rtp_rtcp/source/rtp_header_extension_test.cc
View File

@ -32,31 +32,31 @@ class RtpHeaderExtensionTest : public ::testing::Test {
TEST_F(RtpHeaderExtensionTest, Register) {
EXPECT_EQ(0, map_.Size());
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(1, map_.Size());
EXPECT_EQ(0, map_.Deregister(TRANSMISSION_TIME_OFFSET));
EXPECT_EQ(0, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(0, map_.Size());
}
TEST_F(RtpHeaderExtensionTest, RegisterIllegalArg) {
// Valid range for id: [1-14].
EXPECT_EQ(-1, map_.Register(TRANSMISSION_TIME_OFFSET, 0));
EXPECT_EQ(-1, map_.Register(TRANSMISSION_TIME_OFFSET, 15));
EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 0));
EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 15));
}
TEST_F(RtpHeaderExtensionTest, DeregisterIllegalArg) {
// Not registered.
EXPECT_EQ(-1, map_.Deregister(TRANSMISSION_TIME_OFFSET));
EXPECT_EQ(-1, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
}
TEST_F(RtpHeaderExtensionTest, NonUniqueId) {
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(-1, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
}
TEST_F(RtpHeaderExtensionTest, GetLength) {
EXPECT_EQ(0, map_.GetTotalLengthInBytes());
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES +
TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES,
map_.GetTotalLengthInBytes());
@ -66,41 +66,41 @@ TEST_F(RtpHeaderExtensionTest, GetType) {
RTPExtensionType typeOut;
EXPECT_EQ(-1, map_.GetType(kId, &typeOut));
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(0, map_.GetType(kId, &typeOut));
EXPECT_EQ(TRANSMISSION_TIME_OFFSET, typeOut);
EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, typeOut);
}
TEST_F(RtpHeaderExtensionTest, GetId) {
WebRtc_UWord8 idOut;
EXPECT_EQ(-1, map_.GetId(TRANSMISSION_TIME_OFFSET, &idOut));
EXPECT_EQ(-1, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.GetId(TRANSMISSION_TIME_OFFSET, &idOut));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(0, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
EXPECT_EQ(kId, idOut);
}
TEST_F(RtpHeaderExtensionTest, IterateTypes) {
EXPECT_EQ(NONE, map_.First());
EXPECT_EQ(NONE, map_.Next(TRANSMISSION_TIME_OFFSET));
EXPECT_EQ(kRtpExtensionNone, map_.First());
EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(TRANSMISSION_TIME_OFFSET, map_.First());
EXPECT_EQ(NONE, map_.Next(TRANSMISSION_TIME_OFFSET));
EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, map_.First());
EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
}
TEST_F(RtpHeaderExtensionTest, GetCopy) {
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
RtpHeaderExtensionMap mapOut;
map_.GetCopy(&mapOut);
EXPECT_EQ(1, mapOut.Size());
EXPECT_EQ(TRANSMISSION_TIME_OFFSET, mapOut.First());
EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, mapOut.First());
}
TEST_F(RtpHeaderExtensionTest, Erase) {
EXPECT_EQ(0, map_.Register(TRANSMISSION_TIME_OFFSET, kId));
EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
EXPECT_EQ(1, map_.Size());
map_.Erase();
EXPECT_EQ(0, map_.Size());

20
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -2090,13 +2090,19 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus(
const bool enable,
const WebRtc_UWord8 ID) {
WEBRTC_TRACE(kTraceModuleCall,
kTraceRtpRtcp,
_id,
"SetRTPAudioLevelIndicationStatus(enable=%d, ID=%u)",
enable,
ID);
return _rtpSender.SetAudioLevelIndicationStatus(enable, ID);
WEBRTC_TRACE(kTraceModuleCall,
kTraceRtpRtcp,
_id,
"SetRTPAudioLevelIndicationStatus(enable=%d, ID=%u)",
enable,
ID);
if (enable) {
_rtpReceiver.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, ID);
} else {
_rtpReceiver.DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
}
return _rtpSender.SetAudioLevelIndicationStatus(enable, ID);
}
WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus(

7
src/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -1314,10 +1314,10 @@ RTPSender::BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const
WebRtc_UWord16 total_block_length = 0;
RTPExtensionType type = _rtpHeaderExtensionMap.First();
while (type != NONE)
while (type != kRtpExtensionNone)
{
WebRtc_UWord8 block_length = 0;
if (type == TRANSMISSION_TIME_OFFSET)
if (type == kRtpExtensionTransmissionTimeOffset)
{
block_length = BuildTransmissionTimeOffsetExtension(
dataBuffer + kHeaderLength + total_block_length);
@ -1363,7 +1363,8 @@ RTPSender::BuildTransmissionTimeOffsetExtension(WebRtc_UWord8* dataBuffer) const
// Get id defined by user.
WebRtc_UWord8 id;
if (_rtpHeaderExtensionMap.GetId(TRANSMISSION_TIME_OFFSET, &id) != 0) {
if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id)
!= 0) {
// Not registered.
return 0;
}

2
src/modules/rtp_rtcp/source/rtp_sender_test.cc
View File

@ -37,7 +37,7 @@ class RtpSenderTest : public ::testing::Test {
RtpSenderTest()
: rtp_sender_(new RTPSender(0, false, ModuleRTPUtility::GetSystemClock())),
kMarkerBit(true),
kType(TRANSMISSION_TIME_OFFSET) {
kType(kRtpExtensionTransmissionTimeOffset) {
EXPECT_EQ(0, rtp_sender_->SetSequenceNumber(kSeqNum));
}
~RtpSenderTest() {

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

@ -695,14 +695,14 @@ void ModuleRTPUtility::RTPHeaderParser::ParseOneByteExtensionHeader(
RTPExtensionType type;
if (ptrExtensionMap->GetType(id, &type) != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
"Failed to find extension id: %d", id);
return;
}
switch (type)
{
case TRANSMISSION_TIME_OFFSET:
case kRtpExtensionTransmissionTimeOffset:
{
if (len != 2)
{
@ -722,8 +722,8 @@ void ModuleRTPUtility::RTPHeaderParser::ParseOneByteExtensionHeader(
parsedPacket.extension.transmissionTimeOffset = transmissionTimeOffset;
break;
}
//case RTP_AUDIO_LEVEL_ID;
//{
case kRtpExtensionAudioLevel:
{
// --- Only used for debugging ---
// 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
@ -737,7 +737,8 @@ void ModuleRTPUtility::RTPHeaderParser::ParseOneByteExtensionHeader(
//const WebRtc_UWord8 level = (*ptr & 0x7f);
//DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, level=%u",
// ID, len, V, level);
//}
break;
}
default:
{
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,