admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Clean up abs-send-time for audio.

Browse Source 
BUG=None

Review-Url: https://codereview.webrtc.org/2455013003
Cr-Commit-Position: refs/heads/master@{#14870}
This commit is contained in:
stefan 2016-11-01 03:17:16 -07:00 committed by Commit bot
parent aca3a249c3
commit b521aa704f
13 changed files with 13 additions and 189 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
webrtc/audio/audio_receive_stream.cc
View File

@ -118,11 +118,6 @@ AudioReceiveStream::AudioReceiveStream(
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, extension.id);
RTC_DCHECK(registered);
} else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, extension.id);
RTC_DCHECK(registered);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(

25
webrtc/audio/audio_receive_stream_unittest.cc
View File

@ -51,7 +51,6 @@ const uint32_t kRemoteSsrc = 1234;
const uint32_t kLocalSsrc = 5678;
const size_t kOneByteExtensionHeaderLength = 4;
const size_t kOneByteExtensionLength = 4;
const int kAbsSendTimeId = 2;
const int kAudioLevelId = 3;
const int kTransportSequenceNumberId = 4;
const int kJitterBufferDelay = -7;
@ -89,9 +88,6 @@ struct ConfigHelper {
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
@ -124,8 +120,6 @@ struct ConfigHelper {
stream_config_.rtp.local_ssrc = kLocalSsrc;
stream_config_.rtp.remote_ssrc = kRemoteSsrc;
stream_config_.rtp.nack.rtp_history_ms = 300;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
stream_config_.rtp.extensions.push_back(RtpExtension(
@ -244,15 +238,14 @@ TEST(AudioReceiveStreamTest, ConfigToString) {
AudioReceiveStream::Config config;
config.rtp.remote_ssrc = kRemoteSsrc;
config.rtp.local_ssrc = kLocalSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
config.voe_channel_id = kChannelId;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
EXPECT_EQ(
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, "
"nack: {rtp_history_ms: 0}, extensions: [{uri: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
"rtcp_send_transport: nullptr, "
"voe_channel_id: 2}",
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
"{rtp_history_ms: 0}, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
"rtcp_send_transport: nullptr, voe_channel_id: 2}",
config.ToString());
}
@ -264,11 +257,7 @@ TEST(AudioReceiveStreamTest, ConstructDestruct) {
}
MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
return arg.extension.hasAbsoluteSendTime ==
expected_extension.hasAbsoluteSendTime &&
arg.extension.absoluteSendTime ==
expected_extension.absoluteSendTime &&
arg.extension.hasTransportSequenceNumber ==
return arg.extension.hasTransportSequenceNumber ==
expected_extension.hasTransportSequenceNumber &&
arg.extension.transportSequenceNumber ==
expected_extension.transportSequenceNumber;

4
webrtc/audio/audio_send_stream.cc
View File

@ -75,9 +75,7 @@ AudioSendStream::AudioSendStream(
channel_proxy_->RegisterExternalTransport(config.send_transport);
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAbsSendTimeUri) {
channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kAudioLevelUri) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableSendTransportSequenceNumber(extension.id);

14
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -35,7 +35,6 @@ const int kChannelId = 1;
const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
const int kAbsSendTimeId = 3;
const int kTransportSequenceNumberId = 4;
const int kEchoDelayMedian = 254;
const int kEchoDelayStdDev = -3;
@ -93,8 +92,6 @@ struct ConfigHelper {
stream_config_.rtp.c_name = kCName;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
// Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
@ -120,9 +117,6 @@ struct ConfigHelper {
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
@ -219,8 +213,6 @@ struct ConfigHelper {
TEST(AudioSendStreamTest, ConfigToString) {
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
config.rtp.c_name = kCName;
config.voe_channel_id = kChannelId;
config.min_bitrate_kbps = 12;
@ -235,10 +227,12 @@ TEST(AudioSendStreamTest, ConfigToString) {
config.send_codec_spec.min_ptime_ms = 20;
config.send_codec_spec.max_ptime_ms = 60;
config.send_codec_spec.codec_inst = kIsacCodec;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
EXPECT_EQ(
"{rtp: {ssrc: 1234, extensions: [{uri: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
"nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
"{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
"voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "

3
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -978,9 +978,6 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAudioLevelDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kAbsSendTimeDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
"Enabled") {
capabilities.header_extensions.push_back(webrtc::RtpExtension(

2
webrtc/test/mock_voe_channel_proxy.h
View File

@ -25,9 +25,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));

15
webrtc/voice_engine/channel.cc
View File

@ -2361,21 +2361,6 @@ int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
return 0;
}
int Channel::SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
return SetSendRtpHeaderExtension(enable, kRtpExtensionAbsoluteSendTime, id);
}
int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
rtp_header_parser_->DeregisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime);
if (enable &&
!rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, id)) {
return -1;
}
return 0;
}
void Channel::EnableSendTransportSequenceNumber(int id) {
int ret =
SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);

12
webrtc/voice_engine/channel_proxy.cc
View File

@ -50,24 +50,12 @@ void ChannelProxy::SetNACKStatus(bool enable, int max_packets) {
channel()->SetNACKStatus(enable, max_packets);
}
void ChannelProxy::SetSendAbsoluteSenderTimeStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAbsoluteSenderTimeStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);

2
webrtc/voice_engine/channel_proxy.h
View File

@ -52,9 +52,7 @@ class ChannelProxy {
virtual void SetLocalSSRC(uint32_t ssrc);
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetNACKStatus(bool enable, int max_packets);
virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
virtual void EnableSendTransportSequenceNumber(int id);
virtual void EnableReceiveTransportSequenceNumber(int id);

10
webrtc/voice_engine/include/voe_rtp_rtcp.h
View File

@ -133,16 +133,6 @@ class WEBRTC_DLLEXPORT VoERTP_RTCP {
return 0;
}
// Sets the status of sending absolute sender time on a specific |channel|.
virtual int SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) = 0;
// Sets status of receiving absolute sender time on a specific |channel|.
virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) = 0;
// Sets the RTCP status on a specific |channel|.
virtual int SetRTCPStatus(int channel, bool enable) = 0;

37
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
View File

@ -118,40 +118,3 @@ TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAudioLevel) {
EXPECT_TRUE(verifying_transport_.Wait());
}
TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeNoAbsoluteSenderTime)
{
verifying_transport_.SetAbsoluteSenderTimeId(0);
ResumePlaying();
EXPECT_FALSE(verifying_transport_.Wait());
}
TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAbsoluteSenderTime) {
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
11));
verifying_transport_.SetAbsoluteSenderTimeId(11);
ResumePlaying();
EXPECT_TRUE(verifying_transport_.Wait());
}
TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions1) {
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
9));
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
11));
verifying_transport_.SetAudioLevelId(9);
verifying_transport_.SetAbsoluteSenderTimeId(11);
ResumePlaying();
EXPECT_TRUE(verifying_transport_.Wait());
}
TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions2) {
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
3));
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
9));
verifying_transport_.SetAbsoluteSenderTimeId(3);
// Don't register audio level with header parser - unknown extensions should
// be ignored when parsing.
ResumePlaying();
EXPECT_TRUE(verifying_transport_.Wait());
}

65
webrtc/voice_engine/voe_rtp_rtcp_impl.cc
View File

@ -141,7 +141,7 @@ int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
// the range 1-14 inclusive.
_shared->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetReceiveAbsoluteSenderTimeStatus() invalid id parameter");
"SetReceiveAudioLevelIndicationStatus() invalid id parameter");
return -1;
}
// Set state and id for the specified channel.
@ -156,69 +156,6 @@ int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
return channel_ptr->SetReceiveAudioLevelIndicationStatus(enable, id);
}
int VoERTP_RTCPImpl::SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetSendAbsoluteSenderTimeStatus(channel=%d, enable=%d, id=%u)",
channel, enable, id);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
if (enable && (id < kVoiceEngineMinRtpExtensionId ||
id > kVoiceEngineMaxRtpExtensionId)) {
// [RFC5285] The 4-bit id is the local identifier of this element in
// the range 1-14 inclusive.
_shared->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetSendAbsoluteSenderTimeStatus() invalid id parameter");
return -1;
}
// Set state and id for the specified channel.
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(
VE_CHANNEL_NOT_VALID, kTraceError,
"SetSendAbsoluteSenderTimeStatus() failed to locate channel");
return -1;
}
return channelPtr->SetSendAbsoluteSenderTimeStatus(enable, id);
}
int VoERTP_RTCPImpl::SetReceiveAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) {
WEBRTC_TRACE(
kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetReceiveAbsoluteSenderTimeStatus(channel=%d, enable=%d, id=%u)",
channel, enable, id);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
if (enable && (id < kVoiceEngineMinRtpExtensionId ||
id > kVoiceEngineMaxRtpExtensionId)) {
// [RFC5285] The 4-bit id is the local identifier of this element in
// the range 1-14 inclusive.
_shared->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetReceiveAbsoluteSenderTimeStatus() invalid id parameter");
return -1;
}
// Set state and id for the specified channel.
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(
VE_CHANNEL_NOT_VALID, kTraceError,
"SetReceiveAbsoluteSenderTimeStatus() failed to locate channel");
return -1;
}
return channelPtr->SetReceiveAbsoluteSenderTimeStatus(enable, id);
}
int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetRTCPStatus(channel=%d, enable=%d)", channel, enable);

8
webrtc/voice_engine/voe_rtp_rtcp_impl.h
View File

@ -51,14 +51,6 @@ class VoERTP_RTCPImpl : public VoERTP_RTCP {
bool enable,
unsigned char id) override;
// RTP Header Extension for Absolute Sender Time
int SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) override;
int SetReceiveAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) override;
// Statistics
int GetRTPStatistics(int channel,
unsigned int& averageJitterMs,