Cleanup messy data type of unknown_payload_type
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
index 6b55202..0293911 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -377,7 +377,7 @@
*/
virtual WebRtc_Word32 SetRTPKeepaliveStatus(
const bool enable,
- const WebRtc_Word8 unknownPayloadType,
+ const int unknownPayloadType,
const WebRtc_UWord16 deltaTransmitTimeMS) = 0;
/*
@@ -391,7 +391,7 @@
*/
virtual WebRtc_Word32 RTPKeepaliveStatus(
bool* enable,
- WebRtc_Word8* unknownPayloadType,
+ int* unknownPayloadType,
WebRtc_UWord16* deltaTransmitTimeMS) const = 0;
/*
diff --git a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index cc71e7b..d3ab74f 100644
--- a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -77,7 +77,10 @@
MOCK_METHOD2(IncomingPacket,
WebRtc_Word32(const WebRtc_UWord8* incomingPacket, const WebRtc_UWord16 packetLength));
MOCK_METHOD4(IncomingAudioNTP,
- WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs, const WebRtc_UWord32 audioReceivedNTPfrac, const WebRtc_UWord32 audioRTCPArrivalTimeSecs, const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
+ WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs,
+ const WebRtc_UWord32 audioReceivedNTPfrac,
+ const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
+ const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
MOCK_METHOD0(InitSender,
WebRtc_Word32());
MOCK_METHOD1(RegisterSendTransport,
@@ -85,15 +88,20 @@
MOCK_METHOD1(SetMaxTransferUnit,
WebRtc_Word32(const WebRtc_UWord16 size));
MOCK_METHOD3(SetTransportOverhead,
- WebRtc_Word32(const bool TCP, const bool IPV6, const WebRtc_UWord8 authenticationOverhead));
+ WebRtc_Word32(const bool TCP, const bool IPV6,
+ const WebRtc_UWord8 authenticationOverhead));
MOCK_CONST_METHOD0(MaxPayloadLength,
WebRtc_UWord16());
MOCK_CONST_METHOD0(MaxDataPayloadLength,
WebRtc_UWord16());
MOCK_METHOD3(SetRTPKeepaliveStatus,
- WebRtc_Word32(const bool enable, const WebRtc_Word8 unknownPayloadType, const WebRtc_UWord16 deltaTransmitTimeMS));
+ WebRtc_Word32(const bool enable,
+ const int unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS));
MOCK_CONST_METHOD3(RTPKeepaliveStatus,
- WebRtc_Word32(bool* enable, WebRtc_Word8* unknownPayloadType, WebRtc_UWord16* deltaTransmitTimeMS));
+ WebRtc_Word32(bool* enable,
+ int* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS));
MOCK_CONST_METHOD0(RTPKeepalive,
bool());
MOCK_METHOD1(RegisterSendPayload,
@@ -147,7 +155,13 @@
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(WebRtc_UWord32* available_bandwidth));
MOCK_METHOD7(SendOutgoingData,
- WebRtc_Word32(const FrameType frameType, const WebRtc_Word8 payloadType, const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpVideoHdr));
+ WebRtc_Word32(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ const RTPVideoHeader* rtpVideoHdr));
MOCK_METHOD1(RegisterIncomingRTCPCallback,
WebRtc_Word32(RtcpFeedback* incomingMessagesCallback));
MOCK_CONST_METHOD0(RTCP,
@@ -164,7 +178,8 @@
MOCK_CONST_METHOD4(RemoteNTP,
WebRtc_Word32(WebRtc_UWord32 *ReceivedNTPsecs, WebRtc_UWord32 *ReceivedNTPfrac, WebRtc_UWord32 *RTCPArrivalTimeSecs, WebRtc_UWord32 *RTCPArrivalTimeFrac));
MOCK_METHOD2(AddMixedCNAME,
- WebRtc_Word32(const WebRtc_UWord32 SSRC, const WebRtc_Word8 cName[RTCP_CNAME_SIZE]));
+ WebRtc_Word32(const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]));
MOCK_METHOD1(RemoveMixedCNAME,
WebRtc_Word32(const WebRtc_UWord32 SSRC));
MOCK_CONST_METHOD5(RTT,
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index d25436d..6087cc3 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -870,7 +870,7 @@
WebRtc_Word32 ModuleRtpRtcpImpl::RTPKeepaliveStatus(
bool* enable,
- WebRtc_Word8* unknownPayloadType,
+ int* unknownPayloadType,
WebRtc_UWord16* deltaTransmitTimeMS) const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTPKeepaliveStatus()");
@@ -881,7 +881,7 @@
WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPKeepaliveStatus(
bool enable,
- WebRtc_Word8 unknownPayloadType,
+ const int unknownPayloadType,
WebRtc_UWord16 deltaTransmitTimeMS) {
if (enable) {
WEBRTC_TRACE(
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index e16bf71..19858d5 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -148,13 +148,15 @@
*/
virtual WebRtc_Word32 InitSender();
- virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable,
- const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS);
+ virtual WebRtc_Word32 SetRTPKeepaliveStatus(
+ const bool enable,
+ const int unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS);
- virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const;
+ virtual WebRtc_Word32 RTPKeepaliveStatus(
+ bool* enable,
+ int* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const;
virtual bool RTPKeepalive() const;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 374ce53..7b0755a 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -361,7 +361,7 @@
WebRtc_Word32
RTPSender::RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
+ int* unknownPayloadType,
WebRtc_UWord16* deltaTransmitTimeMS) const
{
CriticalSectionScoped cs(_sendCritsect);
@@ -382,7 +382,7 @@
}
WebRtc_Word32 RTPSender::EnableRTPKeepalive(
- const WebRtc_Word8 unknownPayloadType,
+ const int unknownPayloadType,
const WebRtc_UWord16 deltaTransmitTimeMS) {
CriticalSectionScoped cs(_sendCritsect);
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index 00012e1..1a2cb826 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -199,12 +199,12 @@
/*
* Keep alive
*/
- WebRtc_Word32 EnableRTPKeepalive( const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS);
+ WebRtc_Word32 EnableRTPKeepalive( const int unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS);
WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const;
+ int* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const;
WebRtc_Word32 DisableRTPKeepalive();
diff --git a/src/video_engine/include/vie_rtp_rtcp.h b/src/video_engine/include/vie_rtp_rtcp.h
index 1397222..f618fae 100644
--- a/src/video_engine/include/vie_rtp_rtcp.h
+++ b/src/video_engine/include/vie_rtp_rtcp.h
@@ -263,7 +263,7 @@
virtual int SetRTPKeepAliveStatus(
const int video_channel,
bool enable,
- const char unknown_payload_type,
+ const int unknown_payload_type,
const unsigned int delta_transmit_time_seconds =
KDefaultDeltaTransmitTimeSeconds) = 0;
@@ -271,7 +271,7 @@
virtual int GetRTPKeepAliveStatus(
const int video_channel,
bool& enabled,
- char& unkown_payload_type,
+ int& unkown_payload_type,
unsigned int& delta_transmit_time_seconds) const = 0;
// This function enables capturing of RTP packets to a binary file on a
diff --git a/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index d39f79a..10c1304 100644
--- a/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -661,10 +661,10 @@
// RTP Keepalive
//
{
- char setPT = 123;
+ int setPT = 123;
unsigned int setDeltaTime = 10;
bool enabled = false;
- char getPT = 0;
+ int getPT = 0;
unsigned int getDeltaTime = 0;
EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
tbChannel.videoChannel, true, 119));
diff --git a/src/video_engine/vie_channel.cc b/src/video_engine/vie_channel.cc
index be1bc6d..60de282 100644
--- a/src/video_engine/vie_channel.cc
+++ b/src/video_engine/vie_channel.cc
@@ -1058,7 +1058,7 @@
WebRtc_Word32 ViEChannel::SetKeepAliveStatus(
const bool enable,
- const WebRtc_Word8 unknown_payload_type,
+ const int unknown_payload_type,
const WebRtc_UWord16 delta_transmit_timeMS) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s", __FUNCTION__);
@@ -1109,7 +1109,7 @@
WebRtc_Word32 ViEChannel::GetKeepAliveStatus(
bool& enabled,
- WebRtc_Word8& unknown_payload_type,
+ int& unknown_payload_type,
WebRtc_UWord16& delta_transmit_time_ms) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
diff --git a/src/video_engine/vie_channel.h b/src/video_engine/vie_channel.h
index 2d9bcd2..349a04b 100644
--- a/src/video_engine/vie_channel.h
+++ b/src/video_engine/vie_channel.h
@@ -164,10 +164,10 @@
WebRtc_UWord32& nackBitrateSent) const;
int GetEstimatedReceiveBandwidth(WebRtc_UWord32* estimated_bandwidth) const;
WebRtc_Word32 SetKeepAliveStatus(const bool enable,
- const WebRtc_Word8 unknown_payload_type,
+ const int unknown_payload_type,
const WebRtc_UWord16 delta_transmit_timeMS);
WebRtc_Word32 GetKeepAliveStatus(bool& enable,
- WebRtc_Word8& unknown_payload_type,
+ int& unknown_payload_type,
WebRtc_UWord16& delta_transmit_timeMS);
WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
RTPDirections direction);
diff --git a/src/video_engine/vie_rtp_rtcp_impl.cc b/src/video_engine/vie_rtp_rtcp_impl.cc
index 497481c..0aaf8e4 100644
--- a/src/video_engine/vie_rtp_rtcp_impl.cc
+++ b/src/video_engine/vie_rtp_rtcp_impl.cc
@@ -770,7 +770,7 @@
int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
const int video_channel,
bool enable,
- const char unknown_payload_type,
+ const int unknown_payload_type,
const unsigned int delta_transmit_time_seconds) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
@@ -801,7 +801,7 @@
int ViERTP_RTCPImpl::GetRTPKeepAliveStatus(
const int video_channel,
bool& enabled,
- char& unknown_payload_type,
+ int& unknown_payload_type,
unsigned int& delta_transmit_time_seconds) const {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
diff --git a/src/video_engine/vie_rtp_rtcp_impl.h b/src/video_engine/vie_rtp_rtcp_impl.h
index 72e8131..386c96f 100644
--- a/src/video_engine/vie_rtp_rtcp_impl.h
+++ b/src/video_engine/vie_rtp_rtcp_impl.h
@@ -96,12 +96,12 @@
virtual int SetRTPKeepAliveStatus(
const int video_channel,
bool enable,
- const char unknown_payload_type,
+ const int unknown_payload_type,
const unsigned int delta_transmit_time_seconds);
virtual int GetRTPKeepAliveStatus(
const int video_channel,
bool& enabled,
- char& unkown_payload_type,
+ int& unkown_payload_type,
unsigned int& delta_transmit_time_seconds) const;
virtual int StartRTPDump(const int video_channel,
const char file_nameUTF8[1024],
diff --git a/src/voice_engine/main/interface/voe_rtp_rtcp.h b/src/voice_engine/main/interface/voe_rtp_rtcp.h
index e26d85f..9f8609e 100644
--- a/src/voice_engine/main/interface/voe_rtp_rtcp.h
+++ b/src/voice_engine/main/interface/voe_rtp_rtcp.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -191,12 +191,12 @@
// This functionality can maintain an existing Network Address Translator
// (NAT) mapping while regular RTP is no longer transmitted.
virtual int SetRTPKeepaliveStatus(
- int channel, bool enable, unsigned char unknownPayloadType,
+ int channel, bool enable, int unknownPayloadType,
int deltaTransmitTimeSeconds = 15) = 0;
// Gets the RTP keepalive mechanism status.
virtual int GetRTPKeepaliveStatus(
- int channel, bool& enabled, unsigned char& unknownPayloadType,
+ int channel, bool& enabled, int& unknownPayloadType,
int& deltaTransmitTimeSeconds) = 0;
// Enables capturing of RTP packets to a binary file on a specific
diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc
index 2b0aef8..fb24494 100644
--- a/src/voice_engine/main/source/channel.cc
+++ b/src/voice_engine/main/source/channel.cc
@@ -5646,7 +5646,7 @@
int
Channel::SetRTPKeepaliveStatus(bool enable,
- unsigned char unknownPayloadType,
+ int unknownPayloadType,
int deltaTransmitTimeSeconds)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ -5673,11 +5673,11 @@
int
Channel::GetRTPKeepaliveStatus(bool& enabled,
- unsigned char& unknownPayloadType,
+ int& unknownPayloadType,
int& deltaTransmitTimeSeconds)
{
bool onOff(false);
- WebRtc_Word8 payloadType(0);
+ int payloadType(0);
WebRtc_UWord16 deltaTransmitTimeMS(0);
if (_rtpRtcpModule.RTPKeepaliveStatus(&onOff, &payloadType,
&deltaTransmitTimeMS) != 0)
diff --git a/src/voice_engine/main/source/channel.h b/src/voice_engine/main/source/channel.h
index 2f5b66a..e26413e 100644
--- a/src/voice_engine/main/source/channel.h
+++ b/src/voice_engine/main/source/channel.h
@@ -343,9 +343,9 @@
int GetRTPStatistics(CallStatistics& stats);
int SetFECStatus(bool enable, int redPayloadtype);
int GetFECStatus(bool& enabled, int& redPayloadtype);
- int SetRTPKeepaliveStatus(bool enable, unsigned char unknownPayloadType,
+ int SetRTPKeepaliveStatus(bool enable, int unknownPayloadType,
int deltaTransmitTimeSeconds);
- int GetRTPKeepaliveStatus(bool& enabled, unsigned char& unknownPayloadType,
+ int GetRTPKeepaliveStatus(bool& enabled, int& unknownPayloadType,
int& deltaTransmitTimeSeconds);
int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
int StopRTPDump(RTPDirections direction);
diff --git a/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc b/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
index cbc4d0d..bc7a5c8 100644
--- a/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
+++ b/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -575,7 +575,7 @@
int VoERTP_RTCPImpl::SetRTPKeepaliveStatus(int channel,
bool enable,
- unsigned char unknownPayloadType,
+ int unknownPayloadType,
int deltaTransmitTimeSeconds)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
@@ -603,7 +603,7 @@
int VoERTP_RTCPImpl::GetRTPKeepaliveStatus(int channel,
bool& enabled,
- unsigned char& unknownPayloadType,
+ int& unknownPayloadType,
int& deltaTransmitTimeSeconds)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
diff --git a/src/voice_engine/main/source/voe_rtp_rtcp_impl.h b/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
index 3cdf162..d3a840d0 100644
--- a/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
+++ b/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -90,12 +90,12 @@
// RTP keepalive mechanism (maintains NAT mappings associated to RTP flows)
virtual int SetRTPKeepaliveStatus(int channel,
bool enable,
- unsigned char unknownPayloadType,
+ int unknownPayloadType,
int deltaTransmitTimeSeconds = 15);
virtual int GetRTPKeepaliveStatus(int channel,
bool& enabled,
- unsigned char& unknownPayloadType,
+ int& unknownPayloadType,
int& deltaTransmitTimeSeconds);
// FEC
diff --git a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
index afd0820..666bb4b 100644
--- a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
+++ b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -43,7 +43,7 @@
}
TEST_F(RtpRtcpBeforeStreamingTest, RtpKeepAliveStatusIsOffByDefault) {
- unsigned char payload_type;
+ int payload_type;
int delta_seconds;
bool on;
@@ -56,7 +56,7 @@
}
TEST_F(RtpRtcpBeforeStreamingTest, SetRtpKeepAliveDealsWithInvalidParameters) {
- unsigned char payload_type;
+ int payload_type;
int delta_seconds;
bool on;
@@ -90,7 +90,7 @@
EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
channel_, true, 1));
- unsigned char payload_type;
+ int payload_type;
int delta_seconds;
bool on;
diff --git a/src/voice_engine/main/test/auto_test/voe_extended_test.cc b/src/voice_engine/main/test/auto_test/voe_extended_test.cc
index 8b4ad455..11b41af 100644
--- a/src/voice_engine/main/test/auto_test/voe_extended_test.cc
+++ b/src/voice_engine/main/test/auto_test/voe_extended_test.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -7149,7 +7149,7 @@
ANL();
TEST(GetRTPKeepaliveStatus);
- unsigned char pt;
+ int pt;
int dT;
TEST_MUSTPASS(!rtp_rtcp->GetRTPKeepaliveStatus(-1, enabled, pt, dT));
MARK();
@@ -7609,8 +7609,7 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
-#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE) && \
- !defined(WEBRTC_ANDROID)))
+#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE)))
#if defined(_WIN32)
TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
