zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

[rtp_rtcp] Lint errors cleared from rtp_rtcp/test

Browse Source 
except rand() function that is subject of CL#1519503002
 and namespace that is fixed in CL#1506823002

BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1511413005

Cr-Commit-Position: refs/heads/master@{#11012}
This commit is contained in:
danilchap
2015-12-14 06:39:33 -08:00
committed by Commit bot
parent 3207916f35
commit 40f349fdda
4 changed files with 48 additions and 52 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -165,7 +165,7 @@ TEST_F(RtpRtcpAudioTest, Basic) {
module1->SetStartTimestamp(test_timestamp);
// Test detection at the end of a DTMF tone.
//EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
// EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
EXPECT_EQ(0, module1->SetSendingStatus(true));
@ -334,7 +334,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
// Send RTP packets for 16 tones a 160 ms 100ms
// pause between = 2560ms + 1600ms = 4160ms
for (;timeStamp <= 250 * 160; timeStamp += 160) {
for (; timeStamp <= 250 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
timeStamp, -1, test, 4));
fake_clock.AdvanceTimeMilliseconds(20);
@ -342,7 +342,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
}
EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
for (;timeStamp <= 740 * 160; timeStamp += 160) {
for (; timeStamp <= 740 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
timeStamp, -1, test, 4));
fake_clock.AdvanceTimeMilliseconds(20);

17
webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
View File

@ -30,30 +30,29 @@ class RtcpCallback : public RtcpIntraFrameObserver {
public:
void SetModule(RtpRtcp* module) {
_rtpRtcpModule = module;
};
}
virtual void OnRTCPPacketTimeout(const int32_t id) {
}
virtual void OnLipSyncUpdate(const int32_t id,
const int32_t audioVideoOffset) {
};
virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
};
const int32_t audioVideoOffset) {}
virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
virtual void OnReceivedSLI(uint32_t ssrc,
uint8_t pictureId) {
EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
};
}
virtual void OnReceivedRPSI(uint32_t ssrc,
uint64_t pictureId) {
EXPECT_EQ(kTestPictureId, pictureId);
};
virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {};
}
virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}
private:
RtpRtcp* _rtpRtcpModule;
};
class TestRtpFeedback : public NullRtpFeedback {
public:
TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
virtual ~TestRtpFeedback() {}
void OnIncomingSSRCChanged(const uint32_t ssrc) override {