Enable cpplint in pc/
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index a6d41c6..80e461a 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -30,7 +30,6 @@
'modules/utility',
'modules/video_capture',
'p2p',
- 'pc',
'rtc_base',
'sdk/android/src/jni',
'sdk/objc',
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index c040854..07333b4 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -525,7 +525,7 @@
class ScopedCallThread {
public:
template <class FunctorT>
- ScopedCallThread(const FunctorT& functor)
+ explicit ScopedCallThread(const FunctorT& functor)
: thread_(rtc::Thread::Create()),
task_(new rtc::FunctorMessageHandler<void, FunctorT>(functor)) {
thread_->Start();
diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index 23abdef..e597032 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -11,6 +11,7 @@
#include "pc/channelmanager.h"
#include <algorithm>
+#include <utility>
#include "media/base/device.h"
#include "media/base/rtpdataengine.h"
diff --git a/pc/currentspeakermonitor_unittest.cc b/pc/currentspeakermonitor_unittest.cc
index 88b1821..fa3f0cb 100644
--- a/pc/currentspeakermonitor_unittest.cc
+++ b/pc/currentspeakermonitor_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <utility>
+
#include "pc/currentspeakermonitor.h"
#include "pc/audiomonitor.h"
#include "rtc_base/gunit.h"
diff --git a/pc/datachannel.cc b/pc/datachannel.cc
index c342908..9bc16df 100644
--- a/pc/datachannel.cc
+++ b/pc/datachannel.cc
@@ -180,7 +180,7 @@
case webrtc::InternalDataChannelInit::kAcker:
handshake_state_ = kHandshakeShouldSendAck;
break;
- };
+ }
// Try to connect to the transport in case the transport channel already
// exists.
diff --git a/pc/datachannel_unittest.cc b/pc/datachannel_unittest.cc
index 7acfa8e0..f8e9880 100644
--- a/pc/datachannel_unittest.cc
+++ b/pc/datachannel_unittest.cc
@@ -9,6 +9,7 @@
*/
#include <memory>
+#include <vector>
#include "pc/datachannel.h"
#include "pc/sctputils.h"
diff --git a/pc/externalhmac.cc b/pc/externalhmac.cc
index 6c6d000..91f7412 100644
--- a/pc/externalhmac.cc
+++ b/pc/externalhmac.cc
@@ -80,7 +80,7 @@
return srtp_err_status_alloc_fail;
// Set pointers
- *a = (srtp_auth_t *)pointer;
+ *a = reinterpret_cast<srtp_auth_t*>(pointer);
// |external_hmac| is const and libsrtp expects |type| to be non-const.
// const conversion is required. |external_hmac| is constant because we don't
// want to increase global count in Chrome.
@@ -95,7 +95,8 @@
srtp_err_status_t external_hmac_dealloc(srtp_auth_t* a) {
// Zeroize entire state
- memset((uint8_t *)a, 0, sizeof(ExternalHmacContext) + sizeof(srtp_auth_t));
+ memset(reinterpret_cast<uint8_t*>(a), 0,
+ sizeof(ExternalHmacContext) + sizeof(srtp_auth_t));
// Free memory
delete[] a;
diff --git a/pc/mediasession.cc b/pc/mediasession.cc
index 835f76d..3ca1074 100644
--- a/pc/mediasession.cc
+++ b/pc/mediasession.cc
@@ -2430,9 +2430,9 @@
// Non-const versions of the above functions.
//
-ContentInfo* GetFirstMediaContent(ContentInfos& contents,
+ContentInfo* GetFirstMediaContent(ContentInfos* contents,
MediaType media_type) {
- for (ContentInfo& content : contents) {
+ for (ContentInfo& content : *contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
}
@@ -2440,15 +2440,15 @@
return nullptr;
}
-ContentInfo* GetFirstAudioContent(ContentInfos& contents) {
+ContentInfo* GetFirstAudioContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}
-ContentInfo* GetFirstVideoContent(ContentInfos& contents) {
+ContentInfo* GetFirstVideoContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}
-ContentInfo* GetFirstDataContent(ContentInfos& contents) {
+ContentInfo* GetFirstDataContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}
@@ -2458,7 +2458,7 @@
return nullptr;
}
- return GetFirstMediaContent(sdesc->contents(), media_type);
+ return GetFirstMediaContent(&sdesc->contents(), media_type);
}
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
diff --git a/pc/mediasession.h b/pc/mediasession.h
index 6de9de9..513f645 100644
--- a/pc/mediasession.h
+++ b/pc/mediasession.h
@@ -618,10 +618,10 @@
const SessionDescription* sdesc);
// Non-const versions of the above functions.
// Useful when modifying an existing description.
-ContentInfo* GetFirstMediaContent(ContentInfos& contents, MediaType media_type);
-ContentInfo* GetFirstAudioContent(ContentInfos& contents);
-ContentInfo* GetFirstVideoContent(ContentInfos& contents);
-ContentInfo* GetFirstDataContent(ContentInfos& contents);
+ContentInfo* GetFirstMediaContent(ContentInfos* contents, MediaType media_type);
+ContentInfo* GetFirstAudioContent(ContentInfos* contents);
+ContentInfo* GetFirstVideoContent(ContentInfos* contents);
+ContentInfo* GetFirstDataContent(ContentInfos* contents);
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
diff --git a/pc/mediasession_unittest.cc b/pc/mediasession_unittest.cc
index 867640c..583362d 100644
--- a/pc/mediasession_unittest.cc
+++ b/pc/mediasession_unittest.cc
@@ -256,7 +256,16 @@
return std::find_if(
opts->media_description_options.begin(),
opts->media_description_options.end(),
- [mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
+ [&mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
+}
+
+std::vector<MediaDescriptionOptions>::const_iterator
+FindFirstMediaDescriptionByMid(const std::string& mid,
+ const MediaSessionOptions& opts) {
+ return std::find_if(
+ opts.media_description_options.begin(),
+ opts.media_description_options.end(),
+ [&mid](const MediaDescriptionOptions& t) { return t.mid == mid; });
}
// Add a media section to the |session_options|.
@@ -402,7 +411,7 @@
}
void TestTransportInfo(bool offer,
- MediaSessionOptions& options,
+ const MediaSessionOptions& options,
bool has_current_desc) {
const std::string current_audio_ufrag = "current_audio_ufrag";
const std::string current_audio_pwd = "current_audio_pwd";
@@ -448,7 +457,7 @@
ti_audio->description.ice_pwd.size());
}
auto media_desc_options_it =
- FindFirstMediaDescriptionByMid("audio", &options);
+ FindFirstMediaDescriptionByMid("audio", options);
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_audio));
@@ -476,7 +485,7 @@
}
}
auto media_desc_options_it =
- FindFirstMediaDescriptionByMid("video", &options);
+ FindFirstMediaDescriptionByMid("video", options);
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_video));
@@ -503,7 +512,7 @@
}
}
auto media_desc_options_it =
- FindFirstMediaDescriptionByMid("data", &options);
+ FindFirstMediaDescriptionByMid("data", options);
EXPECT_EQ(
media_desc_options_it->transport_options.enable_ice_renomination,
GetIceRenomination(ti_data));
diff --git a/pc/mediastream.cc b/pc/mediastream.cc
index 0d6fcda..eb78a8b 100644
--- a/pc/mediastream.cc
+++ b/pc/mediastream.cc
@@ -25,7 +25,7 @@
}
}
return it;
-};
+}
rtc::scoped_refptr<MediaStream> MediaStream::Create(
const std::string& label) {
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 0420b60..03d34d0 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -206,8 +206,8 @@
class SENDERS,
class RECEIVERS>
void SetChannelOnSendersAndReceivers(CHANNEL* channel,
- SENDERS& senders,
- RECEIVERS& receivers,
+ const SENDERS& senders,
+ const RECEIVERS& receivers,
cricket::MediaType media_type) {
for (auto& sender : senders) {
if (sender->media_type() == media_type) {
diff --git a/pc/peerconnectionendtoend_unittest.cc b/pc/peerconnectionendtoend_unittest.cc
index bc80e1e..dce69da 100644
--- a/pc/peerconnectionendtoend_unittest.cc
+++ b/pc/peerconnectionendtoend_unittest.cc
@@ -184,7 +184,7 @@
std::unique_ptr<webrtc::AudioDecoder> real_decoder) {
class ForwardingMockDecoder : public StrictMock<webrtc::MockAudioDecoder> {
public:
- ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder)
+ explicit ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder)
: decoder_(std::move(decoder)) {}
private:
diff --git a/pc/peerconnectionfactory.cc b/pc/peerconnectionfactory.cc
index dc84d76..2dcc4e5 100644
--- a/pc/peerconnectionfactory.cc
+++ b/pc/peerconnectionfactory.cc
@@ -106,7 +106,7 @@
}
}
- // TODO: Currently there is no way creating an external adm in
+ // TODO(deadbeef): Currently there is no way to create an external adm in
// libjingle source tree. So we can 't currently assert if this is NULL.
// RTC_DCHECK(default_adm != NULL);
}
diff --git a/pc/peerconnectionfactory.h b/pc/peerconnectionfactory.h
index f5faecc..398221f 100644
--- a/pc/peerconnectionfactory.h
+++ b/pc/peerconnectionfactory.h
@@ -49,7 +49,7 @@
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) override;
- virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+ rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
@@ -60,13 +60,13 @@
rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) override;
- virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
+ rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) override;
// Deprecated, use version without constraints.
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) override;
- virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
+ rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer) override;
// This version supports filtering on width, height and frame rate.
// For the "constraints=null" case, use the version without constraints.
diff --git a/pc/peerconnectionfactory_unittest.cc b/pc/peerconnectionfactory_unittest.cc
index 6fba8fa..a1bdbf5 100644
--- a/pc/peerconnectionfactory_unittest.cc
+++ b/pc/peerconnectionfactory_unittest.cc
@@ -11,6 +11,7 @@
#include <memory>
#include <string>
#include <utility>
+#include <vector>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
diff --git a/pc/peerconnectionwrapper.cc b/pc/peerconnectionwrapper.cc
index 070deb9..ddfa5a6 100644
--- a/pc/peerconnectionwrapper.cc
+++ b/pc/peerconnectionwrapper.cc
@@ -13,6 +13,7 @@
#include <memory>
#include <string>
#include <utility>
+#include <vector>
#include "api/jsepsessiondescription.h"
#include "media/base/fakevideocapturer.h"
diff --git a/pc/peerconnectionwrapper.h b/pc/peerconnectionwrapper.h
index 88d2f07..2554f1f 100644
--- a/pc/peerconnectionwrapper.h
+++ b/pc/peerconnectionwrapper.h
@@ -14,6 +14,7 @@
#include <functional>
#include <memory>
#include <string>
+#include <vector>
#include "api/peerconnectioninterface.h"
#include "pc/test/mockpeerconnectionobservers.h"
diff --git a/pc/rtcstats_integrationtest.cc b/pc/rtcstats_integrationtest.cc
index e0fb577..3da1083 100644
--- a/pc/rtcstats_integrationtest.cc
+++ b/pc/rtcstats_integrationtest.cc
@@ -56,15 +56,16 @@
}
private:
- static void AddTraceEventHandler(char phase,
- const unsigned char* category_enabled,
- const char* name,
- unsigned long long id,
- int num_args,
- const char** arg_names,
- const unsigned char* arg_types,
- const unsigned long long* arg_values,
- unsigned char flags) {
+ static void AddTraceEventHandler(
+ char phase,
+ const unsigned char* category_enabled,
+ const char* name,
+ unsigned long long id, // NOLINT(runtime/int)
+ int num_args,
+ const char** arg_names,
+ const unsigned char* arg_types,
+ const unsigned long long* arg_values, // NOLINT(runtime/int)
+ unsigned char flags) {
RTC_DCHECK(traced_report_);
EXPECT_STREQ("webrtc_stats",
reinterpret_cast<const char*>(category_enabled));
diff --git a/pc/rtcstatscollector.cc b/pc/rtcstatscollector.cc
index d20f2ed..12a6f5a 100644
--- a/pc/rtcstatscollector.cc
+++ b/pc/rtcstatscollector.cc
@@ -12,6 +12,7 @@
#include <memory>
#include <sstream>
+#include <string>
#include <utility>
#include <vector>
diff --git a/pc/rtcstatscollector.h b/pc/rtcstatscollector.h
index a3a9c1e..6effe73 100644
--- a/pc/rtcstatscollector.h
+++ b/pc/rtcstatscollector.h
@@ -14,6 +14,7 @@
#include <map>
#include <memory>
#include <set>
+#include <string>
#include <vector>
#include "api/optional.h"
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index a5c2ee1..71b3797 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -14,6 +14,7 @@
#include <memory>
#include <ostream>
#include <string>
+#include <utility>
#include <vector>
#include "api/jsepsessiondescription.h"
@@ -181,9 +182,8 @@
return audio_track_stats;
}
- FakeAudioTrackForStats(const std::string& id)
- : MediaStreamTrack<AudioTrackInterface>(id) {
- }
+ explicit FakeAudioTrackForStats(const std::string& id)
+ : MediaStreamTrack<AudioTrackInterface>(id) {}
std::string kind() const override {
return MediaStreamTrackInterface::kAudioKind;
@@ -209,9 +209,8 @@
return video_track;
}
- FakeVideoTrackForStats(const std::string& id)
- : MediaStreamTrack<VideoTrackInterface>(id) {
- }
+ explicit FakeVideoTrackForStats(const std::string& id)
+ : MediaStreamTrack<VideoTrackInterface>(id) {}
std::string kind() const override {
return MediaStreamTrackInterface::kVideoKind;
diff --git a/pc/rtpreceiver.cc b/pc/rtpreceiver.cc
index 4f88f1c..4dc9167 100644
--- a/pc/rtpreceiver.cc
+++ b/pc/rtpreceiver.cc
@@ -10,6 +10,8 @@
#include "pc/rtpreceiver.h"
+#include <vector>
+
#include "api/mediastreamtrackproxy.h"
#include "api/videosourceproxy.h"
#include "pc/audiotrack.h"
diff --git a/pc/rtpreceiver.h b/pc/rtpreceiver.h
index 6b9ca99..d3f0f26 100644
--- a/pc/rtpreceiver.h
+++ b/pc/rtpreceiver.h
@@ -18,6 +18,7 @@
#include <stdint.h>
#include <string>
+#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpreceiverinterface.h"
diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index 3d5594c..ac3a03c 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -10,6 +10,8 @@
#include "pc/rtpsender.h"
+#include <vector>
+
#include "api/mediastreaminterface.h"
#include "pc/localaudiosource.h"
#include "rtc_base/checks.h"
diff --git a/pc/rtpsender.h b/pc/rtpsender.h
index 672637f..ce8c657 100644
--- a/pc/rtpsender.h
+++ b/pc/rtpsender.h
@@ -17,6 +17,7 @@
#include <memory>
#include <string>
+#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpsenderinterface.h"
diff --git a/pc/sdputils.cc b/pc/sdputils.cc
index 8932bea..9bcbc43 100644
--- a/pc/sdputils.cc
+++ b/pc/sdputils.cc
@@ -10,6 +10,7 @@
#include "pc/sdputils.h"
+#include <string>
#include <utility>
#include "api/jsepsessiondescription.h"
diff --git a/pc/sdputils.h b/pc/sdputils.h
index 3a53a41..7444aa7 100644
--- a/pc/sdputils.h
+++ b/pc/sdputils.h
@@ -13,6 +13,7 @@
#include <functional>
#include <memory>
+#include <string>
#include "api/jsep.h"
#include "p2p/base/sessiondescription.h"
diff --git a/pc/srtpfilter_unittest.cc b/pc/srtpfilter_unittest.cc
index a71762c..52f3afa 100644
--- a/pc/srtpfilter_unittest.cc
+++ b/pc/srtpfilter_unittest.cc
@@ -21,21 +21,21 @@
namespace rtc {
-static const std::string kTestKeyParams1 =
+static const char kTestKeyParams1[] =
"inline:WVNfX19zZW1jdGwgKCkgewkyMjA7fQp9CnVubGVz";
-static const std::string kTestKeyParams2 =
+static const char kTestKeyParams2[] =
"inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR";
-static const std::string kTestKeyParams3 =
+static const char kTestKeyParams3[] =
"inline:1234X19zZW1jdGwgKCkgewkyMjA7fQp9CnVubGVz";
-static const std::string kTestKeyParams4 =
+static const char kTestKeyParams4[] =
"inline:4567QCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR";
-static const std::string kTestKeyParamsGcm1 =
+static const char kTestKeyParamsGcm1[] =
"inline:e166KFlKzJsGW0d5apX+rrI05vxbrvMJEzFI14aTDCa63IRTlLK4iH66uOI=";
-static const std::string kTestKeyParamsGcm2 =
+static const char kTestKeyParamsGcm2[] =
"inline:6X0oCd55zfz4VgtOwsuqcFq61275PDYN5uwuu3p7ZUHbfUY2FMpdP4m2PEo=";
-static const std::string kTestKeyParamsGcm3 =
+static const char kTestKeyParamsGcm3[] =
"inline:YKlABGZWMgX32xuMotrG0v0T7G83veegaVzubQ==";
-static const std::string kTestKeyParamsGcm4 =
+static const char kTestKeyParamsGcm4[] =
"inline:gJ6tWoUym2v+/F6xjr7xaxiS3QbJJozl3ZD/0A==";
static const cricket::CryptoParams kTestCryptoParams1(
1, "AES_CM_128_HMAC_SHA1_80", kTestKeyParams1, "");
diff --git a/pc/srtptransport.cc b/pc/srtptransport.cc
index aa76aa2..0270da2 100644
--- a/pc/srtptransport.cc
+++ b/pc/srtptransport.cc
@@ -11,6 +11,7 @@
#include "pc/srtptransport.h"
#include <string>
+#include <vector>
#include "media/base/rtputils.h"
#include "pc/rtptransport.h"
diff --git a/pc/srtptransport.h b/pc/srtptransport.h
index 9f20e1d..03c353c 100644
--- a/pc/srtptransport.h
+++ b/pc/srtptransport.h
@@ -14,6 +14,7 @@
#include <memory>
#include <string>
#include <utility>
+#include <vector>
#include "pc/rtptransportinternal.h"
#include "pc/srtpfilter.h"
diff --git a/pc/srtptransport_unittest.cc b/pc/srtptransport_unittest.cc
index d3bc259..d551c83 100644
--- a/pc/srtptransport_unittest.cc
+++ b/pc/srtptransport_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <vector>
+
#include "pc/srtptransport.h"
#include "media/base/fakertp.h"
@@ -141,14 +143,14 @@
TestRtpAuthParams(srtp_transport1_.get(), cipher_suite_name);
} else {
ASSERT_TRUE(last_recv_packet2_.data());
- EXPECT_TRUE(
- memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len) == 0);
+ EXPECT_EQ(0,
+ memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
// Get the encrypted packet from underneath packet transport and verify
// the data is actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport1_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- original_rtp_data, rtp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ original_rtp_data, rtp_len));
}
// Do the same thing in the opposite direction;
@@ -158,12 +160,12 @@
TestRtpAuthParams(srtp_transport2_.get(), cipher_suite_name);
} else {
ASSERT_TRUE(last_recv_packet1_.data());
- EXPECT_TRUE(
- memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len) == 0);
+ EXPECT_EQ(0,
+ memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- original_rtp_data, rtp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ original_rtp_data, rtp_len));
}
}
@@ -186,25 +188,23 @@
ASSERT_TRUE(srtp_transport1_->SendRtcpPacket(&rtcp_packet1to2, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(last_recv_packet2_.data());
- EXPECT_TRUE(memcmp(last_recv_packet2_.data(), rtcp_packet_data, rtcp_len) ==
- 0);
+ EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), rtcp_packet_data, rtcp_len));
// Get the encrypted packet from underneath packet transport and verify the
// data is actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport1_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- rtcp_packet_data, rtcp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ rtcp_packet_data, rtcp_len));
// Do the same thing in the opposite direction;
ASSERT_TRUE(srtp_transport2_->SendRtcpPacket(&rtcp_packet2to1, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(last_recv_packet1_.data());
- EXPECT_TRUE(memcmp(last_recv_packet1_.data(), rtcp_packet_data, rtcp_len) ==
- 0);
+ EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), rtcp_packet_data, rtcp_len));
fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- rtcp_packet_data, rtcp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ rtcp_packet_data, rtcp_len));
}
void TestSendRecvPacket(bool enable_external_auth,
@@ -267,14 +267,13 @@
ASSERT_TRUE(srtp_transport1_->SendRtpPacket(&rtp_packet1to2, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(last_recv_packet2_.data());
- EXPECT_TRUE(memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len) ==
- 0);
+ EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
// Get the encrypted packet from underneath packet transport and verify the
// data and header extension are actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport1_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- original_rtp_data, rtp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ original_rtp_data, rtp_len));
CompareHeaderExtensions(
reinterpret_cast<const char*>(
fake_rtp_packet_transport->last_sent_packet()->data()),
@@ -285,12 +284,11 @@
ASSERT_TRUE(srtp_transport2_->SendRtpPacket(&rtp_packet2to1, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(last_recv_packet1_.data());
- EXPECT_TRUE(memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len) ==
- 0);
+ EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
- EXPECT_FALSE(memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
- original_rtp_data, rtp_len) == 0);
+ EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+ original_rtp_data, rtp_len));
CompareHeaderExtensions(
reinterpret_cast<const char*>(
fake_rtp_packet_transport->last_sent_packet()->data()),
diff --git a/pc/statscollector.cc b/pc/statscollector.cc
index 919ea46..15fe173 100644
--- a/pc/statscollector.cc
+++ b/pc/statscollector.cc
@@ -74,13 +74,14 @@
}
template <class TrackVector>
-void CreateTrackReports(const TrackVector& tracks, StatsCollection* reports,
- TrackIdMap& track_ids) {
+void CreateTrackReports(const TrackVector& tracks,
+ StatsCollection* reports,
+ TrackIdMap* track_ids) {
for (const auto& track : tracks) {
const std::string& track_id = track->id();
StatsReport* report = AddTrackReport(reports, track_id);
RTC_DCHECK(report != nullptr);
- track_ids[track_id] = report;
+ (*track_ids)[track_id] = report;
}
}
@@ -463,10 +464,10 @@
RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
RTC_DCHECK(stream != NULL);
- CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(),
- &reports_, track_ids_);
- CreateTrackReports<VideoTrackVector>(stream->GetVideoTracks(),
- &reports_, track_ids_);
+ CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(), &reports_,
+ &track_ids_);
+ CreateTrackReports<VideoTrackVector>(stream->GetVideoTracks(), &reports_,
+ &track_ids_);
}
void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track,
diff --git a/pc/statscollector.h b/pc/statscollector.h
index 5c90066..bdfe64b 100644
--- a/pc/statscollector.h
+++ b/pc/statscollector.h
@@ -16,6 +16,7 @@
#include <map>
#include <string>
+#include <utility>
#include <vector>
#include "api/mediastreaminterface.h"
diff --git a/pc/statscollector_unittest.cc b/pc/statscollector_unittest.cc
index 5e34f7f..48e8959 100644
--- a/pc/statscollector_unittest.cc
+++ b/pc/statscollector_unittest.cc
@@ -12,6 +12,7 @@
#include <algorithm>
#include <memory>
+#include <utility>
#include "pc/statscollector.h"
diff --git a/pc/test/fakeaudiocapturemodule.h b/pc/test/fakeaudiocapturemodule.h
index 25edea7..aa10efc 100644
--- a/pc/test/fakeaudiocapturemodule.h
+++ b/pc/test/fakeaudiocapturemodule.h
@@ -164,7 +164,7 @@
// exposed in which case the burden of proper instantiation would be put on
// the creator of a FakeAudioCaptureModule instance. To create an instance of
// this class use the Create(..) API.
- explicit FakeAudioCaptureModule();
+ FakeAudioCaptureModule();
// The destructor is protected because it is reference counted and should not
// be deleted directly.
virtual ~FakeAudioCaptureModule();
@@ -201,11 +201,11 @@
// Callback for playout and recording.
webrtc::AudioTransport* audio_callback_;
- bool recording_; // True when audio is being pushed from the instance.
- bool playing_; // True when audio is being pulled by the instance.
+ bool recording_; // True when audio is being pushed from the instance.
+ bool playing_; // True when audio is being pulled by the instance.
- bool play_is_initialized_; // True when the instance is ready to pull audio.
- bool rec_is_initialized_; // True when the instance is ready to push audio.
+ bool play_is_initialized_; // True when the instance is ready to pull audio.
+ bool rec_is_initialized_; // True when the instance is ready to push audio.
// Input to and output from RecordedDataIsAvailable(..) makes it possible to
// modify the current mic level. The implementation does not care about the
diff --git a/pc/test/fakedatachannelprovider.h b/pc/test/fakedatachannelprovider.h
index bafcb17..2ac4f94 100644
--- a/pc/test/fakedatachannelprovider.h
+++ b/pc/test/fakedatachannelprovider.h
@@ -11,6 +11,8 @@
#ifndef PC_TEST_FAKEDATACHANNELPROVIDER_H_
#define PC_TEST_FAKEDATACHANNELPROVIDER_H_
+#include <set>
+
#include "pc/datachannel.h"
#include "rtc_base/checks.h"
diff --git a/pc/test/fakevideotrackrenderer.h b/pc/test/fakevideotrackrenderer.h
index caec548..617261a 100644
--- a/pc/test/fakevideotrackrenderer.h
+++ b/pc/test/fakevideotrackrenderer.h
@@ -18,7 +18,7 @@
class FakeVideoTrackRenderer : public cricket::FakeVideoRenderer {
public:
- FakeVideoTrackRenderer(VideoTrackInterface* video_track)
+ explicit FakeVideoTrackRenderer(VideoTrackInterface* video_track)
: video_track_(video_track) {
video_track_->AddOrUpdateSink(this, rtc::VideoSinkWants());
}
diff --git a/pc/test/mock_datachannel.h b/pc/test/mock_datachannel.h
index 4a77a6b..20a4da6 100644
--- a/pc/test/mock_datachannel.h
+++ b/pc/test/mock_datachannel.h
@@ -11,6 +11,8 @@
#ifndef PC_TEST_MOCK_DATACHANNEL_H_
#define PC_TEST_MOCK_DATACHANNEL_H_
+#include <string>
+
#include "pc/datachannel.h"
#include "test/gmock.h"
diff --git a/pc/test/mock_peerconnection.h b/pc/test/mock_peerconnection.h
index 7944a4d..d57ada8 100644
--- a/pc/test/mock_peerconnection.h
+++ b/pc/test/mock_peerconnection.h
@@ -11,6 +11,7 @@
#ifndef PC_TEST_MOCK_PEERCONNECTION_H_
#define PC_TEST_MOCK_PEERCONNECTION_H_
+#include <memory>
#include <vector>
#include "call/call.h"
diff --git a/pc/test/mock_webrtcsession.h b/pc/test/mock_webrtcsession.h
index afd1473..e027d0b 100644
--- a/pc/test/mock_webrtcsession.h
+++ b/pc/test/mock_webrtcsession.h
@@ -28,10 +28,11 @@
// http://crbug.com/428099.
explicit MockWebRtcSession(cricket::ChannelManager* channel_manager,
const cricket::MediaConfig& media_config)
- : WebRtcSession(nullptr /* Call */,
+ : WebRtcSession(
+ nullptr, // call
channel_manager,
media_config,
- nullptr, // event_log
+ nullptr, // event_log
rtc::Thread::Current(),
rtc::Thread::Current(),
rtc::Thread::Current(),
diff --git a/pc/test/mockpeerconnectionobservers.h b/pc/test/mockpeerconnectionobservers.h
index 845dbc7..9128b8c 100644
--- a/pc/test/mockpeerconnectionobservers.h
+++ b/pc/test/mockpeerconnectionobservers.h
@@ -16,6 +16,8 @@
#include <memory>
#include <string>
+#include <utility>
+#include <vector>
#include "api/datachannelinterface.h"
#include "api/jsepicecandidate.h"
diff --git a/pc/test/peerconnectiontestwrapper.cc b/pc/test/peerconnectiontestwrapper.cc
index cc91217..6d28ef8 100644
--- a/pc/test/peerconnectiontestwrapper.cc
+++ b/pc/test/peerconnectiontestwrapper.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
#include <utility>
#include "p2p/base/fakeportallocator.h"
diff --git a/pc/test/peerconnectiontestwrapper.h b/pc/test/peerconnectiontestwrapper.h
index d9488b4..aadaa8e 100644
--- a/pc/test/peerconnectiontestwrapper.h
+++ b/pc/test/peerconnectiontestwrapper.h
@@ -12,6 +12,7 @@
#define PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
#include <memory>
+#include <string>
#include "api/peerconnectioninterface.h"
#include "api/test/fakeconstraints.h"
@@ -52,7 +53,7 @@
void OnRemoveStream(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
void OnDataChannel(
- rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override ;
+ rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
diff --git a/pc/test/testsdpstrings.h b/pc/test/testsdpstrings.h
index 2c9912e..fc884a1 100644
--- a/pc/test/testsdpstrings.h
+++ b/pc/test/testsdpstrings.h
@@ -76,7 +76,7 @@
"a=candidate:5 2 UDP 1694302206 74.95.2.170 45468 typ srflx raddr"
" 10.0.254.2 rport 61232\r\n"
#endif
- ;
+ ; // NOLINT(whitespace/semicolon)
// Audio SDP with a limited set of audio codecs.
static const char kAudioSdp[] =
diff --git a/pc/videocapturertracksource.cc b/pc/videocapturertracksource.cc
index 97c0994..2426cea 100644
--- a/pc/videocapturertracksource.cc
+++ b/pc/videocapturertracksource.cc
@@ -12,6 +12,7 @@
#include <cstdlib>
#include <string>
+#include <utility>
#include <vector>
#include "api/mediaconstraintsinterface.h"
diff --git a/pc/videocapturertracksource_unittest.cc b/pc/videocapturertracksource_unittest.cc
index c64c83e..a492643 100644
--- a/pc/videocapturertracksource_unittest.cc
+++ b/pc/videocapturertracksource_unittest.cc
@@ -10,6 +10,7 @@
#include <memory>
#include <string>
+#include <utility>
#include <vector>
#include "api/test/fakeconstraints.h"
diff --git a/pc/videotrack.cc b/pc/videotrack.cc
index 718c0d6..bd6d9c2 100644
--- a/pc/videotrack.cc
+++ b/pc/videotrack.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
+
#include "pc/videotrack.h"
#include "rtc_base/refcountedobject.h"
-#include <string>
-
namespace webrtc {
VideoTrack::VideoTrack(const std::string& label,
diff --git a/pc/webrtcsdp.cc b/pc/webrtcsdp.cc
index 47e4298..f053bf5 100644
--- a/pc/webrtcsdp.cc
+++ b/pc/webrtcsdp.cc
@@ -15,7 +15,9 @@
#include <stdio.h>
#include <algorithm>
+#include <map>
#include <memory>
+#include <set>
#include <string>
#include <unordered_map>
#include <vector>
@@ -174,7 +176,7 @@
static const char kReturn = '\r';
static const char kLineBreak[] = "\r\n";
-// TODO: Generate the Session and Time description
+// TODO(deadbeef): Generate the Session and Time description
// instead of hardcoding.
static const char kSessionVersion[] = "v=0";
// RFC 4566
@@ -675,7 +677,7 @@
// likely to work, typically IPv4 relay.
// RFC 5245
// The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP).
-// TODO: Decide the default destination in webrtcsession and
+// TODO(deadbeef): Decide the default destination in webrtcsession and
// pass it down via SessionDescription.
static void GetDefaultDestination(
const std::vector<Candidate>& candidates,
@@ -1179,7 +1181,8 @@
bool encrypted = false;
if (uri == RtpExtension::kEncryptHeaderExtensionsUri) {
// RFC 6904
- // a=extmap:<value["/"<direction>] urn:ietf:params:rtp-hdrext:encrypt <URI> <extensionattributes>
+ // a=extmap:<value["/"<direction>] urn:ietf:params:rtp-hdrext:encrypt <URI>
+ // <extensionattributes>
const size_t expected_min_fields_encrypted = expected_min_fields + 1;
if (fields.size() < expected_min_fields_encrypted) {
return ParseFailedExpectMinFieldNum(line, expected_min_fields_encrypted,
@@ -1207,7 +1210,7 @@
if (content_info == NULL || message == NULL) {
return;
}
- // TODO: Rethink if we should use sprintfn instead of stringstream.
+ // TODO(deadbeef): Rethink if we should use sprintfn instead of stringstream.
// According to the style guide, streams should only be used for logging.
// http://google-styleguide.googlecode.com/svn/
// trunk/cppguide.xml?showone=Streams#Streams
@@ -2768,7 +2771,7 @@
}
if (!IsLineType(line, kLineTypeAttributes)) {
- // TODO: Handle other lines if needed.
+ // TODO(deadbeef): Handle other lines if needed.
LOG(LS_INFO) << "Ignored line: " << line;
continue;
}
@@ -2892,7 +2895,7 @@
} else if (HasAttribute(line, kAttributeXGoogleFlag)) {
// Experimental attribute. Conference mode activates more aggressive
// AEC and NS settings.
- // TODO: expose API to set these directly.
+ // TODO(deadbeef): expose API to set these directly.
std::string flag_value;
if (!GetValue(line, kAttributeXGoogleFlag, &flag_value, error)) {
return false;
diff --git a/pc/webrtcsdp_unittest.cc b/pc/webrtcsdp_unittest.cc
index 75dac25..e532324 100644
--- a/pc/webrtcsdp_unittest.cc
+++ b/pc/webrtcsdp_unittest.cc
@@ -2598,7 +2598,7 @@
// No crash is a pass.
}
-void MutateJsepSctpPort(JsepSessionDescription& jdesc,
+void MutateJsepSctpPort(JsepSessionDescription* jdesc,
const SessionDescription& desc) {
// take our pre-built session description and change the SCTP port.
cricket::SessionDescription* mutant = desc.Copy();
@@ -2611,7 +2611,7 @@
dcdesc->set_codecs(codecs);
// note: mutant's owned by jdesc now.
- ASSERT_TRUE(jdesc.Initialize(mutant, kSessionId, kSessionVersion));
+ ASSERT_TRUE(jdesc->Initialize(mutant, kSessionId, kSessionVersion));
mutant = NULL;
}
@@ -2621,7 +2621,7 @@
// First setup the expected JsepSessionDescription.
JsepSessionDescription jdesc(kDummyString);
- MutateJsepSctpPort(jdesc, desc_);
+ MutateJsepSctpPort(&jdesc, desc_);
// Then get the deserialized JsepSessionDescription.
std::string sdp_with_data = kSdpString;
@@ -2641,7 +2641,7 @@
AddSctpDataChannel(use_sctpmap);
JsepSessionDescription jdesc(kDummyString);
- MutateJsepSctpPort(jdesc, desc_);
+ MutateJsepSctpPort(&jdesc, desc_);
// We need to test the deserialized JsepSessionDescription from
// kSdpSctpDataChannelStringWithSctpPort for
diff --git a/pc/webrtcsession.cc b/pc/webrtcsession.cc
index ebdcf4e..477b1ff 100644
--- a/pc/webrtcsession.cc
+++ b/pc/webrtcsession.cc
@@ -1872,7 +1872,7 @@
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::CreateSctpTransport_n,
this, content->name, transport_name))) {
return false;
- };
+ }
} else {
bool require_rtcp_mux =
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
diff --git a/pc/webrtcsession.h b/pc/webrtcsession.h
index 07202ee..09e5a3e 100644
--- a/pc/webrtcsession.h
+++ b/pc/webrtcsession.h
@@ -11,6 +11,7 @@
#ifndef PC_WEBRTCSESSION_H_
#define PC_WEBRTCSESSION_H_
+#include <map>
#include <memory>
#include <set>
#include <string>
diff --git a/pc/webrtcsessiondescriptionfactory.cc b/pc/webrtcsessiondescriptionfactory.cc
index bbab646..69c629e 100644
--- a/pc/webrtcsessiondescriptionfactory.cc
+++ b/pc/webrtcsessiondescriptionfactory.cc
@@ -10,7 +10,10 @@
#include "pc/webrtcsessiondescriptionfactory.h"
+#include <algorithm>
+#include <string>
#include <utility>
+#include <vector>
#include "api/jsep.h"
#include "api/jsepsessiondescription.h"
diff --git a/pc/webrtcsessiondescriptionfactory.h b/pc/webrtcsessiondescriptionfactory.h
index 5c3e18e..7ad418f 100644
--- a/pc/webrtcsessiondescriptionfactory.h
+++ b/pc/webrtcsessiondescriptionfactory.h
@@ -12,6 +12,8 @@
#define PC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#include <memory>
+#include <queue>
+#include <string>
#include "api/peerconnectioninterface.h"
#include "p2p/base/transportdescriptionfactory.h"
