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Enable cpplint in pc/

Browse Source 
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}
This commit is contained in:
Steve Anton 2017-10-30 09:57:42 -07:00 committed by Commit Bot
parent ef1140eec0
commit 36b29d1df3
54 changed files with 155 additions and 105 deletions
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1
PRESUBMIT.py
View File

@ -30,7 +30,6 @@ CPPLINT_BLACKLIST = [
'modules/utility',
'modules/video_capture',
'p2p',
'pc',
'rtc_base',
'sdk/android/src/jni',
'sdk/objc',

2
pc/channel_unittest.cc
View File

@ -525,7 +525,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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();

1
pc/channelmanager.cc
View File

@ -11,6 +11,7 @@
#include "pc/channelmanager.h"
#include <algorithm>
#include <utility>
#include "media/base/device.h"
#include "media/base/rtpdataengine.h"

2
pc/currentspeakermonitor_unittest.cc
View File

@ -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"

2
pc/datachannel.cc
View File

@ -180,7 +180,7 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
case webrtc::InternalDataChannelInit::kAcker:
handshake_state_ = kHandshakeShouldSendAck;
break;
};
}
// Try to connect to the transport in case the transport channel already
// exists.

1
pc/datachannel_unittest.cc
View File

@ -9,6 +9,7 @@
*/
#include <memory>
#include <vector>
#include "pc/datachannel.h"
#include "pc/sctputils.h"

5
pc/externalhmac.cc
View File

@ -80,7 +80,7 @@ srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
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_alloc(srtp_auth_t** a,
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;

12
pc/mediasession.cc
View File

@ -2430,9 +2430,9 @@ const DataContentDescription* GetFirstDataContentDescription(
// 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 @@ ContentInfo* GetFirstMediaContent(ContentInfos& contents,
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 @@ static ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
return nullptr;
}
return GetFirstMediaContent(sdesc->contents(), media_type);
return GetFirstMediaContent(&sdesc->contents(), media_type);
}
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {

8
pc/mediasession.h
View File

@ -618,10 +618,10 @@ const DataContentDescription* GetFirstDataContentDescription(
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);

19
pc/mediasession_unittest.cc
View File

@ -256,7 +256,16 @@ std::vector<MediaDescriptionOptions>::iterator FindFirstMediaDescriptionByMid(
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 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
}
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 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
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 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
}
}
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 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
}
}
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));

2
pc/mediastream.cc
View File

@ -25,7 +25,7 @@ static typename V::iterator FindTrack(V* vector,
}
}
return it;
};
}
rtc::scoped_refptr<MediaStream> MediaStream::Create(
const std::string& label) {

4
pc/peerconnection.cc
View File

@ -206,8 +206,8 @@ template <class SENDER,
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) {

2
pc/peerconnectionendtoend_unittest.cc
View File

@ -184,7 +184,7 @@ std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
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:

2
pc/peerconnectionfactory.cc
View File

@ -106,7 +106,7 @@ PeerConnectionFactory::PeerConnectionFactory(
}
}
// 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);
}

6
pc/peerconnectionfactory.h
View File

@ -49,7 +49,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
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 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
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.

1
pc/peerconnectionfactory_unittest.cc
View File

@ -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"

1
pc/peerconnectionwrapper.cc
View File

@ -13,6 +13,7 @@
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/jsepsessiondescription.h"
#include "media/base/fakevideocapturer.h"

1
pc/peerconnectionwrapper.h
View File

@ -14,6 +14,7 @@
#include <functional>
#include <memory>
#include <string>
#include <vector>
#include "api/peerconnectioninterface.h"
#include "pc/test/mockpeerconnectionobservers.h"

19
pc/rtcstats_integrationtest.cc
View File

@ -56,15 +56,16 @@ class RTCStatsReportTraceListener {
}
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));

1
pc/rtcstatscollector.cc
View File

@ -12,6 +12,7 @@
#include <memory>
#include <sstream>
#include <string>
#include <utility>
#include <vector>

1
pc/rtcstatscollector.h
View File

@ -14,6 +14,7 @@
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "api/optional.h"

11
pc/rtcstatscollector_unittest.cc
View File

@ -14,6 +14,7 @@
#include <memory>
#include <ostream>
#include <string>
#include <utility>
#include <vector>
#include "api/jsepsessiondescription.h"
@ -181,9 +182,8 @@ class FakeAudioTrackForStats
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 @@ class FakeVideoTrackForStats
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;

2
pc/rtpreceiver.cc
View File

@ -10,6 +10,8 @@
#include "pc/rtpreceiver.h"
#include <vector>
#include "api/mediastreamtrackproxy.h"
#include "api/videosourceproxy.h"
#include "pc/audiotrack.h"

1
pc/rtpreceiver.h
View File

@ -18,6 +18,7 @@
#include <stdint.h>
#include <string>
#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpreceiverinterface.h"

2
pc/rtpsender.cc
View File

@ -10,6 +10,8 @@
#include "pc/rtpsender.h"
#include <vector>
#include "api/mediastreaminterface.h"
#include "pc/localaudiosource.h"
#include "rtc_base/checks.h"

1
pc/rtpsender.h
View File

@ -17,6 +17,7 @@
#include <memory>
#include <string>
#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpsenderinterface.h"

1
pc/sdputils.cc
View File

@ -10,6 +10,7 @@
#include "pc/sdputils.h"
#include <string>
#include <utility>
#include "api/jsepsessiondescription.h"

1
pc/sdputils.h
View File

@ -13,6 +13,7 @@
#include <functional>
#include <memory>
#include <string>
#include "api/jsep.h"
#include "p2p/base/sessiondescription.h"

16
pc/srtpfilter_unittest.cc
View File

@ -21,21 +21,21 @@ using cricket::CS_REMOTE;
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, "");

1
pc/srtptransport.cc
View File

@ -11,6 +11,7 @@
#include "pc/srtptransport.h"
#include <string>
#include <vector>
#include "media/base/rtputils.h"
#include "pc/rtptransport.h"

1
pc/srtptransport.h
View File

@ -14,6 +14,7 @@
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "pc/rtptransportinternal.h"
#include "pc/srtpfilter.h"

46
pc/srtptransport_unittest.cc
View File

@ -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 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
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 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
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 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
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 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
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 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
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()),

15
pc/statscollector.cc
View File

@ -74,13 +74,14 @@ StatsReport* AddTrackReport(StatsCollection* reports,
}
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 @@ void StatsCollector::AddStream(MediaStreamInterface* stream) {
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,

1
pc/statscollector.h
View File

@ -16,6 +16,7 @@
#include <map>
#include <string>
#include <utility>
#include <vector>
#include "api/mediastreaminterface.h"

1
pc/statscollector_unittest.cc
View File

@ -12,6 +12,7 @@
#include <algorithm>
#include <memory>
#include <utility>
#include "pc/statscollector.h"

10
pc/test/fakeaudiocapturemodule.h
View File

@ -164,7 +164,7 @@ class FakeAudioCaptureModule
// 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 @@ class FakeAudioCaptureModule
// 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

2
pc/test/fakedatachannelprovider.h
View File

@ -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"

2
pc/test/fakevideotrackrenderer.h
View File

@ -18,7 +18,7 @@ namespace webrtc {
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());
}

2
pc/test/mock_datachannel.h
View File

@ -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"

1
pc/test/mock_peerconnection.h
View File

@ -11,6 +11,7 @@
#ifndef PC_TEST_MOCK_PEERCONNECTION_H_
#define PC_TEST_MOCK_PEERCONNECTION_H_
#include <memory>
#include <vector>
#include "call/call.h"

5
pc/test/mock_webrtcsession.h
View File

@ -28,10 +28,11 @@ class MockWebRtcSession : public webrtc::WebRtcSession {
// 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(),

2
pc/test/mockpeerconnectionobservers.h
View File

@ -16,6 +16,8 @@
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/datachannelinterface.h"
#include "api/jsepicecandidate.h"

1
pc/test/peerconnectiontestwrapper.cc
View File

@ -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"

3
pc/test/peerconnectiontestwrapper.h
View File

@ -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 @@ class PeerConnectionTestWrapper
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 {}

2
pc/test/testsdpstrings.h
View File

@ -76,7 +76,7 @@ static const char kFireFoxSdpOffer[] =
"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[] =

1
pc/videocapturertracksource.cc
View File

@ -12,6 +12,7 @@
#include <cstdlib>
#include <string>
#include <utility>
#include <vector>
#include "api/mediaconstraintsinterface.h"

1
pc/videocapturertracksource_unittest.cc
View File

@ -10,6 +10,7 @@
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/test/fakeconstraints.h"

4
pc/videotrack.cc
View File

@ -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,

15
pc/webrtcsdp.cc
View File

@ -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 kNewLine = '\n';
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 @@ static int GetCandidatePreferenceFromType(const std::string& type) {
// 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 ParseExtmap(const std::string& line,
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 @@ void BuildMediaDescription(const ContentInfo* content_info,
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 @@ bool ParseContent(const std::string& message,
}
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 @@ bool ParseContent(const std::string& message,
} 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;

8
pc/webrtcsdp_unittest.cc
View File

@ -2598,7 +2598,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithCorruptedSctpDataChannels) {
// 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 @@ void MutateJsepSctpPort(JsepSessionDescription& jdesc,
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 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelAndUnusualPort) {
// 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 @@ TEST_F(WebRtcSdpTest,
AddSctpDataChannel(use_sctpmap);
JsepSessionDescription jdesc(kDummyString);
MutateJsepSctpPort(jdesc, desc_);
MutateJsepSctpPort(&jdesc, desc_);
// We need to test the deserialized JsepSessionDescription from
// kSdpSctpDataChannelStringWithSctpPort for

2
pc/webrtcsession.cc
View File

@ -1872,7 +1872,7 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
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;

1
pc/webrtcsession.h
View File

@ -11,6 +11,7 @@
#ifndef PC_WEBRTCSESSION_H_
#define PC_WEBRTCSESSION_H_
#include <map>
#include <memory>
#include <set>
#include <string>

3
pc/webrtcsessiondescriptionfactory.cc
View File

@ -10,7 +10,10 @@
#include "pc/webrtcsessiondescriptionfactory.h"
#include <algorithm>
#include <string>
#include <utility>
#include <vector>
#include "api/jsep.h"
#include "api/jsepsessiondescription.h"

2
pc/webrtcsessiondescriptionfactory.h
View File

@ -12,6 +12,8 @@
#define PC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#include <memory>
#include <queue>
#include <string>
#include "api/peerconnectioninterface.h"
#include "p2p/base/transportdescriptionfactory.h"