in rtp_rtcp module:

fixed build/namespaces lint errors
  fixed readability/namespace lint errors

BUG=webrtc:5277
R=mflodman,stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1506823002

Cr-Commit-Position: refs/heads/master@{#10978}

webrtc/modules/rtp_rtcp/source/byte_io.h

@@ -51,17 +51,14 @@ namespace webrtc {
 // platform that doesn't use two's complement, implement conversion to/from
 // wire format.
 
-namespace {
-inline void AssertTwosComplement() {
-  // Assume the if any one signed integer type is two's complement, then all
-  // other will be too.
-  static_assert(
-      (-1 & 0x03) == 0x03,
-      "Only two's complement representation of signed integers supported.");
-}
+// Assume the if any one signed integer type is two's complement, then all
+// other will be too.
+static_assert(
+    (-1 & 0x03) == 0x03,
+    "Only two's complement representation of signed integers supported.");
+
 // Plain const char* won't work for static_assert, use #define instead.
 #define kSizeErrorMsg "Byte size must be less than or equal to data type size."
-}
 
 // Utility class for getting the unsigned equivalent of a signed type.
 template <typename T>

webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h

@@ -27,7 +27,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskBursty1_1[2] = {
   0x80, 0x00
@@ -756,5 +757,6 @@ const uint8_t** kPacketMaskBurstyTbl[12] = {
   kPacketMaskBursty12
 };
 
-}  // namespace
+}  // namespace fec_private_tables
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_

webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h

@@ -17,7 +17,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskRandom10_1[2] = {
   0xff, 0xc0
@@ -24518,5 +24519,6 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
   kPacketMaskRandom48
 };
 
-}  // namespace
+}  // namespace fec_private_tables
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_

webrtc/modules/rtp_rtcp/source/fec_test_helper.h

@@ -54,6 +54,6 @@ class FrameGenerator {
   uint16_t seq_num_;
   uint32_t timestamp_;
 };
-}
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_

webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc

@@ -17,6 +17,8 @@
 #include "webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h"
 
 namespace {
+using webrtc::fec_private_tables::kPacketMaskBurstyTbl;
+using webrtc::fec_private_tables::kPacketMaskRandomTbl;
 
 // Allow for different modes of protection for packets in UEP case.
 enum ProtectionMode {

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

@@ -24,7 +24,7 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
-using namespace webrtc;
+namespace webrtc {
 
 const int kVideoNackListSize = 30;
 const uint32_t kTestSsrc = 3456;
@@ -338,3 +338,5 @@ TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
+
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc

@@ -21,10 +21,9 @@
 #include "webrtc/test/null_transport.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
 namespace {
 
-using namespace webrtc;
-
 class TestTransport : public Transport {
  public:
   TestTransport(RTCPReceiver* rtcp_receiver) : rtcp_receiver_(rtcp_receiver) {}
@@ -130,3 +129,4 @@ TEST_F(RtcpFormatRembTest, TestCompund) {
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb));
 }
 }  // namespace
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -23,8 +23,13 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 
 namespace webrtc {
-using namespace RTCPUtility;
-using namespace RTCPHelp;
+using RTCPHelp::RTCPPacketInformation;
+using RTCPHelp::RTCPReceiveInformation;
+using RTCPHelp::RTCPReportBlockInformation;
+using RTCPUtility::kBtVoipMetric;
+using RTCPUtility::RTCPCnameInformation;
+using RTCPUtility::RTCPPacketReportBlockItem;
+using RTCPUtility::RTCPPacketTypes;
 
 // The number of RTCP time intervals needed to trigger a timeout.
 const int kRrTimeoutIntervals = 3;

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc

@@ -17,7 +17,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
-using namespace RTCPHelp;
+namespace RTCPHelp {
 
 RTCPPacketInformation::RTCPPacketInformation()
     : rtcpPacketTypeFlags(0),
@@ -190,4 +190,5 @@ void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
     const uint32_t minimumSize) {
   TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
 }
+}  // namespace RTCPHelp
 }  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_utility.cc

@@ -41,8 +41,8 @@ void NackStats::ReportRequest(uint16_t sequence_number) {
 
 uint32_t MidNtp(uint32_t ntp_sec, uint32_t ntp_frac) {
   return (ntp_sec << 16) + (ntp_frac >> 16);
-}  // end RTCPUtility
 }
+}  // namespace RTCPUtility
 
 // RTCPParserV2 : currently read only
 RTCPUtility::RTCPParserV2::RTCPParserV2(const uint8_t* rtcpData,

webrtc/modules/rtp_rtcp/source/rtcp_utility.h

@@ -487,6 +487,6 @@ class RTCPPacketIterator {
 
   RtcpCommonHeader _header;
 };
-}  // RTCPUtility
+}  // namespace RTCPUtility
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_

webrtc/modules/rtp_rtcp/source/rtp_header_extension.h

@@ -112,7 +112,7 @@ class RtpHeaderExtensionMap {
   int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
   std::map<uint8_t, HeaderExtension*> extensionMap_;
 };
-}
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc

@@ -265,4 +265,4 @@ size_t Vp8PartitionAggregator::CalcNumberOfFragments(
   return num_fragments;
 }
 
-}  // namespace
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h

@@ -132,6 +132,6 @@ class Vp8PartitionAggregator {
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
 };
-}  // namespace
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_

webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc

@@ -209,4 +209,4 @@ TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
                 1600, kMTU, 1, 900, 1000));
 }
 
-}  // namespace
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc

@@ -24,7 +24,10 @@
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
-using namespace webrtc;
+using webrtc::CriticalSectionScoped;
+using webrtc::CriticalSectionWrapper;
+using webrtc::EventWrapper;
+using webrtc::TickTime;
 
 #ifdef MATLAB
 MatlabEngine eng;

webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h

@@ -21,9 +21,6 @@ class TestLoadGenerator;
 namespace webrtc {
 class CriticalSectionWrapper;
 class EventWrapper;
-}
-
-using namespace webrtc;
 
 #define MAX_BITRATE_KBPS 50000
 
@@ -149,5 +146,5 @@ private:
     int64_t _lastTime;
 
 };
-
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

@@ -15,9 +15,8 @@
 
 #include "webrtc/test/null_transport.h"
 
-using namespace webrtc;
-
 namespace webrtc {
+
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
@@ -77,7 +76,6 @@ int32_t TestRtpReceiver::OnReceivedPayloadData(
   payload_size_ = payload_size;
   return 0;
 }
-}  // namespace webrtc
 
 class RtpRtcpAPITest : public ::testing::Test {
  protected:
@@ -188,3 +186,5 @@ TEST_F(RtpRtcpAPITest, RtxReceiver) {
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
+
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -19,8 +19,8 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
-using namespace webrtc;
-
+namespace webrtc {
+namespace {
 #define test_rate 64000u
 
 class VerifyingAudioReceiver : public NullRtpData {
@@ -349,3 +349,6 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
     module1->Process();
   }
 }
+
+}  // namespace
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

@@ -20,7 +20,8 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
-using namespace webrtc;
+namespace webrtc {
+namespace {
 
 const uint64_t kTestPictureId = 12345678;
 const uint8_t kSliPictureId = 156;
@@ -266,3 +267,6 @@ TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
   EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
   EXPECT_EQ(0u, report_blocks[0].fractionLost);
 }
+
+}  // namespace
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h

@@ -10,7 +10,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
 #define WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
 
-namespace {
+namespace webrtc {
 
 // Maximum number of media packets allowed in this test. The burst mask types
 // are currently defined up to (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
@@ -187,5 +187,5 @@ const float kMaxResidualLossBurstyMask[kNumberCodes] = {
   0.009657f
 };
 
-}  // namespace
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_

webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc

@@ -22,7 +22,6 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 
@@ -32,7 +31,11 @@
 //#define VERBOSE_OUTPUT
 
 namespace webrtc {
+namespace fec_private_tables {
+extern const uint8_t** kPacketMaskBurstyTbl[12];
+}
 namespace test {
+using fec_private_tables::kPacketMaskBurstyTbl;
 
 void ReceivePackets(
     ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
@@ -94,10 +97,6 @@ TEST(FecTest, FecTest) {
   const FecMaskType kMaskTypes[] = { kFecMaskRandom, kFecMaskBursty };
   const int kNumFecMaskTypes = sizeof(kMaskTypes) / sizeof(*kMaskTypes);
 
-  // TODO(pbos): Fix this. Hack to prevent a warning
-  // ('-Wunneeded-internal-declaration') from clang.
-  (void) kPacketMaskBurstyTbl;
-
   // Maximum number of media packets allowed for the mask type.
   const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets,
       sizeof(kPacketMaskBurstyTbl) / sizeof(*kPacketMaskBurstyTbl)};

webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc

@@ -59,13 +59,6 @@ enum { kMaxNumberMediaPackets = 48 };
 // Maximum number of media packets allowed for each mask type.
 const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets, 12};
 
-// Maximum number of media packets allowed in this test. The burst mask types
-// are currently defined up to (k=12,m=12).
-const int kMaxMediaPacketsTest = 12;
-
-// Maximum number of FEC codes considered in this test.
-const int kNumberCodes = kMaxMediaPacketsTest * (kMaxMediaPacketsTest + 1) / 2;
-
 // Maximum gap size for characterizing the consecutiveness of the loss.
 const int kMaxGapSize = 2 * kMaxMediaPacketsTest;