diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index b2ff3bcf49..94d51b7708 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc @@ -54,7 +54,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, ~AcmReceiverTest() {} - void SetUp() { + virtual void SetUp() OVERRIDE { ASSERT_TRUE(receiver_.get() != NULL); ASSERT_TRUE(acm_.get() != NULL); for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) { @@ -75,7 +75,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, rtp_header_.type.Audio.isCNG = false; } - void TearDown() { + virtual void TearDown() OVERRIDE { } void InsertOnePacketOfSilence(int codec_id) { @@ -125,7 +125,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, uint32_t timestamp, const uint8_t* payload_data, uint16_t payload_len_bytes, - const RTPFragmentationHeader* fragmentation) { + const RTPFragmentationHeader* fragmentation) OVERRIDE { if (frame_type == kFrameEmpty) return 0; diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h index 5e9bd977d3..db5d9e5519 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h @@ -42,7 +42,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource { // Returns the next encoded packet. Returns NULL if the test duration was // exceeded. Ownership of the packet is handed over to the caller. // Inherited from PacketSource. - Packet* NextPacket(); + virtual Packet* NextPacket() OVERRIDE; // Inherited from AudioPacketizationCallback. virtual int32_t SendData( diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc index 621f183d98..009218d194 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -124,9 +124,9 @@ class AudioCodingModuleTest : public ::testing::Test { ~AudioCodingModuleTest() {} - void TearDown() {} + void TearDown() OVERRIDE {} - void SetUp() { + void SetUp() OVERRIDE { acm_.reset(AudioCodingModule::Create(id_, clock_)); RegisterCodec(); @@ -309,7 +309,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { clock_ = fake_clock_.get(); } - void SetUp() { + virtual void SetUp() OVERRIDE { AudioCodingModuleTest::SetUp(); StartThreads(); } @@ -321,7 +321,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { ASSERT_TRUE(pull_audio_thread_->Start(thread_id)); } - void TearDown() { + virtual void TearDown() OVERRIDE { AudioCodingModuleTest::TearDown(); pull_audio_thread_->Stop(); send_thread_->Stop(); @@ -437,7 +437,7 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { ~AcmIsacMtTest() {} - void SetUp() { + virtual void SetUp() OVERRIDE { AudioCodingModuleTest::SetUp(); // Set up input audio source to read from specified file, loop after 5 @@ -460,7 +460,7 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { StartThreads(); } - virtual void RegisterCodec() { + virtual void RegisterCodec() OVERRIDE { COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz); AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1); codec_.pltype = kPayloadType; @@ -471,7 +471,7 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); } - void InsertPacket() { + virtual void InsertPacket() OVERRIDE { int num_calls = packet_cb_.num_calls(); // Store locally for thread safety. if (num_calls > last_packet_number_) { // Get the new payload out from the callback handler. @@ -490,16 +490,16 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); } - void InsertAudio() { + virtual void InsertAudio() OVERRIDE { memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms); AudioCodingModuleTest::InsertAudio(); } - void Encode() { ASSERT_GE(acm_->Process(), 0); } + virtual void Encode() OVERRIDE { ASSERT_GE(acm_->Process(), 0); } // This method is the same as AudioCodingModuleMtTest::TestDone(), but here // it is using the constants defined in this class (i.e., shorter test run). - virtual bool TestDone() { + virtual bool TestDone() OVERRIDE { if (packet_cb_.num_calls() > kNumPackets) { CriticalSectionScoped lock(crit_sect_.get()); if (pull_audio_count_ > kNumPullCalls) { @@ -694,7 +694,7 @@ class AcmSenderBitExactness : public ::testing::Test, // Returns a pointer to the next packet. Returns NULL if the source is // depleted (i.e., the test duration is exceeded), or if an error occurred. // Inherited from test::PacketSource. - test::Packet* NextPacket() OVERRIDE { + virtual test::Packet* NextPacket() OVERRIDE { // Get the next packet from AcmSendTest. Ownership of |packet| is // transferred to this method. test::Packet* packet = send_test_->NextPacket(); diff --git a/webrtc/modules/audio_coding/main/test/ACMTest.cc b/webrtc/modules/audio_coding/main/test/ACMTest.cc deleted file mode 100644 index dbbdade803..0000000000 --- a/webrtc/modules/audio_coding/main/test/ACMTest.cc +++ /dev/null @@ -1,13 +0,0 @@ -/* - * Copyright (c) 2011 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 - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "ACMTest.h" - -ACMTest::~ACMTest() {} diff --git a/webrtc/modules/audio_coding/main/test/ACMTest.h b/webrtc/modules/audio_coding/main/test/ACMTest.h index 767add1715..f73961f5e5 100644 --- a/webrtc/modules/audio_coding/main/test/ACMTest.h +++ b/webrtc/modules/audio_coding/main/test/ACMTest.h @@ -14,7 +14,7 @@ class ACMTest { public: ACMTest() {} - virtual ~ACMTest() = 0; + virtual ~ACMTest() {} virtual void Perform() = 0; }; diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h index 7611c9d6e3..cdb99c03e2 100644 --- a/webrtc/modules/audio_coding/main/test/Channel.h +++ b/webrtc/modules/audio_coding/main/test/Channel.h @@ -50,10 +50,11 @@ class Channel : public AudioPacketizationCallback { Channel(int16_t chID = -1); ~Channel(); - int32_t SendData(const FrameType frameType, const uint8_t payloadType, - const uint32_t timeStamp, const uint8_t* payloadData, - const uint16_t payloadSize, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + const FrameType frameType, const uint8_t payloadType, + const uint32_t timeStamp, const uint8_t* payloadData, + const uint16_t payloadSize, + const RTPFragmentationHeader* fragmentation) OVERRIDE; void RegisterReceiverACM(AudioCodingModule *acm); diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h index dbe3f0cb35..693c96e013 100644 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h +++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h @@ -29,10 +29,11 @@ class TestPacketization : public AudioPacketizationCallback { public: TestPacketization(RTPStream *rtpStream, uint16_t frequency); ~TestPacketization(); - virtual int32_t SendData(const FrameType frameType, const uint8_t payloadType, - const uint32_t timeStamp, const uint8_t* payloadData, - const uint16_t payloadSize, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + const FrameType frameType, const uint8_t payloadType, + const uint32_t timeStamp, const uint8_t* payloadData, + const uint16_t payloadSize, + const RTPFragmentationHeader* fragmentation) OVERRIDE; private: static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, @@ -100,7 +101,7 @@ class EncodeDecodeTest : public ACMTest { public: EncodeDecodeTest(); explicit EncodeDecodeTest(int testMode); - virtual void Perform(); + virtual void Perform() OVERRIDE; uint16_t _playoutFreq; uint8_t _testMode; diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h index 9b6d5fcafe..460553bebd 100644 --- a/webrtc/modules/audio_coding/main/test/RTPFile.h +++ b/webrtc/modules/audio_coding/main/test/RTPFile.h @@ -65,14 +65,14 @@ class RTPBuffer : public RTPStream { ~RTPBuffer(); - void Write(const uint8_t payloadType, const uint32_t timeStamp, - const int16_t seqNo, const uint8_t* payloadData, - const uint16_t payloadSize, uint32_t frequency); + virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, + const int16_t seqNo, const uint8_t* payloadData, + const uint16_t payloadSize, uint32_t frequency) OVERRIDE; - uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, - uint16_t payloadSize, uint32_t* offset); + virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, + uint16_t payloadSize, uint32_t* offset) OVERRIDE; - virtual bool EndOfFile() const; + virtual bool EndOfFile() const OVERRIDE; private: RWLockWrapper* _queueRWLock; @@ -97,14 +97,14 @@ class RTPFile : public RTPStream { void ReadHeader(); - void Write(const uint8_t payloadType, const uint32_t timeStamp, - const int16_t seqNo, const uint8_t* payloadData, - const uint16_t payloadSize, uint32_t frequency); + virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, + const int16_t seqNo, const uint8_t* payloadData, + const uint16_t payloadSize, uint32_t frequency) OVERRIDE; - uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, - uint16_t payloadSize, uint32_t* offset); + virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, + uint16_t payloadSize, uint32_t* offset) OVERRIDE; - bool EndOfFile() const { + virtual bool EndOfFile() const OVERRIDE { return _rtpEOF; } diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h index 73eac47f21..2fbf9ef0f3 100644 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h +++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h @@ -28,10 +28,11 @@ class TestPack : public AudioPacketizationCallback { void RegisterReceiverACM(AudioCodingModule* acm); - int32_t SendData(FrameType frame_type, uint8_t payload_type, - uint32_t timestamp, const uint8_t* payload_data, - uint16_t payload_size, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + FrameType frame_type, uint8_t payload_type, + uint32_t timestamp, const uint8_t* payload_data, + uint16_t payload_size, + const RTPFragmentationHeader* fragmentation) OVERRIDE; uint16_t payload_size(); uint32_t timestamp_diff(); @@ -52,7 +53,7 @@ class TestAllCodecs : public ACMTest { explicit TestAllCodecs(int test_mode); ~TestAllCodecs(); - void Perform(); + virtual void Perform() OVERRIDE; private: // The default value of '-1' indicates that the registration is based only on diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h index 9cb70e916f..8aefa7fc40 100644 --- a/webrtc/modules/audio_coding/main/test/TestStereo.h +++ b/webrtc/modules/audio_coding/main/test/TestStereo.h @@ -33,12 +33,13 @@ class TestPackStereo : public AudioPacketizationCallback { void RegisterReceiverACM(AudioCodingModule* acm); - virtual int32_t SendData(const FrameType frame_type, - const uint8_t payload_type, - const uint32_t timestamp, - const uint8_t* payload_data, - const uint16_t payload_size, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + const FrameType frame_type, + const uint8_t payload_type, + const uint32_t timestamp, + const uint8_t* payload_data, + const uint16_t payload_size, + const RTPFragmentationHeader* fragmentation) OVERRIDE; uint16_t payload_size(); uint32_t timestamp_diff(); @@ -63,7 +64,7 @@ class TestStereo : public ACMTest { explicit TestStereo(int test_mode); ~TestStereo(); - void Perform(); + virtual void Perform() OVERRIDE; private: // The default value of '-1' indicates that the registration is based only on // codec name and a sampling frequncy matching is not required. This is useful diff --git a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc index ecd1e79d9c..7cd2466f68 100644 --- a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc +++ b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc @@ -35,10 +35,11 @@ class DualStreamTest : public AudioPacketizationCallback, void ApiTest(); - int32_t SendData(FrameType frameType, uint8_t payload_type, - uint32_t timestamp, const uint8_t* payload_data, - uint16_t payload_size, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + FrameType frameType, uint8_t payload_type, + uint32_t timestamp, const uint8_t* payload_data, + uint16_t payload_size, + const RTPFragmentationHeader* fragmentation) OVERRIDE; void Perform(bool start_in_sync, int num_channels_input); diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h index bc7734b467..d3dff185a0 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_impl.h +++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h @@ -82,7 +82,7 @@ class NetEqImpl : public webrtc::NetEq { virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, const uint8_t* payload, int length_bytes, - uint32_t receive_timestamp); + uint32_t receive_timestamp) OVERRIDE; // Inserts a sync-packet into packet queue. Sync-packets are decoded to // silence and are intended to keep AV-sync intact in an event of long packet @@ -94,7 +94,7 @@ class NetEqImpl : public webrtc::NetEq { // can be implied by inserting a sync-packet. // Returns kOk on success, kFail on failure. virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header, - uint32_t receive_timestamp); + uint32_t receive_timestamp) OVERRIDE; // Instructs NetEq to deliver 10 ms of audio data. The data is written to // |output_audio|, which can hold (at least) |max_length| elements. @@ -106,12 +106,12 @@ class NetEqImpl : public webrtc::NetEq { // Returns kOK on success, or kFail in case of an error. virtual int GetAudio(size_t max_length, int16_t* output_audio, int* samples_per_channel, int* num_channels, - NetEqOutputType* type); + NetEqOutputType* type) OVERRIDE; // Associates |rtp_payload_type| with |codec| and stores the information in // the codec database. Returns kOK on success, kFail on failure. virtual int RegisterPayloadType(enum NetEqDecoder codec, - uint8_t rtp_payload_type); + uint8_t rtp_payload_type) OVERRIDE; // Provides an externally created decoder object |decoder| to insert in the // decoder database. The decoder implements a decoder of type |codec| and @@ -119,80 +119,81 @@ class NetEqImpl : public webrtc::NetEq { // failure. virtual int RegisterExternalDecoder(AudioDecoder* decoder, enum NetEqDecoder codec, - uint8_t rtp_payload_type); + uint8_t rtp_payload_type) OVERRIDE; // Removes |rtp_payload_type| from the codec database. Returns 0 on success, // -1 on failure. - virtual int RemovePayloadType(uint8_t rtp_payload_type); + virtual int RemovePayloadType(uint8_t rtp_payload_type) OVERRIDE; - virtual bool SetMinimumDelay(int delay_ms); + virtual bool SetMinimumDelay(int delay_ms) OVERRIDE; - virtual bool SetMaximumDelay(int delay_ms); + virtual bool SetMaximumDelay(int delay_ms) OVERRIDE; - virtual int LeastRequiredDelayMs() const; + virtual int LeastRequiredDelayMs() const OVERRIDE; - virtual int SetTargetDelay() { return kNotImplemented; } + virtual int SetTargetDelay() OVERRIDE { return kNotImplemented; } - virtual int TargetDelay() { return kNotImplemented; } + virtual int TargetDelay() OVERRIDE { return kNotImplemented; } - virtual int CurrentDelay() { return kNotImplemented; } + virtual int CurrentDelay() OVERRIDE { return kNotImplemented; } // Sets the playout mode to |mode|. - virtual void SetPlayoutMode(NetEqPlayoutMode mode); + virtual void SetPlayoutMode(NetEqPlayoutMode mode) OVERRIDE; // Returns the current playout mode. - virtual NetEqPlayoutMode PlayoutMode() const; + virtual NetEqPlayoutMode PlayoutMode() const OVERRIDE; // Writes the current network statistics to |stats|. The statistics are reset // after the call. - virtual int NetworkStatistics(NetEqNetworkStatistics* stats); + virtual int NetworkStatistics(NetEqNetworkStatistics* stats) OVERRIDE; // Writes the last packet waiting times (in ms) to |waiting_times|. The number // of values written is no more than 100, but may be smaller if the interface // is polled again before 100 packets has arrived. - virtual void WaitingTimes(std::vector* waiting_times); + virtual void WaitingTimes(std::vector* waiting_times) OVERRIDE; // Writes the current RTCP statistics to |stats|. The statistics are reset // and a new report period is started with the call. - virtual void GetRtcpStatistics(RtcpStatistics* stats); + virtual void GetRtcpStatistics(RtcpStatistics* stats) OVERRIDE; // Same as RtcpStatistics(), but does not reset anything. - virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats); + virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) OVERRIDE; // Enables post-decode VAD. When enabled, GetAudio() will return // kOutputVADPassive when the signal contains no speech. - virtual void EnableVad(); + virtual void EnableVad() OVERRIDE; // Disables post-decode VAD. - virtual void DisableVad(); + virtual void DisableVad() OVERRIDE; - virtual bool GetPlayoutTimestamp(uint32_t* timestamp); + virtual bool GetPlayoutTimestamp(uint32_t* timestamp) OVERRIDE; - virtual int SetTargetNumberOfChannels() { return kNotImplemented; } + virtual int SetTargetNumberOfChannels() OVERRIDE { return kNotImplemented; } - virtual int SetTargetSampleRate() { return kNotImplemented; } + virtual int SetTargetSampleRate() OVERRIDE { return kNotImplemented; } // Returns the error code for the last occurred error. If no error has // occurred, 0 is returned. - virtual int LastError(); + virtual int LastError() OVERRIDE; // Returns the error code last returned by a decoder (audio or comfort noise). // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check // this method to get the decoder's error code. - virtual int LastDecoderError(); + virtual int LastDecoderError() OVERRIDE; // Flushes both the packet buffer and the sync buffer. - virtual void FlushBuffers(); + virtual void FlushBuffers() OVERRIDE; virtual void PacketBufferStatistics(int* current_num_packets, - int* max_num_packets) const; + int* max_num_packets) const OVERRIDE; // Get sequence number and timestamp of the latest RTP. // This method is to facilitate NACK. - virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const; + virtual int DecodedRtpInfo(int* sequence_number, + uint32_t* timestamp) const OVERRIDE; // This accessor method is only intended for testing purposes. - virtual const SyncBuffer* sync_buffer_for_test() const; + const SyncBuffer* sync_buffer_for_test() const; protected: static const int kOutputSizeMs = 10; diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 8ea307962c..a1a1bf0de2 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -329,7 +329,6 @@ '<@(audio_coding_defines)', ], 'sources': [ - 'audio_coding/main/test/ACMTest.cc', 'audio_coding/main/test/APITest.cc', 'audio_coding/main/test/Channel.cc', 'audio_coding/main/test/dual_stream_unittest.cc', diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h index ead1e22345..e04372d9a5 100644 --- a/webrtc/modules/utility/source/coder.h +++ b/webrtc/modules/utility/source/coder.h @@ -27,11 +27,11 @@ public: int32_t SetEncodeCodec( const CodecInst& codecInst, - ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); + ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); int32_t SetDecodeCodec( const CodecInst& codecInst, - ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); + ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz, const int8_t* incomingPayload, int32_t payloadLength); @@ -42,12 +42,13 @@ public: uint32_t& encodedLengthInBytes); protected: - virtual int32_t SendData(FrameType frameType, - uint8_t payloadType, - uint32_t timeStamp, - const uint8_t* payloadData, - uint16_t payloadSize, - const RTPFragmentationHeader* fragmentation); + virtual int32_t SendData( + FrameType frameType, + uint8_t payloadType, + uint32_t timeStamp, + const uint8_t* payloadData, + uint16_t payloadSize, + const RTPFragmentationHeader* fragmentation) OVERRIDE; private: scoped_ptr _acm;