From ac891627c63b37247a313bb9099d374a33679edd Mon Sep 17 00:00:00 2001 From: "pbos@webrtc.org" Date: Tue, 9 Apr 2013 17:40:15 +0000 Subject: [PATCH] WebRtc_Word32 -> int32_t in audio_conference_mixer/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1306004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3804 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../interface/audio_conference_mixer.h | 36 +++-- .../audio_conference_mixer_defines.h | 29 ++-- .../source/audio_conference_mixer_impl.cc | 66 ++++----- .../source/audio_conference_mixer_impl.h | 68 ++++----- .../source/audio_frame_manipulator.cc | 8 +- .../source/level_indicator.cc | 14 +- .../source/level_indicator.h | 12 +- .../source/memory_pool.h | 24 +-- .../source/memory_pool_posix.h | 30 ++-- .../source/memory_pool_win.h | 24 +-- .../source/time_scheduler.cc | 20 +-- .../source/time_scheduler.h | 12 +- .../test/FunctionTest/functionTest.cc | 139 +++++++++--------- .../test/FunctionTest/functionTest.h | 88 +++++------ 14 files changed, 286 insertions(+), 284 deletions(-) diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h index 9ffac2d2d..ef7432438 100644 --- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h +++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h @@ -40,44 +40,42 @@ public: virtual ~AudioConferenceMixer() {} // Module functions - virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0; - virtual WebRtc_Word32 TimeUntilNextProcess() = 0 ; - virtual WebRtc_Word32 Process() = 0; + virtual int32_t ChangeUniqueId(const int32_t id) = 0; + virtual int32_t TimeUntilNextProcess() = 0 ; + virtual int32_t Process() = 0; // Register/unregister a callback class for receiving the mixed audio. - virtual WebRtc_Word32 RegisterMixedStreamCallback( + virtual int32_t RegisterMixedStreamCallback( AudioMixerOutputReceiver& receiver) = 0; - virtual WebRtc_Word32 UnRegisterMixedStreamCallback() = 0; + virtual int32_t UnRegisterMixedStreamCallback() = 0; // Register/unregister a callback class for receiving status information. - virtual WebRtc_Word32 RegisterMixerStatusCallback( + virtual int32_t RegisterMixerStatusCallback( AudioMixerStatusReceiver& mixerStatusCallback, - const WebRtc_UWord32 amountOf10MsBetweenCallbacks) = 0; - virtual WebRtc_Word32 UnRegisterMixerStatusCallback() = 0; + const uint32_t amountOf10MsBetweenCallbacks) = 0; + virtual int32_t UnRegisterMixerStatusCallback() = 0; // Add/remove participants as candidates for mixing. - virtual WebRtc_Word32 SetMixabilityStatus( - MixerParticipant& participant, - const bool mixable) = 0; + virtual int32_t SetMixabilityStatus(MixerParticipant& participant, + const bool mixable) = 0; // mixable is set to true if a participant is a candidate for mixing. - virtual WebRtc_Word32 MixabilityStatus( - MixerParticipant& participant, - bool& mixable) = 0; + virtual int32_t MixabilityStatus(MixerParticipant& participant, + bool& mixable) = 0; // Inform the mixer that the participant should always be mixed and not // count toward the number of mixed participants. Note that a participant // must have been added to the mixer (by calling SetMixabilityStatus()) // before this function can be successfully called. - virtual WebRtc_Word32 SetAnonymousMixabilityStatus( - MixerParticipant& participant, const bool mixable) = 0; + virtual int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant, + const bool mixable) = 0; // mixable is set to true if the participant is mixed anonymously. - virtual WebRtc_Word32 AnonymousMixabilityStatus( - MixerParticipant& participant, bool& mixable) = 0; + virtual int32_t AnonymousMixabilityStatus(MixerParticipant& participant, + bool& mixable) = 0; // Set the minimum sampling frequency at which to mix. The mixing algorithm // may still choose to mix at a higher samling frequency to avoid // downsampling of audio contributing to the mixed audio. - virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq) = 0; + virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0; protected: AudioConferenceMixer() {} diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h index 718470deb..9b82d5ed7 100644 --- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h +++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h @@ -26,15 +26,14 @@ public: // audio every time it's called. // // If it returns -1, the frame will not be added to the mix. - virtual WebRtc_Word32 GetAudioFrame(const WebRtc_Word32 id, - AudioFrame& audioFrame) = 0; + virtual int32_t GetAudioFrame(const int32_t id, AudioFrame& audioFrame) = 0; // mixed will be set to true if the participant was mixed this mix iteration - WebRtc_Word32 IsMixed(bool& mixed) const; + int32_t IsMixed(bool& mixed) const; // This function specifies the sampling frequency needed for the AudioFrame // for future GetAudioFrame(..) calls. - virtual WebRtc_Word32 NeededFrequency(const WebRtc_Word32 id) = 0; + virtual int32_t NeededFrequency(const int32_t id) = 0; MixHistory* _mixHistory; protected: @@ -45,8 +44,8 @@ protected: // Container struct for participant statistics. struct ParticipantStatistics { - WebRtc_Word32 participant; - WebRtc_Word32 level; + int32_t participant; + int32_t level; }; class AudioMixerStatusReceiver @@ -55,20 +54,20 @@ public: // Callback function that provides an array of ParticipantStatistics for the // participants that were mixed last mix iteration. virtual void MixedParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size) = 0; + const uint32_t size) = 0; // Callback function that provides an array of the ParticipantStatistics for // the participants that had a positiv VAD last mix iteration. virtual void VADPositiveParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size) = 0; + const uint32_t size) = 0; // Callback function that provides the audio level of the mixed audio frame // from the last mix iteration. virtual void MixedAudioLevel( - const WebRtc_Word32 id, - const WebRtc_UWord32 level) = 0; + const int32_t id, + const uint32_t level) = 0; protected: AudioMixerStatusReceiver() {} virtual ~AudioMixerStatusReceiver() {} @@ -80,10 +79,10 @@ public: // This callback function provides the mixed audio for this mix iteration. // Note that uniqueAudioFrames is an array of AudioFrame pointers with the // size according to the size parameter. - virtual void NewMixedAudio(const WebRtc_Word32 id, + virtual void NewMixedAudio(const int32_t id, const AudioFrame& generalAudioFrame, const AudioFrame** uniqueAudioFrames, - const WebRtc_UWord32 size) = 0; + const uint32_t size) = 0; protected: AudioMixerOutputReceiver() {} virtual ~AudioMixerOutputReceiver() {} @@ -95,7 +94,7 @@ public: // This callback function provides the mix decision for this mix iteration. // mixerList is a list of elements of the type // [int,MixerParticipant*] - virtual void NewAudioToRelay(const WebRtc_Word32 id, + virtual void NewAudioToRelay(const int32_t id, const MapWrapper& mixerList) = 0; protected: AudioRelayReceiver() {} diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc index 3fedc487a..29ff9f828 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc +++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc @@ -71,7 +71,7 @@ MixerParticipant::~MixerParticipant() delete _mixHistory; } -WebRtc_Word32 MixerParticipant::IsMixed(bool& mixed) const +int32_t MixerParticipant::IsMixed(bool& mixed) const { return _mixHistory->IsMixed(mixed); } @@ -85,20 +85,20 @@ MixHistory::~MixHistory() { } -WebRtc_Word32 MixHistory::IsMixed(bool& mixed) const +int32_t MixHistory::IsMixed(bool& mixed) const { mixed = _isMixed; return 0; } -WebRtc_Word32 MixHistory::WasMixed(bool& wasMixed) const +int32_t MixHistory::WasMixed(bool& wasMixed) const { // Was mixed is the same as is mixed depending on perspective. This function // is for the perspective of AudioConferenceMixerImpl. return IsMixed(wasMixed); } -WebRtc_Word32 MixHistory::SetIsMixed(const bool mixed) +int32_t MixHistory::SetIsMixed(const bool mixed) { _isMixed = mixed; return 0; @@ -202,16 +202,16 @@ AudioConferenceMixerImpl::~AudioConferenceMixerImpl() assert(_audioFramePool == NULL); } -WebRtc_Word32 AudioConferenceMixerImpl::ChangeUniqueId(const WebRtc_Word32 id) +int32_t AudioConferenceMixerImpl::ChangeUniqueId(const int32_t id) { _id = id; return 0; } // Process should be called every kProcessPeriodicityInMs ms -WebRtc_Word32 AudioConferenceMixerImpl::TimeUntilNextProcess() +int32_t AudioConferenceMixerImpl::TimeUntilNextProcess() { - WebRtc_Word32 timeUntilNextProcess = 0; + int32_t timeUntilNextProcess = 0; CriticalSectionScoped cs(_crit.get()); if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) { @@ -224,9 +224,9 @@ WebRtc_Word32 AudioConferenceMixerImpl::TimeUntilNextProcess() return timeUntilNextProcess; } -WebRtc_Word32 AudioConferenceMixerImpl::Process() +int32_t AudioConferenceMixerImpl::Process() { - WebRtc_UWord32 remainingParticipantsAllowedToMix = + uint32_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants; { CriticalSectionScoped cs(_crit.get()); @@ -244,7 +244,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process() { CriticalSectionScoped cs(_cbCrit.get()); - WebRtc_Word32 lowFreq = GetLowestMixingFrequency(); + int32_t lowFreq = GetLowestMixingFrequency(); // SILK can run in 12 kHz and 24 kHz. These frequencies are not // supported so use the closest higher frequency to not lose any // information. @@ -322,7 +322,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process() bool timeForMixerCallback = false; int retval = 0; - WebRtc_Word32 audioLevel = 0; + int32_t audioLevel = 0; { CriticalSectionScoped cs(_crit.get()); @@ -415,7 +415,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process() return retval; } -WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixedStreamCallback( +int32_t AudioConferenceMixerImpl::RegisterMixedStreamCallback( AudioMixerOutputReceiver& mixReceiver) { CriticalSectionScoped cs(_cbCrit.get()); @@ -427,7 +427,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixedStreamCallback( return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() +int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() { CriticalSectionScoped cs(_cbCrit.get()); if(_mixReceiver == NULL) @@ -438,7 +438,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::SetOutputFrequency( +int32_t AudioConferenceMixerImpl::SetOutputFrequency( const Frequency frequency) { CriticalSectionScoped cs(_crit.get()); @@ -481,9 +481,9 @@ bool AudioConferenceMixerImpl::SetNumLimiterChannels(int numChannels) return true; } -WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback( +int32_t AudioConferenceMixerImpl::RegisterMixerStatusCallback( AudioMixerStatusReceiver& mixerStatusCallback, - const WebRtc_UWord32 amountOf10MsBetweenCallbacks) + const uint32_t amountOf10MsBetweenCallbacks) { if(amountOf10MsBetweenCallbacks == 0) { @@ -513,7 +513,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback( return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() +int32_t AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() { { CriticalSectionScoped cs(_crit.get()); @@ -532,7 +532,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus( +int32_t AudioConferenceMixerImpl::SetMixabilityStatus( MixerParticipant& participant, const bool mixable) { @@ -542,7 +542,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus( // participant is in the _participantList if it is being mixed. SetAnonymousMixabilityStatus(participant, false); } - WebRtc_UWord32 numMixedParticipants; + uint32_t numMixedParticipants; { CriticalSectionScoped cs(_cbCrit.get()); const bool isMixed = @@ -589,7 +589,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus( return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::MixabilityStatus( +int32_t AudioConferenceMixerImpl::MixabilityStatus( MixerParticipant& participant, bool& mixable) { @@ -598,7 +598,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::MixabilityStatus( return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( +int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( MixerParticipant& participant, const bool anonymous) { CriticalSectionScoped cs(_cbCrit.get()); @@ -638,7 +638,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( 0 : -1; } -WebRtc_Word32 AudioConferenceMixerImpl::AnonymousMixabilityStatus( +int32_t AudioConferenceMixerImpl::AnonymousMixabilityStatus( MixerParticipant& participant, bool& mixable) { CriticalSectionScoped cs(_cbCrit.get()); @@ -647,7 +647,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::AnonymousMixabilityStatus( return 0; } -WebRtc_Word32 AudioConferenceMixerImpl::SetMinimumMixingFrequency( +int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency( Frequency freq) { // Make sure that only allowed sampling frequencies are used. Use closest @@ -676,7 +676,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetMinimumMixingFrequency( // Check all AudioFrames that are to be mixed. The highest sampling frequency // found is the lowest that can be used without losing information. -WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequency() +int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() { const int participantListFrequency = GetLowestMixingFrequencyFromList(_participantList); @@ -696,16 +696,16 @@ WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequency() return highestFreq; } -WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList( +int32_t AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList( ListWrapper& mixList) { - WebRtc_Word32 highestFreq = 8000; + int32_t highestFreq = 8000; ListItem* item = mixList.First(); while(item) { MixerParticipant* participant = static_cast(item->GetItem()); - const WebRtc_Word32 neededFrequency = participant->NeededFrequency(_id); + const int32_t neededFrequency = participant->NeededFrequency(_id); if(neededFrequency > highestFreq) { highestFreq = neededFrequency; @@ -719,12 +719,12 @@ void AudioConferenceMixerImpl::UpdateToMix( ListWrapper& mixList, ListWrapper& rampOutList, MapWrapper& mixParticipantList, - WebRtc_UWord32& maxAudioFrameCounter) + uint32_t& maxAudioFrameCounter) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)", maxAudioFrameCounter); - const WebRtc_UWord32 mixListStartSize = mixList.GetSize(); + const uint32_t mixListStartSize = mixList.GetSize(); ListWrapper activeList; // Elements are AudioFrames // Struct needed by the passive lists to keep track of which AudioFrame // belongs to which MixerParticipant. @@ -790,7 +790,7 @@ void AudioConferenceMixerImpl::UpdateToMix( // mixed. Only keep the ones with the highest energy. ListItem* replaceItem = NULL; CalculateEnergy(*audioFrame); - WebRtc_UWord32 lowestEnergy = audioFrame->energy_; + uint32_t lowestEnergy = audioFrame->energy_; ListItem* activeItem = activeList.First(); while(activeItem) @@ -1108,13 +1108,13 @@ bool AudioConferenceMixerImpl::RemoveParticipantFromList( return false; } -WebRtc_Word32 AudioConferenceMixerImpl::MixFromList( +int32_t AudioConferenceMixerImpl::MixFromList( AudioFrame& mixedAudio, const ListWrapper& audioFrameList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "MixFromList(mixedAudio, audioFrameList)"); - WebRtc_UWord32 position = 0; + uint32_t position = 0; ListItem* item = audioFrameList.First(); if(item == NULL) { @@ -1159,7 +1159,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::MixFromList( } // TODO(andrew): consolidate this function with MixFromList. -WebRtc_Word32 AudioConferenceMixerImpl::MixAnonomouslyFromList( +int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList( AudioFrame& mixedAudio, const ListWrapper& audioFrameList) { diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h index c38afd000..c76dbf783 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h +++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h @@ -32,14 +32,14 @@ public: ~MixHistory(); // MixerParticipant function - WebRtc_Word32 IsMixed(bool& mixed) const; + int32_t IsMixed(bool& mixed) const; // Sets wasMixed to true if the participant was mixed previous mix // iteration. - WebRtc_Word32 WasMixed(bool& wasMixed) const; + int32_t WasMixed(bool& wasMixed) const; // Updates the mixed status. - WebRtc_Word32 SetIsMixed(const bool mixed); + int32_t SetIsMixed(const bool mixed); void ResetMixedStatus(); private: @@ -59,32 +59,32 @@ public: bool Init(); // Module functions - virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id); - virtual WebRtc_Word32 TimeUntilNextProcess(); - virtual WebRtc_Word32 Process(); + virtual int32_t ChangeUniqueId(const int32_t id); + virtual int32_t TimeUntilNextProcess(); + virtual int32_t Process(); // AudioConferenceMixer functions - virtual WebRtc_Word32 RegisterMixedStreamCallback( + virtual int32_t RegisterMixedStreamCallback( AudioMixerOutputReceiver& mixReceiver); - virtual WebRtc_Word32 UnRegisterMixedStreamCallback(); - virtual WebRtc_Word32 RegisterMixerStatusCallback( + virtual int32_t UnRegisterMixedStreamCallback(); + virtual int32_t RegisterMixerStatusCallback( AudioMixerStatusReceiver& mixerStatusCallback, - const WebRtc_UWord32 amountOf10MsBetweenCallbacks); - virtual WebRtc_Word32 UnRegisterMixerStatusCallback(); - virtual WebRtc_Word32 SetMixabilityStatus(MixerParticipant& participant, - const bool mixable); - virtual WebRtc_Word32 MixabilityStatus(MixerParticipant& participant, - bool& mixable); - virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq); - virtual WebRtc_Word32 SetAnonymousMixabilityStatus( + const uint32_t amountOf10MsBetweenCallbacks); + virtual int32_t UnRegisterMixerStatusCallback(); + virtual int32_t SetMixabilityStatus(MixerParticipant& participant, + const bool mixable); + virtual int32_t MixabilityStatus(MixerParticipant& participant, + bool& mixable); + virtual int32_t SetMinimumMixingFrequency(Frequency freq); + virtual int32_t SetAnonymousMixabilityStatus( MixerParticipant& participant, const bool mixable); - virtual WebRtc_Word32 AnonymousMixabilityStatus( + virtual int32_t AnonymousMixabilityStatus( MixerParticipant& participant, bool& mixable); private: enum{DEFAULT_AUDIO_FRAME_POOLSIZE = 50}; // Set/get mix frequency - WebRtc_Word32 SetOutputFrequency(const Frequency frequency); + int32_t SetOutputFrequency(const Frequency frequency); Frequency OutputFrequency() const; // Must be called whenever an audio frame indicates the number of channels @@ -101,12 +101,12 @@ private: // should be ramped out over this AudioFrame to avoid audio discontinuities. void UpdateToMix(ListWrapper& mixList, ListWrapper& rampOutList, MapWrapper& mixParticipantList, - WebRtc_UWord32& maxAudioFrameCounter); + uint32_t& maxAudioFrameCounter); // Return the lowest mixing frequency that can be used without having to // downsample any audio. - WebRtc_Word32 GetLowestMixingFrequency(); - WebRtc_Word32 GetLowestMixingFrequencyFromList(ListWrapper& mixList); + int32_t GetLowestMixingFrequency(); + int32_t GetLowestMixingFrequencyFromList(ListWrapper& mixList); // Return the AudioFrames that should be mixed anonymously. void GetAdditionalAudio(ListWrapper& additionalFramesList); @@ -139,31 +139,31 @@ private: ListWrapper& participantList); // Mix the AudioFrames stored in audioFrameList into mixedAudio. - WebRtc_Word32 MixFromList( + int32_t MixFromList( AudioFrame& mixedAudio, const ListWrapper& audioFrameList); // Mix the AudioFrames stored in audioFrameList into mixedAudio. No // record will be kept of this mix (e.g. the corresponding MixerParticipants // will not be marked as IsMixed() - WebRtc_Word32 MixAnonomouslyFromList(AudioFrame& mixedAudio, - const ListWrapper& audioFrameList); + int32_t MixAnonomouslyFromList(AudioFrame& mixedAudio, + const ListWrapper& audioFrameList); bool LimitMixedAudio(AudioFrame& mixedAudio); // Scratch memory // Note that the scratch memory may only be touched in the scope of // Process(). - WebRtc_UWord32 _scratchParticipantsToMixAmount; + uint32_t _scratchParticipantsToMixAmount; ParticipantStatistics _scratchMixedParticipants[ kMaximumAmountOfMixedParticipants]; - WebRtc_UWord32 _scratchVadPositiveParticipantsAmount; + uint32_t _scratchVadPositiveParticipantsAmount; ParticipantStatistics _scratchVadPositiveParticipants[ kMaximumAmountOfMixedParticipants]; scoped_ptr _crit; scoped_ptr _cbCrit; - WebRtc_Word32 _id; + int32_t _id; Frequency _minimumMixingFreq; @@ -171,13 +171,13 @@ private: AudioMixerOutputReceiver* _mixReceiver; AudioMixerStatusReceiver* _mixerStatusCallback; - WebRtc_UWord32 _amountOf10MsBetweenCallbacks; - WebRtc_UWord32 _amountOf10MsUntilNextCallback; + uint32_t _amountOf10MsBetweenCallbacks; + uint32_t _amountOf10MsUntilNextCallback; bool _mixerStatusCb; // The current sample frequency and sample size when mixing. Frequency _outputFrequency; - WebRtc_UWord16 _sampleSize; + uint16_t _sampleSize; // Memory pool to avoid allocating/deallocating AudioFrames MemoryPool* _audioFramePool; @@ -186,9 +186,9 @@ private: ListWrapper _participantList; // May be mixed. ListWrapper _additionalParticipantList; // Always mixed, anonomously. - WebRtc_UWord32 _numMixedParticipants; + uint32_t _numMixedParticipants; - WebRtc_UWord32 _timeStamp; + uint32_t _timeStamp; // Metronome class. TimeScheduler _timeScheduler; @@ -198,7 +198,7 @@ private: // Counter keeping track of concurrent calls to process. // Note: should never be higher than 1 or lower than 0. - WebRtc_Word16 _processCalls; + int16_t _processCalls; // Used for inhibiting saturation in mixing. scoped_ptr _limiter; diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc index 65f8dc073..9545caba7 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc +++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc @@ -60,8 +60,8 @@ void RampIn(AudioFrame& audioFrame) assert(rampSize <= audioFrame.samples_per_channel_); for(int i = 0; i < rampSize; i++) { - audioFrame.data_[i] = static_cast - (rampArray[i] * audioFrame.data_[i]); + audioFrame.data_[i] = static_cast(rampArray[i] * + audioFrame.data_[i]); } } @@ -71,8 +71,8 @@ void RampOut(AudioFrame& audioFrame) for(int i = 0; i < rampSize; i++) { const int rampPos = rampSize - 1 - i; - audioFrame.data_[i] = static_cast - (rampArray[rampPos] * audioFrame.data_[i]); + audioFrame.data_[i] = static_cast(rampArray[rampPos] * + audioFrame.data_[i]); } memset(&audioFrame.data_[rampSize], 0, (audioFrame.samples_per_channel_ - rampSize) * diff --git a/webrtc/modules/audio_conference_mixer/source/level_indicator.cc b/webrtc/modules/audio_conference_mixer/source/level_indicator.cc index 799a47ddd..7afb30da9 100644 --- a/webrtc/modules/audio_conference_mixer/source/level_indicator.cc +++ b/webrtc/modules/audio_conference_mixer/source/level_indicator.cc @@ -12,7 +12,7 @@ namespace webrtc { // Array for adding smothing to level changes (ad-hoc). -const WebRtc_UWord32 perm[] = +const uint32_t perm[] = {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9}; LevelIndicator::LevelIndicator() @@ -27,11 +27,11 @@ LevelIndicator::~LevelIndicator() } // Level is based on the highest absolute value for all samples. -void LevelIndicator::ComputeLevel(const WebRtc_Word16* speech, - const WebRtc_UWord16 nrOfSamples) +void LevelIndicator::ComputeLevel(const int16_t* speech, + const uint16_t nrOfSamples) { - WebRtc_Word32 min = 0; - for(WebRtc_UWord32 i = 0; i < nrOfSamples; i++) + int32_t min = 0; + for(uint32_t i = 0; i < nrOfSamples; i++) { if(_max < speech[i]) { @@ -52,7 +52,7 @@ void LevelIndicator::ComputeLevel(const WebRtc_Word16* speech, if(_count == TICKS_BEFORE_CALCULATION) { // Highest sample value maps directly to a level. - WebRtc_Word32 position = _max / 1000; + int32_t position = _max / 1000; if ((position == 0) && (_max > 250)) { @@ -68,7 +68,7 @@ void LevelIndicator::ComputeLevel(const WebRtc_Word16* speech, } } -WebRtc_Word32 LevelIndicator::GetLevel() +int32_t LevelIndicator::GetLevel() { return _currentLevel; } diff --git a/webrtc/modules/audio_conference_mixer/source/level_indicator.h b/webrtc/modules/audio_conference_mixer/source/level_indicator.h index bdcdf8eb7..cf17d0ea0 100644 --- a/webrtc/modules/audio_conference_mixer/source/level_indicator.h +++ b/webrtc/modules/audio_conference_mixer/source/level_indicator.h @@ -23,14 +23,14 @@ public: ~LevelIndicator(); // Updates the level. - void ComputeLevel(const WebRtc_Word16* speech, - const WebRtc_UWord16 nrOfSamples); + void ComputeLevel(const int16_t* speech, + const uint16_t nrOfSamples); - WebRtc_Word32 GetLevel(); + int32_t GetLevel(); private: - WebRtc_Word32 _max; - WebRtc_UWord32 _count; - WebRtc_UWord32 _currentLevel; + int32_t _max; + uint32_t _count; + uint32_t _currentLevel; }; } // namespace webrtc diff --git a/webrtc/modules/audio_conference_mixer/source/memory_pool.h b/webrtc/modules/audio_conference_mixer/source/memory_pool.h index caf5d933f..e9961cd95 100644 --- a/webrtc/modules/audio_conference_mixer/source/memory_pool.h +++ b/webrtc/modules/audio_conference_mixer/source/memory_pool.h @@ -28,26 +28,26 @@ class MemoryPool { public: // Factory method, constructor disabled. - static WebRtc_Word32 CreateMemoryPool(MemoryPool*& memoryPool, - WebRtc_UWord32 initialPoolSize); + static int32_t CreateMemoryPool(MemoryPool*& memoryPool, + uint32_t initialPoolSize); // Try to delete the memory pool. Fail with return value -1 if there is // outstanding memory. - static WebRtc_Word32 DeleteMemoryPool( + static int32_t DeleteMemoryPool( MemoryPool*& memoryPool); // Get/return unused memory. - WebRtc_Word32 PopMemory(MemoryType*& memory); - WebRtc_Word32 PushMemory(MemoryType*& memory); + int32_t PopMemory(MemoryType*& memory); + int32_t PushMemory(MemoryType*& memory); private: - MemoryPool(WebRtc_Word32 initialPoolSize); + MemoryPool(int32_t initialPoolSize); ~MemoryPool(); MemoryPoolImpl* _ptrImpl; }; template -MemoryPool::MemoryPool(WebRtc_Word32 initialPoolSize) +MemoryPool::MemoryPool(int32_t initialPoolSize) { _ptrImpl = new MemoryPoolImpl(initialPoolSize); } @@ -58,9 +58,9 @@ MemoryPool::~MemoryPool() delete _ptrImpl; } -template WebRtc_Word32 +template int32_t MemoryPool::CreateMemoryPool(MemoryPool*& memoryPool, - WebRtc_UWord32 initialPoolSize) + uint32_t initialPoolSize) { memoryPool = new MemoryPool(initialPoolSize); if(memoryPool == NULL) @@ -83,7 +83,7 @@ MemoryPool::CreateMemoryPool(MemoryPool*& memoryPool, } template -WebRtc_Word32 MemoryPool::DeleteMemoryPool(MemoryPool*& memoryPool) +int32_t MemoryPool::DeleteMemoryPool(MemoryPool*& memoryPool) { if(memoryPool == NULL) { @@ -103,13 +103,13 @@ WebRtc_Word32 MemoryPool::DeleteMemoryPool(MemoryPool*& memoryPool) } template -WebRtc_Word32 MemoryPool::PopMemory(MemoryType*& memory) +int32_t MemoryPool::PopMemory(MemoryType*& memory) { return _ptrImpl->PopMemory(memory); } template -WebRtc_Word32 MemoryPool::PushMemory(MemoryType*& memory) +int32_t MemoryPool::PushMemory(MemoryType*& memory) { if(memory == NULL) { diff --git a/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h b/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h index 45f800b29..20093660c 100644 --- a/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h +++ b/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h @@ -23,18 +23,18 @@ class MemoryPoolImpl { public: // MemoryPool functions. - WebRtc_Word32 PopMemory(MemoryType*& memory); - WebRtc_Word32 PushMemory(MemoryType*& memory); + int32_t PopMemory(MemoryType*& memory); + int32_t PushMemory(MemoryType*& memory); - MemoryPoolImpl(WebRtc_Word32 initialPoolSize); + MemoryPoolImpl(int32_t initialPoolSize); ~MemoryPoolImpl(); // Atomic functions - WebRtc_Word32 Terminate(); + int32_t Terminate(); bool Initialize(); private: // Non-atomic function. - WebRtc_Word32 CreateMemory(WebRtc_UWord32 amountToCreate); + int32_t CreateMemory(uint32_t amountToCreate); CriticalSectionWrapper* _crit; @@ -42,13 +42,13 @@ private: ListWrapper _memoryPool; - WebRtc_UWord32 _initialPoolSize; - WebRtc_UWord32 _createdMemory; - WebRtc_UWord32 _outstandingMemory; + uint32_t _initialPoolSize; + uint32_t _createdMemory; + uint32_t _outstandingMemory; }; template -MemoryPoolImpl::MemoryPoolImpl(WebRtc_Word32 initialPoolSize) +MemoryPoolImpl::MemoryPoolImpl(int32_t initialPoolSize) : _crit(CriticalSectionWrapper::CreateCriticalSection()), _terminate(false), _memoryPool(), @@ -68,7 +68,7 @@ MemoryPoolImpl::~MemoryPoolImpl() } template -WebRtc_Word32 MemoryPoolImpl::PopMemory(MemoryType*& memory) +int32_t MemoryPoolImpl::PopMemory(MemoryType*& memory) { CriticalSectionScoped cs(_crit); if(_terminate) @@ -95,7 +95,7 @@ WebRtc_Word32 MemoryPoolImpl::PopMemory(MemoryType*& memory) } template -WebRtc_Word32 MemoryPoolImpl::PushMemory(MemoryType*& memory) +int32_t MemoryPoolImpl::PushMemory(MemoryType*& memory) { if(memory == NULL) { @@ -124,7 +124,7 @@ bool MemoryPoolImpl::Initialize() } template -WebRtc_Word32 MemoryPoolImpl::Terminate() +int32_t MemoryPoolImpl::Terminate() { CriticalSectionScoped cs(_crit); assert(_createdMemory == _outstandingMemory + _memoryPool.GetSize()); @@ -148,10 +148,10 @@ WebRtc_Word32 MemoryPoolImpl::Terminate() } template -WebRtc_Word32 MemoryPoolImpl::CreateMemory( - WebRtc_UWord32 amountToCreate) +int32_t MemoryPoolImpl::CreateMemory( + uint32_t amountToCreate) { - for(WebRtc_UWord32 i = 0; i < amountToCreate; i++) + for(uint32_t i = 0; i < amountToCreate; i++) { MemoryType* memory = new MemoryType(); if(memory == NULL) diff --git a/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h b/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h index 8ff97f87b..df440d071 100644 --- a/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h +++ b/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h @@ -47,14 +47,14 @@ class MemoryPoolImpl { public: // MemoryPool functions. - WebRtc_Word32 PopMemory(MemoryType*& memory); - WebRtc_Word32 PushMemory(MemoryType*& memory); + int32_t PopMemory(MemoryType*& memory); + int32_t PushMemory(MemoryType*& memory); - MemoryPoolImpl(WebRtc_Word32 /*initialPoolSize*/); + MemoryPoolImpl(int32_t /*initialPoolSize*/); ~MemoryPoolImpl(); // Atomic functions. - WebRtc_Word32 Terminate(); + int32_t Terminate(); bool Initialize(); private: // Non-atomic function. @@ -72,7 +72,7 @@ private: template MemoryPoolImpl::MemoryPoolImpl( - WebRtc_Word32 /*initialPoolSize*/) + int32_t /*initialPoolSize*/) : _pListHead(NULL), _createdMemory(0), _outstandingMemory(0) @@ -94,7 +94,7 @@ MemoryPoolImpl::~MemoryPoolImpl() } template -WebRtc_Word32 MemoryPoolImpl::PopMemory(MemoryType*& memory) +int32_t MemoryPoolImpl::PopMemory(MemoryType*& memory) { PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead); if(pListEntry == NULL) @@ -112,7 +112,7 @@ WebRtc_Word32 MemoryPoolImpl::PopMemory(MemoryType*& memory) } template -WebRtc_Word32 MemoryPoolImpl::PushMemory(MemoryType*& memory) +int32_t MemoryPoolImpl::PushMemory(MemoryType*& memory) { if(memory == NULL) { @@ -122,9 +122,9 @@ WebRtc_Word32 MemoryPoolImpl::PushMemory(MemoryType*& memory) MemoryPoolItem* item = ((MemoryPoolItemPayload*)memory)->base; - const WebRtc_Word32 usedItems = --_outstandingMemory; - const WebRtc_Word32 totalItems = _createdMemory.Value(); - const WebRtc_Word32 freeItems = totalItems - usedItems; + const int32_t usedItems = --_outstandingMemory; + const int32_t totalItems = _createdMemory.Value(); + const int32_t freeItems = totalItems - usedItems; if(freeItems < 0) { assert(false); @@ -157,9 +157,9 @@ bool MemoryPoolImpl::Initialize() } template -WebRtc_Word32 MemoryPoolImpl::Terminate() +int32_t MemoryPoolImpl::Terminate() { - WebRtc_Word32 itemsFreed = 0; + int32_t itemsFreed = 0; PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead); while(pListEntry != NULL) { diff --git a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc index 183005e37..7c4216ed8 100644 --- a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc +++ b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc @@ -12,7 +12,7 @@ #include "time_scheduler.h" namespace webrtc { -TimeScheduler::TimeScheduler(const WebRtc_UWord32 periodicityInMs) +TimeScheduler::TimeScheduler(const uint32_t periodicityInMs) : _crit(CriticalSectionWrapper::CreateCriticalSection()), _isStarted(false), _lastPeriodMark(), @@ -27,7 +27,7 @@ TimeScheduler::~TimeScheduler() delete _crit; } -WebRtc_Word32 TimeScheduler::UpdateScheduler() +int32_t TimeScheduler::UpdateScheduler() { CriticalSectionScoped cs(_crit); if(!_isStarted) @@ -47,11 +47,11 @@ WebRtc_Word32 TimeScheduler::UpdateScheduler() // Calculate the time that has past since previous call to this function. TickTime tickNow = TickTime::Now(); TickInterval amassedTicks = tickNow - _lastPeriodMark; - WebRtc_Word64 amassedMs = amassedTicks.Milliseconds(); + int64_t amassedMs = amassedTicks.Milliseconds(); // Calculate the number of periods the time that has passed correspond to. - WebRtc_Word32 periodsToClaim = (WebRtc_Word32)amassedMs / - ((WebRtc_Word32)_periodicityInMs); + int32_t periodsToClaim = static_cast(amassedMs / + static_cast(_periodicityInMs)); // One period will be worked off by this call. Make sure that the number of // pending periods don't end up being negative (e.g. if this function is @@ -65,7 +65,7 @@ WebRtc_Word32 TimeScheduler::UpdateScheduler() // Note that if this fuunction is called to often _lastPeriodMark can // refer to a time in the future which in turn will yield TimeToNextUpdate // that is greater than the periodicity - for(WebRtc_Word32 i = 0; i < periodsToClaim; i++) + for(int32_t i = 0; i < periodsToClaim; i++) { _lastPeriodMark += _periodicityInTicks; } @@ -76,8 +76,8 @@ WebRtc_Word32 TimeScheduler::UpdateScheduler() return 0; } -WebRtc_Word32 TimeScheduler::TimeToNextUpdate( - WebRtc_Word32& updateTimeInMS) const +int32_t TimeScheduler::TimeToNextUpdate( + int32_t& updateTimeInMS) const { CriticalSectionScoped cs(_crit); // Missed periods means that the next UpdateScheduler() should happen @@ -92,8 +92,8 @@ WebRtc_Word32 TimeScheduler::TimeToNextUpdate( // UpdateScheduler() TickTime tickNow = TickTime::Now(); TickInterval ticksSinceLastUpdate = tickNow - _lastPeriodMark; - const WebRtc_Word32 millisecondsSinceLastUpdate = - (WebRtc_Word32) ticksSinceLastUpdate.Milliseconds(); + const int32_t millisecondsSinceLastUpdate = + static_cast(ticksSinceLastUpdate.Milliseconds()); updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate; updateTimeInMS = (updateTimeInMS < 0) ? 0 : updateTimeInMS; diff --git a/webrtc/modules/audio_conference_mixer/source/time_scheduler.h b/webrtc/modules/audio_conference_mixer/source/time_scheduler.h index e2674d90c..31811d36b 100644 --- a/webrtc/modules/audio_conference_mixer/source/time_scheduler.h +++ b/webrtc/modules/audio_conference_mixer/source/time_scheduler.h @@ -22,15 +22,15 @@ class CriticalSectionWrapper; class TimeScheduler { public: - TimeScheduler(const WebRtc_UWord32 periodicityInMs); + TimeScheduler(const uint32_t periodicityInMs); ~TimeScheduler(); // Signal that a periodic event has been triggered. - WebRtc_Word32 UpdateScheduler(); + int32_t UpdateScheduler(); // Set updateTimeInMs to the amount of time until UpdateScheduler() should // be called. This time will never be negative. - WebRtc_Word32 TimeToNextUpdate(WebRtc_Word32& updateTimeInMS) const; + int32_t TimeToNextUpdate(int32_t& updateTimeInMS) const; private: CriticalSectionWrapper* _crit; @@ -38,9 +38,9 @@ private: bool _isStarted; TickTime _lastPeriodMark; - WebRtc_UWord32 _periodicityInMs; - WebRtc_Word64 _periodicityInTicks; - WebRtc_UWord32 _missedPeriods; + uint32_t _periodicityInMs; + int64_t _periodicityInTicks; + uint32_t _missedPeriods; }; } // namespace webrtc diff --git a/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc b/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc index f79898c0c..92cb33a70 100644 --- a/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc +++ b/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc @@ -46,8 +46,8 @@ int main(int /*argc*/, char* /*argv[]*/) } char versionString[256] = ""; - WebRtc_UWord32 remainingBufferInBytes = 256; - WebRtc_UWord32 position = 0; + uint32_t remainingBufferInBytes = 256; + uint32_t position = 0; AudioConferenceMixer::GetVersion(versionString,remainingBufferInBytes,position); int read = 1; @@ -70,7 +70,7 @@ int main(int /*argc*/, char* /*argv[]*/) getchar(); MixerParticipant::ParticipantType participantType; int option = 0; - WebRtc_UWord32 id = 0; + uint32_t id = 0; ListItem* item = NULL; ListWrapper participants; if(read == 0) @@ -122,7 +122,7 @@ int main(int /*argc*/, char* /*argv[]*/) std::cout << "The following participants have been created: " << std::endl; while(item) { - WebRtc_UWord32 id = item->GetUnsignedItem(); + uint32_t id = item->GetUnsignedItem(); std::cout << id; item = participants.Next(item); if(item != NULL) @@ -147,7 +147,7 @@ int main(int /*argc*/, char* /*argv[]*/) } else if(read == 8) { - const WebRtc_Word32 amountOfParticipants = 4; + const int32_t amountOfParticipants = 4; MixerParticipant::ParticipantType instance1Participants[] = {MixerParticipant::VIP, MixerParticipant::REGULAR, @@ -158,9 +158,9 @@ int main(int /*argc*/, char* /*argv[]*/) MixerParticipant::REGULAR, MixerParticipant::REGULAR, MixerParticipant::REGULAR}; - for(WebRtc_Word32 i = 0; i < amountOfParticipants; i++) + for(int32_t i = 0; i < amountOfParticipants; i++) { - WebRtc_Word32 startPosition = 0; + int32_t startPosition = 0; GenerateRandomPosition(startPosition); testInstance1->CreateParticipant(instance1Participants[i],startPosition); testInstance2->CreateParticipant(instance2Participants[i],startPosition); @@ -210,7 +210,9 @@ bool FileWriter::WriteToFile( const AudioFrame& audioFrame) { - WebRtc_Word32 written = (WebRtc_Word32)fwrite(audioFrame.data_,sizeof(WebRtc_Word16),audioFrame.samples_per_channel_,_file); + int32_t written = + static_cast(fwrite(audioFrame.data_, sizeof(int16_t), + audioFrame.samples_per_channel_, _file)); // Do not flush buffers since that will add (a lot of) delay return written == audioFrame.samples_per_channel_; } @@ -269,7 +271,7 @@ FileReader::ReadFromFile( AudioFrame& audioFrame) { - WebRtc_Word16 buffer[AudioFrame::kMaxDataSizeSamples]; + int16_t buffer[AudioFrame::kMaxDataSizeSamples]; LoopedFileRead(buffer,AudioFrame::kMaxDataSizeSamples,_sampleSize,_file); bool vad = false; @@ -278,7 +280,7 @@ FileReader::ReadFromFile( AudioFrame::kVadPassive; _volumeCalculator.ComputeLevel(buffer,_sampleSize); - const WebRtc_Word32 level = _volumeCalculator.GetLevel(); + const int32_t level = _volumeCalculator.GetLevel(); return audioFrame.UpdateFrame( -1, _timeStamp, buffer, @@ -293,9 +295,9 @@ FileReader::ReadFromFile( bool FileReader::FastForwardFile( - const WebRtc_Word32 samples) + const int32_t samples) { - WebRtc_Word16* tempBuffer = new WebRtc_Word16[samples]; + int16_t* tempBuffer = new int16_t[samples]; bool success = LoopedFileRead(tempBuffer,samples,samples,_file); delete[] tempBuffer; return success; @@ -333,13 +335,13 @@ FileReader::SetVAD( bool FileReader::GetVAD( - WebRtc_Word16* buffer, - WebRtc_UWord8 bufferLengthInSamples, + int16_t* buffer, + uint8_t bufferLengthInSamples, bool& vad) { if(_automaticVad) { - WebRtc_Word16 result = WebRtcVad_Process(_vadInstr,_frequency,buffer,bufferLengthInSamples); + int16_t result = WebRtcVad_Process(_vadInstr,_frequency,buffer,bufferLengthInSamples); if(result == -1) { assert(false); @@ -353,9 +355,9 @@ FileReader::GetVAD( MixerParticipant* MixerParticipant::CreateParticipant( - const WebRtc_UWord32 id, + const uint32_t id, ParticipantType participantType, - const WebRtc_Word32 startPosition, + const int32_t startPosition, char* outputPath) { if(participantType == RANDOM) @@ -376,7 +378,7 @@ MixerParticipant::CreateParticipant( } MixerParticipant::MixerParticipant( - const WebRtc_UWord32 id, + const uint32_t id, ParticipantType participantType) : _id(id), @@ -390,9 +392,9 @@ MixerParticipant::~MixerParticipant() { } -WebRtc_Word32 +int32_t MixerParticipant::GetAudioFrame( - const WebRtc_Word32 /*id*/, + const int32_t /*id*/, AudioFrame& audioFrame) { if(!_fileReader.ReadFromFile(audioFrame)) @@ -403,14 +405,14 @@ MixerParticipant::GetAudioFrame( return 0; } -WebRtc_Word32 +int32_t MixerParticipant::MixedAudioFrame( const AudioFrame& audioFrame) { return _fileWriter.WriteToFile(audioFrame); } -WebRtc_Word32 +int32_t MixerParticipant::GetParticipantType( ParticipantType& participantType) { @@ -420,7 +422,7 @@ MixerParticipant::GetParticipantType( bool MixerParticipant::InitializeFileReader( - const WebRtc_Word32 startPositionInSamples) + const int32_t startPositionInSamples) { char fileName[128] = ""; if(_participantType == REGULAR) @@ -446,7 +448,7 @@ bool MixerParticipant::InitializeFileWriter( char* outputPath) { - const WebRtc_Word32 stringsize = 128; + const int32_t stringsize = 128; char fileName[stringsize] = ""; strncpy(fileName,outputPath,stringsize); fileName[stringsize-1] = '\0'; @@ -461,7 +463,7 @@ MixerParticipant::InitializeFileWriter( } StatusReceiver::StatusReceiver( - const WebRtc_Word32 id) + const int32_t id) : _id(id), _mixedParticipants(NULL), @@ -482,9 +484,9 @@ StatusReceiver::~StatusReceiver() void StatusReceiver::MixedParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size) + const uint32_t size) { if(id != _id) { @@ -502,9 +504,9 @@ StatusReceiver::MixedParticipants( void StatusReceiver::VADPositiveParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size) + const uint32_t size) { if(id != _id) { @@ -523,8 +525,8 @@ StatusReceiver::VADPositiveParticipants( void StatusReceiver::MixedAudioLevel( - const WebRtc_Word32 id, - const WebRtc_UWord32 level) + const int32_t id, + const uint32_t level) { if(id != _id) { @@ -541,7 +543,7 @@ StatusReceiver::PrintMixedParticipants() { std::cout << "N/A" << std::endl; } - for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++) + for(uint16_t i = 0; i < _mixedParticipantsAmount; i++) { std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl; } @@ -555,7 +557,7 @@ StatusReceiver::PrintVadPositiveParticipants() { std::cout << "N/A" << std::endl; } - for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++) + for(uint16_t i = 0; i < _mixedParticipantsAmount; i++) { std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl; } @@ -567,7 +569,7 @@ StatusReceiver::PrintMixedAudioLevel() std::cout << "Mixed audio level = " << _mixedAudioLevel << std::endl; } -WebRtc_Word32 MixerWrapper::_mixerWrapperIdCounter = 0; +int32_t MixerWrapper::_mixerWrapperIdCounter = 0; MixerWrapper::MixerWrapper() : @@ -648,7 +650,7 @@ bool MixerWrapper::CreateParticipant( MixerParticipant::ParticipantType participantType) { - WebRtc_Word32 startPosition = 0; + int32_t startPosition = 0; GenerateRandomPosition(startPosition); return CreateParticipant(participantType,startPosition); } @@ -656,9 +658,9 @@ MixerWrapper::CreateParticipant( bool MixerWrapper::CreateParticipant( MixerParticipant::ParticipantType participantType, - const WebRtc_Word32 startPosition) + const int32_t startPosition) { - WebRtc_UWord32 id; + uint32_t id; if(!GetFreeItemIds(id)) { return false; @@ -684,7 +686,7 @@ MixerWrapper::CreateParticipant( bool MixerWrapper::DeleteParticipant( - const WebRtc_UWord32 id) + const uint32_t id) { bool success = StopMixingParticipant(id); if(!success) @@ -706,7 +708,7 @@ MixerWrapper::DeleteParticipant( bool MixerWrapper::StartMixing( - const WebRtc_UWord32 mixedParticipants) + const uint32_t mixedParticipants) { if(_processThread) { @@ -716,7 +718,7 @@ MixerWrapper::StartMixing( { assert(false); } - WebRtc_UWord32 mixedParticipantsTest = 0; + uint32_t mixedParticipantsTest = 0; _mixer->AmountOfMixedParticipants(mixedParticipantsTest); assert(mixedParticipantsTest == mixedParticipants); @@ -752,10 +754,10 @@ MixerWrapper::StopMixing() void MixerWrapper::NewMixedAudio( - const WebRtc_Word32 id, + const int32_t id, const AudioFrame& generalAudioFrame, const AudioFrame** uniqueAudioFrames, - const WebRtc_UWord32 size) + const uint32_t size) { if(id < 0) { @@ -766,9 +768,9 @@ MixerWrapper::NewMixedAudio( // Send the unique audio frames to its corresponding participants ListWrapper uniqueAudioFrameList; - for(WebRtc_UWord32 i = 0; i < size; i++) + for(uint32_t i = 0; i < size; i++) { - WebRtc_UWord32 id = (uniqueAudioFrames[i])->_id; + uint32_t id = (uniqueAudioFrames[i])->_id; MapItem* resultItem = _mixerParticipants.Find(id); if(resultItem == NULL) { @@ -833,7 +835,7 @@ MixerWrapper::PrintStatus() bool MixerWrapper::InitializeFileWriter() { - const WebRtc_Word32 stringsize = 128; + const int32_t stringsize = 128; char fileName[stringsize] = ""; strncpy(fileName,_instanceOutputPath,stringsize); fileName[stringsize-1] = '\0'; @@ -869,7 +871,7 @@ MixerWrapper::Process() assert(false); return false; } - WebRtc_Word32 processOfset = 0; + int32_t processOfset = 0; const TickTime currentTime = TickTime::Now(); if(_firstProcessCall) { @@ -880,12 +882,12 @@ MixerWrapper::Process() { TickInterval deltaTime = (currentTime - _previousTime); _previousTime += _periodicityInTicks; - processOfset = (WebRtc_Word32) deltaTime.Milliseconds(); + processOfset = (int32_t) deltaTime.Milliseconds(); processOfset -= FileReader::kProcessPeriodicityInMs; } _mixer->Process(); - WebRtc_Word32 timeUntilNextProcess = _mixer->TimeUntilNextProcess(); + int32_t timeUntilNextProcess = _mixer->TimeUntilNextProcess(); if(processOfset > FileReader::kProcessPeriodicityInMs) { std::cout << "Performance Warning: Process running " << processOfset << " too slow" << std::endl; @@ -910,7 +912,7 @@ MixerWrapper::Process() bool MixerWrapper::StartMixingParticipant( - const WebRtc_UWord32 id) + const uint32_t id) { MapItem* item = _mixerParticipants.Find(id); if(item == NULL) @@ -944,7 +946,7 @@ MixerWrapper::StartMixingParticipant( assert(anonymouslyMixed); return success; } - WebRtc_UWord32 previousAmountOfMixableParticipants = 0; + uint32_t previousAmountOfMixableParticipants = 0; bool success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0; assert(success); @@ -963,7 +965,7 @@ MixerWrapper::StartMixingParticipant( return false; } - WebRtc_UWord32 currentAmountOfMixableParticipants = 0; + uint32_t currentAmountOfMixableParticipants = 0; success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0; assert(currentAmountOfMixableParticipants == previousAmountOfMixableParticipants + 1); @@ -1001,7 +1003,7 @@ MixerWrapper::StartMixingParticipant( bool MixerWrapper::StopMixingParticipant( - const WebRtc_UWord32 id) + const uint32_t id) { MapItem* item = _mixerParticipants.Find(id); if(item == NULL) @@ -1010,12 +1012,12 @@ MixerWrapper::StopMixingParticipant( } MixerParticipant* participant = static_cast(item->GetItem()); bool success = false; - WebRtc_UWord32 previousAmountOfMixableParticipants = 0; + uint32_t previousAmountOfMixableParticipants = 0; success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0; assert(success); success = _mixer->SetMixabilityStatus(*participant,false) == 0; assert(success); - WebRtc_UWord32 currentAmountOfMixableParticipants = 0; + uint32_t currentAmountOfMixableParticipants = 0; success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0; assert(success); assert(success ? currentAmountOfMixableParticipants == previousAmountOfMixableParticipants -1 : @@ -1025,17 +1027,17 @@ MixerWrapper::StopMixingParticipant( bool MixerWrapper::GetFreeItemIds( - WebRtc_UWord32& itemId) + uint32_t& itemId) { if(!_freeItemIds.Empty()) { ListItem* item = _freeItemIds.First(); - WebRtc_UWord32* id = static_cast(item->GetItem()); + uint32_t* id = static_cast(item->GetItem()); itemId = *id; delete id; return true; } - if(_itemIdCounter == (WebRtc_UWord32) -1) + if(_itemIdCounter == (uint32_t) -1) { return false; } @@ -1045,9 +1047,9 @@ MixerWrapper::GetFreeItemIds( void MixerWrapper::AddFreeItemIds( - const WebRtc_UWord32 itemId) + const uint32_t itemId) { - WebRtc_UWord32* id = new WebRtc_UWord32; + uint32_t* id = new uint32_t; *id = itemId; _freeItemIds.PushBack(static_cast(id)); } @@ -1058,7 +1060,7 @@ MixerWrapper::ClearAllItemIds() ListItem* item = _freeItemIds.First(); while(item != NULL) { - WebRtc_UWord32* id = static_cast(item->GetItem()); + uint32_t* id = static_cast(item->GetItem()); delete id; _freeItemIds.Erase(item); item = _freeItemIds.First(); @@ -1067,21 +1069,24 @@ MixerWrapper::ClearAllItemIds() bool LoopedFileRead( - WebRtc_Word16* buffer, - WebRtc_UWord32 bufferSizeInSamples, - WebRtc_UWord32 samplesToRead, + int16_t* buffer, + uint32_t bufferSizeInSamples, + uint32_t samplesToRead, FILE* file) { if(bufferSizeInSamples < samplesToRead) { return false; } - WebRtc_UWord32 gottenSamples = (WebRtc_UWord32)fread(buffer,sizeof(WebRtc_Word16),samplesToRead,file); + uint32_t gottenSamples = static_cast( + fread(buffer, sizeof(int16_t), samplesToRead, file)); if(gottenSamples != samplesToRead) { - WebRtc_UWord32 missingSamples = samplesToRead - gottenSamples; + uint32_t missingSamples = samplesToRead - gottenSamples; fseek(file,0,0); - gottenSamples += (WebRtc_UWord32)fread(&buffer[gottenSamples],sizeof(WebRtc_Word16),missingSamples,file); + gottenSamples += + static_cast(fread(&buffer[gottenSamples], sizeof(int16_t), + missingSamples, file)); } if(gottenSamples != samplesToRead) { @@ -1092,7 +1097,7 @@ LoopedFileRead( void GenerateRandomPosition( - WebRtc_Word32& startPosition) + int32_t& startPosition) { startPosition = (rand() % (60*16000/160)) * 160; } diff --git a/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h b/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h index f25e5f16f..505ad91b0 100644 --- a/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h +++ b/webrtc/modules/audio_conference_mixer/test/FunctionTest/functionTest.h @@ -62,7 +62,7 @@ public: AudioFrame& audioFrame); bool FastForwardFile( - const WebRtc_Word32 samples); + const int32_t samples); bool EnableAutomaticVAD( bool enable, @@ -72,14 +72,14 @@ public: bool vad); private: bool GetVAD( - WebRtc_Word16* buffer, - WebRtc_UWord8 bufferLengthInSamples, + int16_t* buffer, + uint8_t bufferLengthInSamples, bool& vad); Frequency _frequency; - WebRtc_UWord8 _sampleSize; + uint8_t _sampleSize; - WebRtc_UWord32 _timeStamp; + uint32_t _timeStamp; FILE* _file; @@ -102,33 +102,33 @@ public: }; static MixerParticipant* CreateParticipant( - const WebRtc_UWord32 id, + const uint32_t id, ParticipantType participantType, - const WebRtc_Word32 startPosition, + const int32_t startPosition, char* outputPath); ~MixerParticipant(); - WebRtc_Word32 GetAudioFrame( - const WebRtc_Word32 id, + int32_t GetAudioFrame( + const int32_t id, AudioFrame& audioFrame); - WebRtc_Word32 MixedAudioFrame( + int32_t MixedAudioFrame( const AudioFrame& audioFrame); - WebRtc_Word32 GetParticipantType( + int32_t GetParticipantType( ParticipantType& participantType); private: MixerParticipant( - const WebRtc_UWord32 id, + const uint32_t id, ParticipantType participantType); bool InitializeFileReader( - const WebRtc_Word32 startPositionInSamples); + const int32_t startPositionInSamples); bool InitializeFileWriter( char* outputPath); - WebRtc_UWord32 _id; + uint32_t _id; ParticipantType _participantType; FileReader _fileReader; @@ -139,22 +139,22 @@ class StatusReceiver : public AudioMixerStatusReceiver { public: StatusReceiver( - const WebRtc_Word32 id); + const int32_t id); ~StatusReceiver(); void MixedParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size); + const uint32_t size); void VADPositiveParticipants( - const WebRtc_Word32 id, + const int32_t id, const ParticipantStatistics* participantStatistics, - const WebRtc_UWord32 size); + const uint32_t size); void MixedAudioLevel( - const WebRtc_Word32 id, - const WebRtc_UWord32 level); + const int32_t id, + const uint32_t level); void PrintMixedParticipants(); @@ -162,17 +162,17 @@ public: void PrintMixedAudioLevel(); private: - WebRtc_Word32 _id; + int32_t _id; ParticipantStatistics* _mixedParticipants; - WebRtc_UWord32 _mixedParticipantsAmount; - WebRtc_UWord32 _mixedParticipantsSize; + uint32_t _mixedParticipantsAmount; + uint32_t _mixedParticipantsSize; ParticipantStatistics* _vadPositiveParticipants; - WebRtc_UWord32 _vadPositiveParticipantsAmount; - WebRtc_UWord32 _vadPositiveParticipantsSize; + uint32_t _vadPositiveParticipantsAmount; + uint32_t _vadPositiveParticipantsSize; - WebRtc_UWord32 _mixedAudioLevel; + uint32_t _mixedAudioLevel; }; class MixerWrapper : public AudioMixerOutputReceiver @@ -189,21 +189,21 @@ public: bool CreateParticipant( MixerParticipant::ParticipantType participantType, - const WebRtc_Word32 startPosition); + const int32_t startPosition); bool DeleteParticipant( - const WebRtc_UWord32 id); + const uint32_t id); bool StartMixing( - const WebRtc_UWord32 mixedParticipants = AudioConferenceMixer::kDefaultAmountOfMixedParticipants); + const uint32_t mixedParticipants = AudioConferenceMixer::kDefaultAmountOfMixedParticipants); bool StopMixing(); void NewMixedAudio( - const WebRtc_Word32 id, + const int32_t id, const AudioFrame& generalAudioFrame, const AudioFrame** uniqueAudioFrames, - const WebRtc_UWord32 size); + const uint32_t size); bool GetParticipantList( ListWrapper& participants); @@ -220,16 +220,16 @@ private: bool Process(); bool StartMixingParticipant( - const WebRtc_UWord32 id); + const uint32_t id); bool StopMixingParticipant( - const WebRtc_UWord32 id); + const uint32_t id); bool GetFreeItemIds( - WebRtc_UWord32& itemId); + uint32_t& itemId); void AddFreeItemIds( - const WebRtc_UWord32 itemId); + const uint32_t itemId); void ClearAllItemIds(); @@ -241,17 +241,17 @@ private: bool _firstProcessCall; TickTime _previousTime; // Tick time of previous process - const WebRtc_Word64 _periodicityInTicks; // Periodicity + const int64_t _periodicityInTicks; // Periodicity webrtc::EventWrapper* _synchronizationEvent; ListWrapper _freeItemIds; - WebRtc_UWord32 _itemIdCounter; + uint32_t _itemIdCounter; MapWrapper _mixerParticipants; - static WebRtc_Word32 _mixerWrapperIdCounter; - WebRtc_Word32 _mixerWrappererId; + static int32_t _mixerWrapperIdCounter; + int32_t _mixerWrappererId; char _instanceOutputPath[128]; webrtc::Trace* _trace; @@ -264,13 +264,13 @@ private: bool LoopedFileRead( - WebRtc_Word16* buffer, - WebRtc_UWord32 bufferSizeInSamples, - WebRtc_UWord32 samplesToRead, + int16_t* buffer, + uint32_t bufferSizeInSamples, + uint32_t samplesToRead, FILE* file); void GenerateRandomPosition( - WebRtc_Word32& startPosition); + int32_t& startPosition); #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_