diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h index 5b7af49c0..43ab5007d 100644 --- a/webrtc/modules/utility/interface/file_player.h +++ b/webrtc/modules/utility/interface/file_player.h @@ -29,7 +29,7 @@ public: // Note: will return NULL for video file formats (e.g. AVI) if the flag // WEBRTC_MODULE_UTILITY_VIDEO is not defined. - static FilePlayer* CreateFilePlayer(const WebRtc_UWord32 instanceID, + static FilePlayer* CreateFilePlayer(const uint32_t instanceID, const FileFormats fileFormat); static void DestroyFilePlayer(FilePlayer* player); @@ -43,65 +43,65 @@ public: int frequencyInHz) = 0; // Register callback for receiving file playing notifications. - virtual WebRtc_Word32 RegisterModuleFileCallback( + virtual int32_t RegisterModuleFileCallback( FileCallback* callback) = 0; // API for playing audio from fileName to channel. // Note: codecInst is used for pre-encoded files. - virtual WebRtc_Word32 StartPlayingFile( + virtual int32_t StartPlayingFile( const char* fileName, bool loop, - WebRtc_UWord32 startPosition, + uint32_t startPosition, float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition = 0, + uint32_t notification, + uint32_t stopPosition = 0, const CodecInst* codecInst = NULL) = 0; // Note: codecInst is used for pre-encoded files. - virtual WebRtc_Word32 StartPlayingFile( + virtual int32_t StartPlayingFile( InStream& sourceStream, - WebRtc_UWord32 startPosition, + uint32_t startPosition, float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition = 0, + uint32_t notification, + uint32_t stopPosition = 0, const CodecInst* codecInst = NULL) = 0; - virtual WebRtc_Word32 StopPlayingFile() = 0; + virtual int32_t StopPlayingFile() = 0; virtual bool IsPlayingFile() const = 0; - virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs) = 0; + virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0; // Set audioCodec to the currently used audio codec. - virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const = 0; + virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0; - virtual WebRtc_Word32 Frequency() const = 0; + virtual int32_t Frequency() const = 0; // Note: scaleFactor is in the range [0.0 - 2.0] - virtual WebRtc_Word32 SetAudioScaling(float scaleFactor) = 0; + virtual int32_t SetAudioScaling(float scaleFactor) = 0; // Return the time in ms until next video frame should be pulled (by // calling GetVideoFromFile(..)). // Note: this API reads one video frame from file. This means that it should // be called exactly once per GetVideoFromFile(..) API call. - virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;} + virtual int32_t TimeUntilNextVideoFrame() { return -1;} - virtual WebRtc_Word32 StartPlayingVideoFile( + virtual int32_t StartPlayingVideoFile( const char* /*fileName*/, bool /*loop*/, bool /*videoOnly*/) { return -1;} - virtual WebRtc_Word32 video_codec_info(VideoCodec& /*videoCodec*/) const + virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const {return -1;} - virtual WebRtc_Word32 GetVideoFromFile(I420VideoFrame& /*videoFrame*/) + virtual int32_t GetVideoFromFile(I420VideoFrame& /*videoFrame*/) { return -1;} // Same as GetVideoFromFile(). videoFrame will have the resolution specified // by the width outWidth and height outHeight in pixels. - virtual WebRtc_Word32 GetVideoFromFile(I420VideoFrame& /*videoFrame*/, - const WebRtc_UWord32 /*outWidth*/, - const WebRtc_UWord32 /*outHeight*/) + virtual int32_t GetVideoFromFile(I420VideoFrame& /*videoFrame*/, + const uint32_t /*outWidth*/, + const uint32_t /*outHeight*/) {return -1;} protected: virtual ~FilePlayer() {} diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/interface/file_recorder.h index 01d4600d5..2c3268d66 100644 --- a/webrtc/modules/utility/interface/file_recorder.h +++ b/webrtc/modules/utility/interface/file_recorder.h @@ -28,40 +28,40 @@ public: // Note: will return NULL for video file formats (e.g. AVI) if the flag // WEBRTC_MODULE_UTILITY_VIDEO is not defined. - static FileRecorder* CreateFileRecorder(const WebRtc_UWord32 instanceID, + static FileRecorder* CreateFileRecorder(const uint32_t instanceID, const FileFormats fileFormat); static void DestroyFileRecorder(FileRecorder* recorder); - virtual WebRtc_Word32 RegisterModuleFileCallback( + virtual int32_t RegisterModuleFileCallback( FileCallback* callback) = 0; virtual FileFormats RecordingFileFormat() const = 0; - virtual WebRtc_Word32 StartRecordingAudioFile( + virtual int32_t StartRecordingAudioFile( const char* fileName, const CodecInst& codecInst, - WebRtc_UWord32 notification, + uint32_t notification, ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0; - virtual WebRtc_Word32 StartRecordingAudioFile( + virtual int32_t StartRecordingAudioFile( OutStream& destStream, const CodecInst& codecInst, - WebRtc_UWord32 notification, + uint32_t notification, ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0; // Stop recording. // Note: this API is for both audio and video. - virtual WebRtc_Word32 StopRecording() = 0; + virtual int32_t StopRecording() = 0; // Return true if recording. // Note: this API is for both audio and video. virtual bool IsRecording() const = 0; - virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0; + virtual int32_t codec_info(CodecInst& codecInst) const = 0; // Write frame to file. Frame should contain 10ms of un-ecoded audio data. - virtual WebRtc_Word32 RecordAudioToFile( + virtual int32_t RecordAudioToFile( const AudioFrame& frame, const TickTime* playoutTS = NULL) = 0; @@ -71,7 +71,7 @@ public: // Only video data will be recorded if videoOnly is true. amrFormat // specifies the amr/amrwb storage format. // Note: the file format is AVI. - virtual WebRtc_Word32 StartRecordingVideoFile( + virtual int32_t StartRecordingVideoFile( const char* fileName, const CodecInst& audioCodecInst, const VideoCodec& videoCodecInst, @@ -79,7 +79,7 @@ public: bool videoOnly = false) = 0; // Record the video frame in videoFrame to AVI file. - virtual WebRtc_Word32 RecordVideoToFile( + virtual int32_t RecordVideoToFile( const I420VideoFrame& videoFrame) = 0; protected: diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/interface/process_thread.h index 6c5140497..3e25bd0dc 100644 --- a/webrtc/modules/utility/interface/process_thread.h +++ b/webrtc/modules/utility/interface/process_thread.h @@ -22,11 +22,11 @@ public: static ProcessThread* CreateProcessThread(); static void DestroyProcessThread(ProcessThread* module); - virtual WebRtc_Word32 Start() = 0; - virtual WebRtc_Word32 Stop() = 0; + virtual int32_t Start() = 0; + virtual int32_t Stop() = 0; - virtual WebRtc_Word32 RegisterModule(const Module* module) = 0; - virtual WebRtc_Word32 DeRegisterModule(const Module* module) = 0; + virtual int32_t RegisterModule(const Module* module) = 0; + virtual int32_t DeRegisterModule(const Module* module) = 0; protected: virtual ~ProcessThread(); }; diff --git a/webrtc/modules/utility/interface/rtp_dump.h b/webrtc/modules/utility/interface/rtp_dump.h index 9291a1c36..9a852d07c 100644 --- a/webrtc/modules/utility/interface/rtp_dump.h +++ b/webrtc/modules/utility/interface/rtp_dump.h @@ -31,10 +31,10 @@ public: // Open the file fileNameUTF8 for writing RTP/RTCP packets. // Note: this API also adds the rtpplay header. - virtual WebRtc_Word32 Start(const char* fileNameUTF8) = 0; + virtual int32_t Start(const char* fileNameUTF8) = 0; // Close the existing file. No more packets will be recorded. - virtual WebRtc_Word32 Stop() = 0; + virtual int32_t Stop() = 0; // Return true if a file is open for recording RTP/RTCP packets. virtual bool IsActive() const = 0; @@ -42,8 +42,8 @@ public: // Writes the RTP/RTCP packet in packet with length packetLength in bytes. // Note: packet should contain the RTP/RTCP part of the packet. I.e. the // first bytes of packet should be the RTP/RTCP header. - virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet, - WebRtc_UWord16 packetLength) = 0; + virtual int32_t DumpPacket(const uint8_t* packet, + uint16_t packetLength) = 0; protected: virtual ~RtpDump(); diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc index f023d2264..85fb6988b 100644 --- a/webrtc/modules/utility/source/coder.cc +++ b/webrtc/modules/utility/source/coder.cc @@ -20,7 +20,7 @@ #endif namespace webrtc { -AudioCoder::AudioCoder(WebRtc_UWord32 instanceID) +AudioCoder::AudioCoder(uint32_t instanceID) : _acm(AudioCodingModule::Create(instanceID)), _receiveCodec(), _encodeTimestamp(0), @@ -38,8 +38,8 @@ AudioCoder::~AudioCoder() AudioCodingModule::Destroy(_acm); } -WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst, - ACMAMRPackingFormat amrFormat) +int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst, + ACMAMRPackingFormat amrFormat) { if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1) { @@ -48,8 +48,8 @@ WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst, return 0; } -WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst, - ACMAMRPackingFormat amrFormat) +int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst, + ACMAMRPackingFormat amrFormat) { if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1) { @@ -59,16 +59,16 @@ WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst, return 0; } -WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio, - WebRtc_UWord32 sampFreqHz, - const WebRtc_Word8* incomingPayload, - WebRtc_Word32 payloadLength) +int32_t AudioCoder::Decode(AudioFrame& decodedAudio, + uint32_t sampFreqHz, + const int8_t* incomingPayload, + int32_t payloadLength) { if (payloadLength > 0) { - const WebRtc_UWord8 payloadType = _receiveCodec.pltype; + const uint8_t payloadType = _receiveCodec.pltype; _decodeTimestamp += _receiveCodec.pacsize; - if(_acm->IncomingPayload((const WebRtc_UWord8*) incomingPayload, + if(_acm->IncomingPayload((const uint8_t*) incomingPayload, payloadLength, payloadType, _decodeTimestamp) == -1) @@ -76,18 +76,18 @@ WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio, return -1; } } - return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz, &decodedAudio); + return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio); } -WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio, - WebRtc_UWord16& sampFreqHz) +int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio, + uint16_t& sampFreqHz) { return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio); } -WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio, - WebRtc_Word8* encodedData, - WebRtc_UWord32& encodedLengthInBytes) +int32_t AudioCoder::Encode(const AudioFrame& audio, + int8_t* encodedData, + uint32_t& encodedLengthInBytes) { // Fake a timestamp in case audio doesn't contain a correct timestamp. // Make a local copy of the audio frame since audio is const @@ -112,15 +112,15 @@ WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio, return 0; } -WebRtc_Word32 AudioCoder::SendData( +int32_t AudioCoder::SendData( FrameType /* frameType */, - WebRtc_UWord8 /* payloadType */, - WebRtc_UWord32 /* timeStamp */, - const WebRtc_UWord8* payloadData, - WebRtc_UWord16 payloadSize, + uint8_t /* payloadType */, + uint32_t /* timeStamp */, + const uint8_t* payloadData, + uint16_t payloadSize, const RTPFragmentationHeader* /* fragmentation*/) { - memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize); + memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); _encodedLengthInBytes = payloadSize; return 0; } diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h index e7cbfb8bb..9dd2566b8 100644 --- a/webrtc/modules/utility/source/coder.h +++ b/webrtc/modules/utility/source/coder.h @@ -21,46 +21,43 @@ class AudioFrame; class AudioCoder : public AudioPacketizationCallback { public: - AudioCoder(WebRtc_UWord32 instanceID); + AudioCoder(uint32_t instanceID); ~AudioCoder(); - WebRtc_Word32 SetEncodeCodec( + int32_t SetEncodeCodec( const CodecInst& codecInst, ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); - WebRtc_Word32 SetDecodeCodec( + int32_t SetDecodeCodec( const CodecInst& codecInst, ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient); - WebRtc_Word32 Decode(AudioFrame& decodedAudio, WebRtc_UWord32 sampFreqHz, - const WebRtc_Word8* incomingPayload, - WebRtc_Word32 payloadLength); + int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz, + const int8_t* incomingPayload, int32_t payloadLength); - WebRtc_Word32 PlayoutData(AudioFrame& decodedAudio, - WebRtc_UWord16& sampFreqHz); + int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz); - WebRtc_Word32 Encode(const AudioFrame& audio, - WebRtc_Word8* encodedData, - WebRtc_UWord32& encodedLengthInBytes); + int32_t Encode(const AudioFrame& audio, int8_t* encodedData, + uint32_t& encodedLengthInBytes); protected: - virtual WebRtc_Word32 SendData(FrameType frameType, - WebRtc_UWord8 payloadType, - WebRtc_UWord32 timeStamp, - const WebRtc_UWord8* payloadData, - WebRtc_UWord16 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); private: AudioCodingModule* _acm; CodecInst _receiveCodec; - WebRtc_UWord32 _encodeTimestamp; - WebRtc_Word8* _encodedData; - WebRtc_UWord32 _encodedLengthInBytes; + uint32_t _encodeTimestamp; + int8_t* _encodedData; + uint32_t _encodedLengthInBytes; - WebRtc_UWord32 _decodeTimestamp; + uint32_t _decodeTimestamp; }; } // namespace webrtc diff --git a/webrtc/modules/utility/source/file_player_impl.cc b/webrtc/modules/utility/source/file_player_impl.cc index 2ca205a14..52ebe32b1 100644 --- a/webrtc/modules/utility/source/file_player_impl.cc +++ b/webrtc/modules/utility/source/file_player_impl.cc @@ -25,7 +25,7 @@ #endif namespace webrtc { -FilePlayer* FilePlayer::CreateFilePlayer(WebRtc_UWord32 instanceID, +FilePlayer* FilePlayer::CreateFilePlayer(uint32_t instanceID, FileFormats fileFormat) { switch(fileFormat) @@ -57,7 +57,7 @@ void FilePlayer::DestroyFilePlayer(FilePlayer* player) delete player; } -FilePlayerImpl::FilePlayerImpl(const WebRtc_UWord32 instanceID, +FilePlayerImpl::FilePlayerImpl(const uint32_t instanceID, const FileFormats fileFormat) : _instanceID(instanceID), _fileFormat(fileFormat), @@ -78,7 +78,7 @@ FilePlayerImpl::~FilePlayerImpl() MediaFile::DestroyMediaFile(&_fileModule); } -WebRtc_Word32 FilePlayerImpl::Frequency() const +int32_t FilePlayerImpl::Frequency() const { if(_codec.plfreq == 0) { @@ -108,13 +108,13 @@ WebRtc_Word32 FilePlayerImpl::Frequency() const } } -WebRtc_Word32 FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const +int32_t FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const { audioCodec = _codec; return 0; } -WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( +int32_t FilePlayerImpl::Get10msAudioFromFile( int16_t* outBuffer, int& lengthInSamples, int frequencyInHz) @@ -134,10 +134,10 @@ WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( unresampledAudioFrame.sample_rate_hz_ = _codec.plfreq; // L16 is un-encoded data. Just pull 10 ms. - WebRtc_UWord32 lengthInBytes = + uint32_t lengthInBytes = sizeof(unresampledAudioFrame.data_); if (_fileModule.PlayoutAudioData( - (WebRtc_Word8*)unresampledAudioFrame.data_, + (int8_t*)unresampledAudioFrame.data_, lengthInBytes) == -1) { // End of file reached. @@ -150,19 +150,19 @@ WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( } // One sample is two bytes. unresampledAudioFrame.samples_per_channel_ = - (WebRtc_UWord16)lengthInBytes >> 1; + (uint16_t)lengthInBytes >> 1; }else { // Decode will generate 10 ms of audio data. PlayoutAudioData(..) // expects a full frame. If the frame size is larger than 10 ms, // PlayoutAudioData(..) data should be called proportionally less often. - WebRtc_Word16 encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES]; - WebRtc_UWord32 encodedLengthInBytes = 0; + int16_t encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES]; + uint32_t encodedLengthInBytes = 0; if(++_numberOf10MsInDecoder >= _numberOf10MsPerFrame) { _numberOf10MsInDecoder = 0; - WebRtc_UWord32 bytesFromFile = sizeof(encodedBuffer); - if (_fileModule.PlayoutAudioData((WebRtc_Word8*)encodedBuffer, + uint32_t bytesFromFile = sizeof(encodedBuffer); + if (_fileModule.PlayoutAudioData((int8_t*)encodedBuffer, bytesFromFile) == -1) { // End of file reached. @@ -171,7 +171,7 @@ WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( encodedLengthInBytes = bytesFromFile; } if(_audioDecoder.Decode(unresampledAudioFrame,frequencyInHz, - (WebRtc_Word8*)encodedBuffer, + (int8_t*)encodedBuffer, encodedLengthInBytes) == -1) { return -1; @@ -187,7 +187,7 @@ WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( // New sampling frequency. Update state. outLen = frequencyInHz / 100; - memset(outBuffer, 0, outLen * sizeof(WebRtc_Word16)); + memset(outBuffer, 0, outLen * sizeof(int16_t)); return 0; } _resampler.Push(unresampledAudioFrame.data_, @@ -202,19 +202,19 @@ WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile( { for (int i = 0;i < outLen; i++) { - outBuffer[i] = (WebRtc_Word16)(outBuffer[i] * _scaling); + outBuffer[i] = (int16_t)(outBuffer[i] * _scaling); } } _decodedLengthInMS += 10; return 0; } -WebRtc_Word32 FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback) +int32_t FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback) { return _fileModule.SetModuleFileCallback(callback); } -WebRtc_Word32 FilePlayerImpl::SetAudioScaling(float scaleFactor) +int32_t FilePlayerImpl::SetAudioScaling(float scaleFactor) { if((scaleFactor >= 0)&&(scaleFactor <= 2.0)) { @@ -226,13 +226,13 @@ WebRtc_Word32 FilePlayerImpl::SetAudioScaling(float scaleFactor) return -1; } -WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const char* fileName, - bool loop, - WebRtc_UWord32 startPosition, - float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition, - const CodecInst* codecInst) +int32_t FilePlayerImpl::StartPlayingFile(const char* fileName, + bool loop, + uint32_t startPosition, + float volumeScaling, + uint32_t notification, + uint32_t stopPosition, + const CodecInst* codecInst) { if (_fileFormat == kFileFormatPcm16kHzFile || _fileFormat == kFileFormatPcm8kHzFile|| @@ -322,12 +322,12 @@ WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const char* fileName, return 0; } -WebRtc_Word32 FilePlayerImpl::StartPlayingFile(InStream& sourceStream, - WebRtc_UWord32 startPosition, - float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition, - const CodecInst* codecInst) +int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream, + uint32_t startPosition, + float volumeScaling, + uint32_t notification, + uint32_t stopPosition, + const CodecInst* codecInst) { if (_fileFormat == kFileFormatPcm16kHzFile || _fileFormat == kFileFormatPcm32kHzFile || @@ -415,7 +415,7 @@ WebRtc_Word32 FilePlayerImpl::StartPlayingFile(InStream& sourceStream, return 0; } -WebRtc_Word32 FilePlayerImpl::StopPlayingFile() +int32_t FilePlayerImpl::StopPlayingFile() { memset(&_codec, 0, sizeof(CodecInst)); _numberOf10MsPerFrame = 0; @@ -428,12 +428,12 @@ bool FilePlayerImpl::IsPlayingFile() const return _fileModule.IsPlaying(); } -WebRtc_Word32 FilePlayerImpl::GetPlayoutPosition(WebRtc_UWord32& durationMs) +int32_t FilePlayerImpl::GetPlayoutPosition(uint32_t& durationMs) { return _fileModule.PlayoutPositionMs(durationMs); } -WebRtc_Word32 FilePlayerImpl::SetUpAudioDecoder() +int32_t FilePlayerImpl::SetUpAudioDecoder() { if ((_fileModule.codec_info(_codec) == -1)) { @@ -462,7 +462,7 @@ WebRtc_Word32 FilePlayerImpl::SetUpAudioDecoder() } #ifdef WEBRTC_MODULE_UTILITY_VIDEO -VideoFilePlayerImpl::VideoFilePlayerImpl(WebRtc_UWord32 instanceID, +VideoFilePlayerImpl::VideoFilePlayerImpl(uint32_t instanceID, FileFormats fileFormat) : FilePlayerImpl(instanceID,fileFormat), _videoDecoder(*new VideoCoder(instanceID)), @@ -488,7 +488,7 @@ VideoFilePlayerImpl::~VideoFilePlayerImpl() delete &_encodedData; } -WebRtc_Word32 VideoFilePlayerImpl::StartPlayingVideoFile( +int32_t VideoFilePlayerImpl::StartPlayingVideoFile( const char* fileName, bool loop, bool videoOnly) @@ -525,7 +525,7 @@ WebRtc_Word32 VideoFilePlayerImpl::StartPlayingVideoFile( return 0; } -WebRtc_Word32 VideoFilePlayerImpl::StopPlayingFile() +int32_t VideoFilePlayerImpl::StopPlayingFile() { CriticalSectionScoped lock( _critSec); @@ -535,13 +535,13 @@ WebRtc_Word32 VideoFilePlayerImpl::StopPlayingFile() return FilePlayerImpl::StopPlayingFile(); } -WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame, - WebRtc_UWord32 outWidth, - WebRtc_UWord32 outHeight) +int32_t VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame, + uint32_t outWidth, + uint32_t outHeight) { CriticalSectionScoped lock( _critSec); - WebRtc_Word32 retVal = GetVideoFromFile(videoFrame); + int32_t retVal = GetVideoFromFile(videoFrame); if(retVal != 0) { return retVal; @@ -554,7 +554,7 @@ WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame, return retVal; } -WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame) +int32_t VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame) { CriticalSectionScoped lock( _critSec); // No new video data read from file. @@ -563,7 +563,7 @@ WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame) videoFrame.ResetSize(); return -1; } - WebRtc_Word32 retVal = 0; + int32_t retVal = 0; if(strncmp(video_codec_info_.plName, "I420", 5) == 0) { int size_y = video_codec_info_.width * video_codec_info_.height; @@ -588,7 +588,7 @@ WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame) retVal = _videoDecoder.Decode(videoFrame, _encodedData); } - WebRtc_Word64 renderTimeMs = TickTime::MillisecondTimestamp(); + int64_t renderTimeMs = TickTime::MillisecondTimestamp(); videoFrame.set_render_time_ms(renderTimeMs); // Indicate that the current frame in the encoded buffer is old/has @@ -601,7 +601,7 @@ WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(I420VideoFrame& videoFrame) return retVal; } -WebRtc_Word32 VideoFilePlayerImpl::video_codec_info( +int32_t VideoFilePlayerImpl::video_codec_info( VideoCodec& videoCodec) const { if(video_codec_info_.plName[0] == 0) @@ -612,7 +612,7 @@ WebRtc_Word32 VideoFilePlayerImpl::video_codec_info( return 0; } -WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame() +int32_t VideoFilePlayerImpl::TimeUntilNextVideoFrame() { if(_fileFormat != kFileFormatAviFile) { @@ -630,9 +630,9 @@ WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame() if(_fileFormat == kFileFormatAviFile) { // Get next video frame - WebRtc_UWord32 encodedBufferLengthInBytes = _encodedData.bufferSize; + uint32_t encodedBufferLengthInBytes = _encodedData.bufferSize; if(_fileModule.PlayoutAVIVideoData( - reinterpret_cast< WebRtc_Word8*>(_encodedData.payloadData), + reinterpret_cast< int8_t*>(_encodedData.payloadData), encodedBufferLengthInBytes) != 0) { WEBRTC_TRACE( @@ -659,7 +659,7 @@ WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame() // Frame rate is in frames per seconds. Frame length is // calculated as an integer division which means it may // be rounded down. Compensate for this every second. - WebRtc_UWord32 rest = 1000%_frameLengthMS; + uint32_t rest = 1000%_frameLengthMS; _accumulatedRenderTimeMs += rest; } _accumulatedRenderTimeMs += _frameLengthMS; @@ -667,7 +667,7 @@ WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame() } } - WebRtc_Word64 timeToNextFrame; + int64_t timeToNextFrame; if(_videoOnly) { timeToNextFrame = _accumulatedRenderTimeMs - @@ -686,10 +686,10 @@ WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame() // Wraparound or audio stream has gone to far ahead of the video stream. return -1; } - return static_cast(timeToNextFrame); + return static_cast(timeToNextFrame); } -WebRtc_Word32 VideoFilePlayerImpl::SetUpVideoDecoder() +int32_t VideoFilePlayerImpl::SetUpVideoDecoder() { if (_fileModule.VideoCodecInst(video_codec_info_) != 0) { @@ -702,7 +702,7 @@ WebRtc_Word32 VideoFilePlayerImpl::SetUpVideoDecoder() return -1; } - WebRtc_Word32 useNumberOfCores = 1; + int32_t useNumberOfCores = 1; if(_videoDecoder.SetDecodeCodec(video_codec_info_, useNumberOfCores) != 0) { WEBRTC_TRACE( @@ -718,7 +718,7 @@ WebRtc_Word32 VideoFilePlayerImpl::SetUpVideoDecoder() // Size of unencoded data (I420) should be the largest possible frame size // in a file. - const WebRtc_UWord32 KReadBufferSize = 3 * video_codec_info_.width * + const uint32_t KReadBufferSize = 3 * video_codec_info_.width * video_codec_info_.height / 2; _encodedData.VerifyAndAllocate(KReadBufferSize); _encodedData.encodedHeight = video_codec_info_.height; diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h index c188e23b1..ebf0da17e 100644 --- a/webrtc/modules/utility/source/file_player_impl.h +++ b/webrtc/modules/utility/source/file_player_impl.h @@ -29,51 +29,51 @@ class FrameScaler; class FilePlayerImpl : public FilePlayer { public: - FilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat); + FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat); ~FilePlayerImpl(); virtual int Get10msAudioFromFile( int16_t* outBuffer, int& lengthInSamples, int frequencyInHz); - virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback); - virtual WebRtc_Word32 StartPlayingFile( + virtual int32_t RegisterModuleFileCallback(FileCallback* callback); + virtual int32_t StartPlayingFile( const char* fileName, bool loop, - WebRtc_UWord32 startPosition, + uint32_t startPosition, float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition = 0, + uint32_t notification, + uint32_t stopPosition = 0, const CodecInst* codecInst = NULL); - virtual WebRtc_Word32 StartPlayingFile( + virtual int32_t StartPlayingFile( InStream& sourceStream, - WebRtc_UWord32 startPosition, + uint32_t startPosition, float volumeScaling, - WebRtc_UWord32 notification, - WebRtc_UWord32 stopPosition = 0, + uint32_t notification, + uint32_t stopPosition = 0, const CodecInst* codecInst = NULL); - virtual WebRtc_Word32 StopPlayingFile(); + virtual int32_t StopPlayingFile(); virtual bool IsPlayingFile() const; - virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs); - virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const; - virtual WebRtc_Word32 Frequency() const; - virtual WebRtc_Word32 SetAudioScaling(float scaleFactor); + virtual int32_t GetPlayoutPosition(uint32_t& durationMs); + virtual int32_t AudioCodec(CodecInst& audioCodec) const; + virtual int32_t Frequency() const; + virtual int32_t SetAudioScaling(float scaleFactor); protected: - WebRtc_Word32 SetUpAudioDecoder(); + int32_t SetUpAudioDecoder(); - WebRtc_UWord32 _instanceID; + uint32_t _instanceID; const FileFormats _fileFormat; MediaFile& _fileModule; - WebRtc_UWord32 _decodedLengthInMS; + uint32_t _decodedLengthInMS; private: AudioCoder _audioDecoder; CodecInst _codec; - WebRtc_Word32 _numberOf10MsPerFrame; - WebRtc_Word32 _numberOf10MsInDecoder; + int32_t _numberOf10MsPerFrame; + int32_t _numberOf10MsInDecoder; Resampler _resampler; float _scaling; @@ -83,37 +83,37 @@ private: class VideoFilePlayerImpl: public FilePlayerImpl { public: - VideoFilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat); + VideoFilePlayerImpl(uint32_t instanceID, FileFormats fileFormat); ~VideoFilePlayerImpl(); // FilePlayer functions. - virtual WebRtc_Word32 TimeUntilNextVideoFrame(); - virtual WebRtc_Word32 StartPlayingVideoFile(const char* fileName, - bool loop, - bool videoOnly); - virtual WebRtc_Word32 StopPlayingFile(); - virtual WebRtc_Word32 video_codec_info(VideoCodec& videoCodec) const; - virtual WebRtc_Word32 GetVideoFromFile(I420VideoFrame& videoFrame); - virtual WebRtc_Word32 GetVideoFromFile(I420VideoFrame& videoFrame, - const WebRtc_UWord32 outWidth, - const WebRtc_UWord32 outHeight); + virtual int32_t TimeUntilNextVideoFrame(); + virtual int32_t StartPlayingVideoFile(const char* fileName, + bool loop, + bool videoOnly); + virtual int32_t StopPlayingFile(); + virtual int32_t video_codec_info(VideoCodec& videoCodec) const; + virtual int32_t GetVideoFromFile(I420VideoFrame& videoFrame); + virtual int32_t GetVideoFromFile(I420VideoFrame& videoFrame, + const uint32_t outWidth, + const uint32_t outHeight); private: - WebRtc_Word32 SetUpVideoDecoder(); + int32_t SetUpVideoDecoder(); VideoCoder& _videoDecoder; VideoCodec video_codec_info_; - WebRtc_Word32 _decodedVideoFrames; + int32_t _decodedVideoFrames; EncodedVideoData& _encodedData; FrameScaler& _frameScaler; CriticalSectionWrapper* _critSec; TickTime _startTime; - WebRtc_Word64 _accumulatedRenderTimeMs; - WebRtc_UWord32 _frameLengthMS; + int64_t _accumulatedRenderTimeMs; + uint32_t _frameLengthMS; - WebRtc_Word32 _numberOfFramesRead; + int32_t _numberOfFramesRead; bool _videoOnly; }; #endif //WEBRTC_MODULE_UTILITY_VIDEO diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc index 840c79f7d..fefa4dc00 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.cc +++ b/webrtc/modules/utility/source/file_recorder_impl.cc @@ -29,7 +29,7 @@ #endif namespace webrtc { -FileRecorder* FileRecorder::CreateFileRecorder(WebRtc_UWord32 instanceID, +FileRecorder* FileRecorder::CreateFileRecorder(uint32_t instanceID, FileFormats fileFormat) { switch(fileFormat) @@ -60,7 +60,7 @@ void FileRecorder::DestroyFileRecorder(FileRecorder* recorder) delete recorder; } -FileRecorderImpl::FileRecorderImpl(WebRtc_UWord32 instanceID, +FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat) : _instanceID(instanceID), _fileFormat(fileFormat), @@ -83,7 +83,7 @@ FileFormats FileRecorderImpl::RecordingFileFormat() const return _fileFormat; } -WebRtc_Word32 FileRecorderImpl::RegisterModuleFileCallback( +int32_t FileRecorderImpl::RegisterModuleFileCallback( FileCallback* callback) { if(_moduleFile == NULL) @@ -93,10 +93,10 @@ WebRtc_Word32 FileRecorderImpl::RegisterModuleFileCallback( return _moduleFile->SetModuleFileCallback(callback); } -WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile( +int32_t FileRecorderImpl::StartRecordingAudioFile( const char* fileName, const CodecInst& codecInst, - WebRtc_UWord32 notificationTimeMs, + uint32_t notificationTimeMs, ACMAMRPackingFormat amrFormat) { if(_moduleFile == NULL) @@ -106,7 +106,7 @@ WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile( codec_info_ = codecInst; _amrFormat = amrFormat; - WebRtc_Word32 retVal = 0; + int32_t retVal = 0; if(_fileFormat != kFileFormatAviFile) { // AVI files should be started using StartRecordingVideoFile(..) all @@ -138,16 +138,16 @@ WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile( return retVal; } -WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile( +int32_t FileRecorderImpl::StartRecordingAudioFile( OutStream& destStream, const CodecInst& codecInst, - WebRtc_UWord32 notificationTimeMs, + uint32_t notificationTimeMs, ACMAMRPackingFormat amrFormat) { codec_info_ = codecInst; _amrFormat = amrFormat; - WebRtc_Word32 retVal = _moduleFile->StartRecordingAudioStream( + int32_t retVal = _moduleFile->StartRecordingAudioStream( destStream, _fileFormat, codecInst, @@ -174,7 +174,7 @@ WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile( return retVal; } -WebRtc_Word32 FileRecorderImpl::StopRecording() +int32_t FileRecorderImpl::StopRecording() { memset(&codec_info_, 0, sizeof(CodecInst)); return _moduleFile->StopRecording(); @@ -185,7 +185,7 @@ bool FileRecorderImpl::IsRecording() const return _moduleFile->IsRecording(); } -WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( +int32_t FileRecorderImpl::RecordAudioToFile( const AudioFrame& incomingAudioFrame, const TickTime* playoutTS) { @@ -209,7 +209,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; tempAudioFrame.samples_per_channel_ = incomingAudioFrame.samples_per_channel_; - for (WebRtc_UWord16 i = 0; + for (uint16_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) { // Sample value is the average of left and right buffer rounded to @@ -227,7 +227,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; tempAudioFrame.samples_per_channel_ = incomingAudioFrame.samples_per_channel_; - for (WebRtc_UWord16 i = 0; + for (uint16_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) { // Duplicate sample to both channels @@ -250,7 +250,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( // NOTE: stereo recording is only supported for WAV files. // TODO (hellner): WAV expect PCM in little endian byte order. Not // "encoding" with PCM coder should be a problem for big endian systems. - WebRtc_UWord32 encodedLenInBytes = 0; + uint32_t encodedLenInBytes = 0; if (_fileFormat == kFileFormatPreencodedFile || STR_CASE_CMP(codec_info_.plname, "L16") != 0) { @@ -277,7 +277,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( _audioResampler.Push(ptrAudioFrame->data_, ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_, - (WebRtc_Word16*)_audioBuffer, + (int16_t*)_audioBuffer, MAX_AUDIO_BUFFER_IN_BYTES, outLen); } else { _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, @@ -285,10 +285,10 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( kResamplerSynchronous); _audioResampler.Push(ptrAudioFrame->data_, ptrAudioFrame->samples_per_channel_, - (WebRtc_Word16*)_audioBuffer, + (int16_t*)_audioBuffer, MAX_AUDIO_BUFFER_IN_BYTES, outLen); } - encodedLenInBytes = outLen * sizeof(WebRtc_Word16); + encodedLenInBytes = outLen * sizeof(int16_t); } // Codec may not be operating at a frame rate of 10 ms. Whenever enough @@ -296,11 +296,11 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( // will be available. Wait until then. if (encodedLenInBytes) { - WebRtc_UWord16 msOfData = + uint16_t msOfData = ptrAudioFrame->samples_per_channel_ / - WebRtc_UWord16(ptrAudioFrame->sample_rate_hz_ / 1000); + uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000); if (WriteEncodedAudioData(_audioBuffer, - (WebRtc_UWord16)encodedLenInBytes, + (uint16_t)encodedLenInBytes, msOfData, playoutTS) == -1) { return -1; @@ -309,7 +309,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile( return 0; } -WebRtc_Word32 FileRecorderImpl::SetUpAudioEncoder() +int32_t FileRecorderImpl::SetUpAudioEncoder() { if (_fileFormat == kFileFormatPreencodedFile || STR_CASE_CMP(codec_info_.plname, "L16") != 0) @@ -328,7 +328,7 @@ WebRtc_Word32 FileRecorderImpl::SetUpAudioEncoder() return 0; } -WebRtc_Word32 FileRecorderImpl::codec_info(CodecInst& codecInst) const +int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const { if(codec_info_.plfreq == 0) { @@ -338,10 +338,10 @@ WebRtc_Word32 FileRecorderImpl::codec_info(CodecInst& codecInst) const return 0; } -WebRtc_Word32 FileRecorderImpl::WriteEncodedAudioData( - const WebRtc_Word8* audioBuffer, - WebRtc_UWord16 bufferLength, - WebRtc_UWord16 /*millisecondsOfData*/, +int32_t FileRecorderImpl::WriteEncodedAudioData( + const int8_t* audioBuffer, + uint16_t bufferLength, + uint16_t /*millisecondsOfData*/, const TickTime* /*playoutTS*/) { return _moduleFile->IncomingAudioData(audioBuffer, bufferLength); @@ -352,9 +352,9 @@ WebRtc_Word32 FileRecorderImpl::WriteEncodedAudioData( class AudioFrameFileInfo { public: - AudioFrameFileInfo(const WebRtc_Word8* audioData, - const WebRtc_UWord16 audioSize, - const WebRtc_UWord16 audioMS, + AudioFrameFileInfo(const int8_t* audioData, + const uint16_t audioSize, + const uint16_t audioMS, const TickTime& playoutTS) : _audioData(), _audioSize(audioSize), _audioMS(audioMS), _playoutTS(playoutTS) @@ -368,13 +368,13 @@ class AudioFrameFileInfo memcpy(_audioData, audioData, audioSize); }; // TODO (hellner): either turn into a struct or provide get/set functions. - WebRtc_Word8 _audioData[MAX_AUDIO_BUFFER_IN_BYTES]; - WebRtc_UWord16 _audioSize; - WebRtc_UWord16 _audioMS; + int8_t _audioData[MAX_AUDIO_BUFFER_IN_BYTES]; + uint16_t _audioSize; + uint16_t _audioMS; TickTime _playoutTS; }; -AviRecorder::AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat) +AviRecorder::AviRecorder(uint32_t instanceID, FileFormats fileFormat) : FileRecorderImpl(instanceID, fileFormat), _videoOnly(false), _thread( 0), @@ -403,7 +403,7 @@ AviRecorder::~AviRecorder( ) delete _critSec; } -WebRtc_Word32 AviRecorder::StartRecordingVideoFile( +int32_t AviRecorder::StartRecordingVideoFile( const char* fileName, const CodecInst& audioCodecInst, const VideoCodec& videoCodecInst, @@ -446,7 +446,7 @@ WebRtc_Word32 AviRecorder::StartRecordingVideoFile( return 0; } -WebRtc_Word32 AviRecorder::StopRecording() +int32_t AviRecorder::StopRecording() { _timeEvent.StopTimer(); @@ -454,12 +454,12 @@ WebRtc_Word32 AviRecorder::StopRecording() return FileRecorderImpl::StopRecording(); } -WebRtc_Word32 AviRecorder::CalcI420FrameSize( ) const +int32_t AviRecorder::CalcI420FrameSize( ) const { return 3 * _videoCodecInst.width * _videoCodecInst.height / 2; } -WebRtc_Word32 AviRecorder::SetUpVideoEncoder() +int32_t AviRecorder::SetUpVideoEncoder() { // Size of unencoded data (I420) should be the largest possible frame size // in a file. @@ -469,7 +469,7 @@ WebRtc_Word32 AviRecorder::SetUpVideoEncoder() _videoCodecInst.plType = _videoEncoder->DefaultPayloadType( _videoCodecInst.plName); - WebRtc_Word32 useNumberOfCores = 1; + int32_t useNumberOfCores = 1; // Set the max payload size to 16000. This means that the codec will try to // create slices that will fit in 16000 kByte packets. However, the // Encode() call will still generate one full frame. @@ -481,7 +481,7 @@ WebRtc_Word32 AviRecorder::SetUpVideoEncoder() return 0; } -WebRtc_Word32 AviRecorder::RecordVideoToFile(const I420VideoFrame& videoFrame) +int32_t AviRecorder::RecordVideoToFile(const I420VideoFrame& videoFrame) { CriticalSectionScoped lock(_critSec); if(!IsRecording() || videoFrame.IsZeroSize()) @@ -489,7 +489,7 @@ WebRtc_Word32 AviRecorder::RecordVideoToFile(const I420VideoFrame& videoFrame) return -1; } // The frame is written to file in AviRecorder::Process(). - WebRtc_Word32 retVal = _videoFramesQueue->AddFrame(videoFrame); + int32_t retVal = _videoFramesQueue->AddFrame(videoFrame); if(retVal != 0) { StopRecording(); @@ -540,7 +540,7 @@ bool AviRecorder::Run( ThreadObj threadObj) return static_cast( threadObj)->Process(); } -WebRtc_Word32 AviRecorder::ProcessAudio() +int32_t AviRecorder::ProcessAudio() { if (_writtenVideoFramesCounter == 0) { @@ -552,9 +552,9 @@ WebRtc_Word32 AviRecorder::ProcessAudio() { // Syncronize audio to the current frame to process by throwing away // audio samples with older timestamp than the video frame. - WebRtc_UWord32 numberOfAudioElements = + uint32_t numberOfAudioElements = _audioFramesToWrite.GetSize(); - for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i) + for (uint32_t i = 0; i < numberOfAudioElements; ++i) { AudioFrameFileInfo* frameInfo = (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); @@ -575,9 +575,9 @@ WebRtc_Word32 AviRecorder::ProcessAudio() } } // Write all audio up to current timestamp. - WebRtc_Word32 error = 0; - WebRtc_UWord32 numberOfAudioElements = _audioFramesToWrite.GetSize(); - for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i) + int32_t error = 0; + uint32_t numberOfAudioElements = _audioFramesToWrite.GetSize(); + for (uint32_t i = 0; i < numberOfAudioElements; ++i) { AudioFrameFileInfo* frameInfo = (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); @@ -626,7 +626,7 @@ bool AviRecorder::Process() { return true; } - WebRtc_Word32 error = 0; + int32_t error = 0; if(!_videoOnly) { if(!_firstAudioFrameReceived) @@ -646,7 +646,7 @@ bool AviRecorder::Process() "AviRecorder::Process() error writing to file."); break; } else { - WebRtc_UWord32 frameLengthMS = 1000 / + uint32_t frameLengthMS = 1000 / _videoCodecInst.maxFramerate; _writtenVideoFramesCounter++; _writtenVideoMS += frameLengthMS; @@ -656,7 +656,7 @@ bool AviRecorder::Process() // Frame rate is in frames per seconds. Frame length is // calculated as an integer division which means it may // be rounded down. Compensate for this every second. - WebRtc_UWord32 rest = 1000 % frameLengthMS; + uint32_t rest = 1000 % frameLengthMS; _writtenVideoMS += rest; } } @@ -667,10 +667,10 @@ bool AviRecorder::Process() // drift. Once a full frame worth of drift has happened, skip writing // one frame. Note that frame rate is in frames per second so the // drift is completely compensated for. - WebRtc_UWord32 frameLengthMS = 1000/_videoCodecInst.maxFramerate; - WebRtc_UWord32 restMS = 1000 % frameLengthMS; - WebRtc_UWord32 frameSkip = (_videoCodecInst.maxFramerate * - frameLengthMS) / restMS; + uint32_t frameLengthMS = 1000/_videoCodecInst.maxFramerate; + uint32_t restMS = 1000 % frameLengthMS; + uint32_t frameSkip = (_videoCodecInst.maxFramerate * + frameLengthMS) / restMS; _writtenVideoFramesCounter++; if(_writtenVideoFramesCounter % frameSkip == 0) @@ -691,7 +691,7 @@ bool AviRecorder::Process() return error == 0; } -WebRtc_Word32 AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) +int32_t AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) { if (!IsRecording() || videoFrame.IsZeroSize()) { @@ -731,7 +731,7 @@ WebRtc_Word32 AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) if(_videoEncodedData.payloadSize > 0) { if(_moduleFile->IncomingAVIVideoData( - (WebRtc_Word8*)(_videoEncodedData.payloadData), + (int8_t*)(_videoEncodedData.payloadData), _videoEncodedData.payloadSize)) { WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, @@ -751,10 +751,10 @@ WebRtc_Word32 AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) // Store audio frame in the _audioFramesToWrite buffer. The writing to file // happens in AviRecorder::Process(). -WebRtc_Word32 AviRecorder::WriteEncodedAudioData( - const WebRtc_Word8* audioBuffer, - WebRtc_UWord16 bufferLength, - WebRtc_UWord16 millisecondsOfData, +int32_t AviRecorder::WriteEncodedAudioData( + const int8_t* audioBuffer, + uint16_t bufferLength, + uint16_t millisecondsOfData, const TickTime* playoutTS) { if (!IsRecording()) diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h index 60d3b5c55..f8921d07d 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.h +++ b/webrtc/modules/utility/source/file_recorder_impl.h @@ -43,29 +43,29 @@ enum { kMaxAudioBufferQueueLength = 100 }; class FileRecorderImpl : public FileRecorder { public: - FileRecorderImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat); + FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat); virtual ~FileRecorderImpl(); // FileRecorder functions. - virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback); + virtual int32_t RegisterModuleFileCallback(FileCallback* callback); virtual FileFormats RecordingFileFormat() const; - virtual WebRtc_Word32 StartRecordingAudioFile( + virtual int32_t StartRecordingAudioFile( const char* fileName, const CodecInst& codecInst, - WebRtc_UWord32 notificationTimeMs, + uint32_t notificationTimeMs, ACMAMRPackingFormat amrFormat = AMRFileStorage); - virtual WebRtc_Word32 StartRecordingAudioFile( + virtual int32_t StartRecordingAudioFile( OutStream& destStream, const CodecInst& codecInst, - WebRtc_UWord32 notificationTimeMs, + uint32_t notificationTimeMs, ACMAMRPackingFormat amrFormat = AMRFileStorage); - virtual WebRtc_Word32 StopRecording(); + virtual int32_t StopRecording(); virtual bool IsRecording() const; - virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const; - virtual WebRtc_Word32 RecordAudioToFile( + virtual int32_t codec_info(CodecInst& codecInst) const; + virtual int32_t RecordAudioToFile( const AudioFrame& frame, const TickTime* playoutTS = NULL); - virtual WebRtc_Word32 StartRecordingVideoFile( + virtual int32_t StartRecordingVideoFile( const char* fileName, const CodecInst& audioCodecInst, const VideoCodec& videoCodecInst, @@ -74,21 +74,21 @@ public: { return -1; } - virtual WebRtc_Word32 RecordVideoToFile(const I420VideoFrame& videoFrame) + virtual int32_t RecordVideoToFile(const I420VideoFrame& videoFrame) { return -1; } protected: - virtual WebRtc_Word32 WriteEncodedAudioData( - const WebRtc_Word8* audioBuffer, - WebRtc_UWord16 bufferLength, - WebRtc_UWord16 millisecondsOfData, + virtual int32_t WriteEncodedAudioData( + const int8_t* audioBuffer, + uint16_t bufferLength, + uint16_t millisecondsOfData, const TickTime* playoutTS); - WebRtc_Word32 SetUpAudioEncoder(); + int32_t SetUpAudioEncoder(); - WebRtc_UWord32 _instanceID; + uint32_t _instanceID; FileFormats _fileFormat; MediaFile* _moduleFile; @@ -96,7 +96,7 @@ private: CodecInst codec_info_; ACMAMRPackingFormat _amrFormat; - WebRtc_Word8 _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES]; + int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES]; AudioCoder _audioEncoder; Resampler _audioResampler; }; @@ -106,24 +106,24 @@ private: class AviRecorder : public FileRecorderImpl { public: - AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat); + AviRecorder(uint32_t instanceID, FileFormats fileFormat); virtual ~AviRecorder(); // FileRecorder functions. - virtual WebRtc_Word32 StartRecordingVideoFile( + virtual int32_t StartRecordingVideoFile( const char* fileName, const CodecInst& audioCodecInst, const VideoCodec& videoCodecInst, ACMAMRPackingFormat amrFormat = AMRFileStorage, bool videoOnly = false); - virtual WebRtc_Word32 StopRecording(); - virtual WebRtc_Word32 RecordVideoToFile(const I420VideoFrame& videoFrame); + virtual int32_t StopRecording(); + virtual int32_t RecordVideoToFile(const I420VideoFrame& videoFrame); protected: - virtual WebRtc_Word32 WriteEncodedAudioData( - const WebRtc_Word8* audioBuffer, - WebRtc_UWord16 bufferLength, - WebRtc_UWord16 millisecondsOfData, + virtual int32_t WriteEncodedAudioData( + const int8_t* audioBuffer, + uint16_t bufferLength, + uint16_t millisecondsOfData, const TickTime* playoutTS); private: static bool Run(ThreadObj threadObj); @@ -132,11 +132,11 @@ private: bool StartThread(); bool StopThread(); - WebRtc_Word32 EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame); - WebRtc_Word32 ProcessAudio(); + int32_t EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame); + int32_t ProcessAudio(); - WebRtc_Word32 CalcI420FrameSize() const; - WebRtc_Word32 SetUpVideoEncoder(); + int32_t CalcI420FrameSize() const; + int32_t SetUpVideoEncoder(); VideoCodec _videoCodecInst; bool _videoOnly; @@ -148,15 +148,15 @@ private: FrameScaler* _frameScaler; VideoCoder* _videoEncoder; - WebRtc_Word32 _videoMaxPayloadSize; + int32_t _videoMaxPayloadSize; EncodedVideoData _videoEncodedData; ThreadWrapper* _thread; EventWrapper& _timeEvent; CriticalSectionWrapper* _critSec; - WebRtc_Word64 _writtenVideoFramesCounter; - WebRtc_Word64 _writtenAudioMS; - WebRtc_Word64 _writtenVideoMS; + int64_t _writtenVideoFramesCounter; + int64_t _writtenAudioMS; + int64_t _writtenVideoMS; }; #endif // WEBRTC_MODULE_UTILITY_VIDEO } // namespace webrtc diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc index bdbd5ca08..61b3e332f 100644 --- a/webrtc/modules/utility/source/process_thread_impl.cc +++ b/webrtc/modules/utility/source/process_thread_impl.cc @@ -42,7 +42,7 @@ ProcessThreadImpl::~ProcessThreadImpl() WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__); } -WebRtc_Word32 ProcessThreadImpl::Start() +int32_t ProcessThreadImpl::Start() { CriticalSectionScoped lock(_critSectModules); if(_thread) @@ -52,7 +52,7 @@ WebRtc_Word32 ProcessThreadImpl::Start() _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority, "ProcessThread"); unsigned int id; - WebRtc_Word32 retVal = _thread->Start(id); + int32_t retVal = _thread->Start(id); if(retVal >= 0) { return 0; @@ -62,7 +62,7 @@ WebRtc_Word32 ProcessThreadImpl::Start() return -1; } -WebRtc_Word32 ProcessThreadImpl::Stop() +int32_t ProcessThreadImpl::Stop() { _critSectModules->Enter(); if(_thread) @@ -87,13 +87,13 @@ WebRtc_Word32 ProcessThreadImpl::Stop() return 0; } -WebRtc_Word32 ProcessThreadImpl::RegisterModule(const Module* module) +int32_t ProcessThreadImpl::RegisterModule(const Module* module) { CriticalSectionScoped lock(_critSectModules); // Only allow module to be registered once. ListItem* item = _modules.First(); - for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++) + for(uint32_t i = 0; i < _modules.GetSize() && item; i++) { if(module == item->GetItem()) { @@ -113,12 +113,12 @@ WebRtc_Word32 ProcessThreadImpl::RegisterModule(const Module* module) return 0; } -WebRtc_Word32 ProcessThreadImpl::DeRegisterModule(const Module* module) +int32_t ProcessThreadImpl::DeRegisterModule(const Module* module) { CriticalSectionScoped lock(_critSectModules); ListItem* item = _modules.First(); - for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++) + for(uint32_t i = 0; i < _modules.GetSize() && item; i++) { if(module == item->GetItem()) { @@ -142,13 +142,13 @@ bool ProcessThreadImpl::Process() { // Wait for the module that should be called next, but don't block thread // longer than 100 ms. - WebRtc_Word32 minTimeToNext = 100; + int32_t minTimeToNext = 100; { CriticalSectionScoped lock(_critSectModules); ListItem* item = _modules.First(); - for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++) + for(uint32_t i = 0; i < _modules.GetSize() && item; i++) { - WebRtc_Word32 timeToNext = + int32_t timeToNext = static_cast(item->GetItem())->TimeUntilNextProcess(); if(minTimeToNext > timeToNext) { @@ -173,9 +173,9 @@ bool ProcessThreadImpl::Process() { CriticalSectionScoped lock(_critSectModules); ListItem* item = _modules.First(); - for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++) + for(uint32_t i = 0; i < _modules.GetSize() && item; i++) { - WebRtc_Word32 timeToNext = + int32_t timeToNext = static_cast(item->GetItem())->TimeUntilNextProcess(); if(timeToNext < 1) { diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h index 79b12725d..7edb56534 100644 --- a/webrtc/modules/utility/source/process_thread_impl.h +++ b/webrtc/modules/utility/source/process_thread_impl.h @@ -25,11 +25,11 @@ public: ProcessThreadImpl(); virtual ~ProcessThreadImpl(); - virtual WebRtc_Word32 Start(); - virtual WebRtc_Word32 Stop(); + virtual int32_t Start(); + virtual int32_t Stop(); - virtual WebRtc_Word32 RegisterModule(const Module* module); - virtual WebRtc_Word32 DeRegisterModule(const Module* module); + virtual int32_t RegisterModule(const Module* module); + virtual int32_t DeRegisterModule(const Module* module); protected: static bool Run(void* obj); diff --git a/webrtc/modules/utility/source/rtp_dump_impl.cc b/webrtc/modules/utility/source/rtp_dump_impl.cc index 69a52ecc5..74de4ac61 100644 --- a/webrtc/modules/utility/source/rtp_dump_impl.cc +++ b/webrtc/modules/utility/source/rtp_dump_impl.cc @@ -40,7 +40,7 @@ namespace webrtc { const char RTPFILE_VERSION[] = "1.0"; -const WebRtc_UWord32 MAX_UWORD32 = 0xffffffff; +const uint32_t MAX_UWORD32 = 0xffffffff; // This stucture is specified in the rtpdump documentation. // This struct corresponds to RD_packet_t in @@ -49,11 +49,11 @@ typedef struct { // Length of packet, including this header (may be smaller than plen if not // whole packet recorded). - WebRtc_UWord16 length; + uint16_t length; // Actual header+payload length for RTP, 0 for RTCP. - WebRtc_UWord16 plen; + uint16_t plen; // Milliseconds since the start of recording. - WebRtc_UWord32 offset; + uint32_t offset; } rtpDumpPktHdr_t; RtpDump* RtpDump::CreateRtpDump() @@ -87,7 +87,7 @@ RtpDumpImpl::~RtpDumpImpl() WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__); } -WebRtc_Word32 RtpDumpImpl::Start(const char* fileNameUTF8) +int32_t RtpDumpImpl::Start(const char* fileNameUTF8) { if (fileNameUTF8 == NULL) @@ -136,7 +136,7 @@ WebRtc_Word32 RtpDumpImpl::Start(const char* fileNameUTF8) return 0; } -WebRtc_Word32 RtpDumpImpl::Stop() +int32_t RtpDumpImpl::Stop() { CriticalSectionScoped lock(_critSect); _file.Flush(); @@ -150,8 +150,7 @@ bool RtpDumpImpl::IsActive() const return _file.Open(); } -WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet, - WebRtc_UWord16 packetLength) +int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength) { CriticalSectionScoped lock(_critSect); if (!IsActive()) @@ -174,7 +173,7 @@ WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet, bool isRTCP = RTCP(packet); rtpDumpPktHdr_t hdr; - WebRtc_UWord32 offset; + uint32_t offset; // Offset is relative to when recording was started. offset = GetTimeInMS(); @@ -187,14 +186,14 @@ WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet, } hdr.offset = RtpDumpHtonl(offset); - hdr.length = RtpDumpHtons((WebRtc_UWord16)(packetLength + sizeof(hdr))); + hdr.length = RtpDumpHtons((uint16_t)(packetLength + sizeof(hdr))); if (isRTCP) { hdr.plen = 0; } else { - hdr.plen = RtpDumpHtons((WebRtc_UWord16)packetLength); + hdr.plen = RtpDumpHtons((uint16_t)packetLength); } if (!_file.Write(&hdr, sizeof(hdr))) @@ -213,9 +212,9 @@ WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet, return 0; } -bool RtpDumpImpl::RTCP(const WebRtc_UWord8* packet) const +bool RtpDumpImpl::RTCP(const uint8_t* packet) const { - const WebRtc_UWord8 payloadType = packet[1]; + const uint8_t payloadType = packet[1]; bool is_rtcp = false; switch(payloadType) @@ -234,7 +233,7 @@ bool RtpDumpImpl::RTCP(const WebRtc_UWord8* packet) const } // TODO (hellner): why is TickUtil not used here? -inline WebRtc_UWord32 RtpDumpImpl::GetTimeInMS() const +inline uint32_t RtpDumpImpl::GetTimeInMS() const { #if defined(_WIN32) return timeGetTime(); @@ -253,7 +252,7 @@ inline WebRtc_UWord32 RtpDumpImpl::GetTimeInMS() const #endif } -inline WebRtc_UWord32 RtpDumpImpl::RtpDumpHtonl(WebRtc_UWord32 x) const +inline uint32_t RtpDumpImpl::RtpDumpHtonl(uint32_t x) const { #if defined(WEBRTC_BIG_ENDIAN) return x; @@ -267,7 +266,7 @@ inline WebRtc_UWord32 RtpDumpImpl::RtpDumpHtonl(WebRtc_UWord32 x) const #endif } -inline WebRtc_UWord16 RtpDumpImpl::RtpDumpHtons(WebRtc_UWord16 x) const +inline uint16_t RtpDumpImpl::RtpDumpHtons(uint16_t x) const { #if defined(WEBRTC_BIG_ENDIAN) return x; diff --git a/webrtc/modules/utility/source/rtp_dump_impl.h b/webrtc/modules/utility/source/rtp_dump_impl.h index 9715c352c..bb72dff3e 100644 --- a/webrtc/modules/utility/source/rtp_dump_impl.h +++ b/webrtc/modules/utility/source/rtp_dump_impl.h @@ -22,28 +22,27 @@ public: RtpDumpImpl(); virtual ~RtpDumpImpl(); - virtual WebRtc_Word32 Start(const char* fileNameUTF8); - virtual WebRtc_Word32 Stop(); + virtual int32_t Start(const char* fileNameUTF8); + virtual int32_t Stop(); virtual bool IsActive() const; - virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet, - WebRtc_UWord16 packetLength); + virtual int32_t DumpPacket(const uint8_t* packet, uint16_t packetLength); private: // Return the system time in ms. - inline WebRtc_UWord32 GetTimeInMS() const; + inline uint32_t GetTimeInMS() const; // Return x in network byte order (big endian). - inline WebRtc_UWord32 RtpDumpHtonl(WebRtc_UWord32 x) const; + inline uint32_t RtpDumpHtonl(uint32_t x) const; // Return x in network byte order (big endian). - inline WebRtc_UWord16 RtpDumpHtons(WebRtc_UWord16 x) const; + inline uint16_t RtpDumpHtons(uint16_t x) const; // Return true if the packet starts with a valid RTCP header. // Note: See ModuleRTPUtility::RTPHeaderParser::RTCP() for details on how // to determine if the packet is an RTCP packet. - bool RTCP(const WebRtc_UWord8* packet) const; + bool RTCP(const uint8_t* packet) const; private: CriticalSectionWrapper* _critSect; FileWrapper& _file; - WebRtc_UWord32 _startTime; + uint32_t _startTime; }; } // namespace webrtc #endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_ diff --git a/webrtc/modules/utility/source/video_coder.cc b/webrtc/modules/utility/source/video_coder.cc index cc331614b..01e4ae052 100644 --- a/webrtc/modules/utility/source/video_coder.cc +++ b/webrtc/modules/utility/source/video_coder.cc @@ -13,7 +13,7 @@ #include "video_coder.h" namespace webrtc { -VideoCoder::VideoCoder(WebRtc_UWord32 instanceID) +VideoCoder::VideoCoder(uint32_t instanceID) : _vcm(VideoCodingModule::Create(instanceID)), _decodedVideo(0) { @@ -29,7 +29,7 @@ VideoCoder::~VideoCoder() VideoCodingModule::Destroy(_vcm); } -WebRtc_Word32 VideoCoder::ResetDecoder() +int32_t VideoCoder::ResetDecoder() { _vcm->ResetDecoder(); @@ -41,9 +41,9 @@ WebRtc_Word32 VideoCoder::ResetDecoder() return 0; } -WebRtc_Word32 VideoCoder::SetEncodeCodec(VideoCodec& videoCodecInst, - WebRtc_UWord32 numberOfCores, - WebRtc_UWord32 maxPayloadSize) +int32_t VideoCoder::SetEncodeCodec(VideoCodec& videoCodecInst, + uint32_t numberOfCores, + uint32_t maxPayloadSize) { if(_vcm->RegisterSendCodec(&videoCodecInst, numberOfCores, maxPayloadSize) != VCM_OK) @@ -54,12 +54,12 @@ WebRtc_Word32 VideoCoder::SetEncodeCodec(VideoCodec& videoCodecInst, } -WebRtc_Word32 VideoCoder::SetDecodeCodec(VideoCodec& videoCodecInst, - WebRtc_Word32 numberOfCores) +int32_t VideoCoder::SetDecodeCodec(VideoCodec& videoCodecInst, + int32_t numberOfCores) { if (videoCodecInst.plType == 0) { - WebRtc_Word8 plType = DefaultPayloadType(videoCodecInst.plName); + int8_t plType = DefaultPayloadType(videoCodecInst.plName); if (plType == -1) { return -1; @@ -74,8 +74,8 @@ WebRtc_Word32 VideoCoder::SetDecodeCodec(VideoCodec& videoCodecInst, return 0; } -WebRtc_Word32 VideoCoder::Decode(I420VideoFrame& decodedVideo, - const EncodedVideoData& encodedData) +int32_t VideoCoder::Decode(I420VideoFrame& decodedVideo, + const EncodedVideoData& encodedData) { decodedVideo.ResetSize(); if(encodedData.payloadSize <= 0) @@ -92,8 +92,8 @@ WebRtc_Word32 VideoCoder::Decode(I420VideoFrame& decodedVideo, } -WebRtc_Word32 VideoCoder::Encode(const I420VideoFrame& videoFrame, - EncodedVideoData& videoEncodedData) +int32_t VideoCoder::Encode(const I420VideoFrame& videoFrame, + EncodedVideoData& videoEncodedData) { // The AddVideoFrame(..) call will (indirectly) call SendData(). Store a // pointer to videoFrame so that it can be updated. @@ -106,11 +106,11 @@ WebRtc_Word32 VideoCoder::Encode(const I420VideoFrame& videoFrame, return 0; } -WebRtc_Word8 VideoCoder::DefaultPayloadType(const char* plName) +int8_t VideoCoder::DefaultPayloadType(const char* plName) { VideoCodec tmpCodec; - WebRtc_Word32 numberOfCodecs = _vcm->NumberOfCodecs(); - for (WebRtc_UWord8 i = 0; i < numberOfCodecs; i++) + int32_t numberOfCodecs = _vcm->NumberOfCodecs(); + for (uint8_t i = 0; i < numberOfCodecs; i++) { _vcm->Codec(i, &tmpCodec); if(strncmp(tmpCodec.plName, plName, kPayloadNameSize) == 0) @@ -121,18 +121,18 @@ WebRtc_Word8 VideoCoder::DefaultPayloadType(const char* plName) return -1; } -WebRtc_Word32 VideoCoder::FrameToRender(I420VideoFrame& videoFrame) +int32_t VideoCoder::FrameToRender(I420VideoFrame& videoFrame) { return _decodedVideo->CopyFrame(videoFrame); } -WebRtc_Word32 VideoCoder::SendData( +int32_t VideoCoder::SendData( const FrameType frameType, - const WebRtc_UWord8 payloadType, - const WebRtc_UWord32 timeStamp, + const uint8_t payloadType, + const uint32_t timeStamp, int64_t capture_time_ms, - const WebRtc_UWord8* payloadData, - WebRtc_UWord32 payloadSize, + const uint8_t* payloadData, + uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* /*rtpVideoHdr*/) { @@ -144,7 +144,7 @@ WebRtc_Word32 VideoCoder::SendData( _videoEncodedData->timeStamp = timeStamp; _videoEncodedData->fragmentationHeader.CopyFrom(fragmentationHeader); memcpy(_videoEncodedData->payloadData, payloadData, - sizeof(WebRtc_UWord8) * payloadSize); + sizeof(uint8_t) * payloadSize); _videoEncodedData->payloadSize = payloadSize; return 0; } diff --git a/webrtc/modules/utility/source/video_coder.h b/webrtc/modules/utility/source/video_coder.h index b1d8c7d42..c69c3e46d 100644 --- a/webrtc/modules/utility/source/video_coder.h +++ b/webrtc/modules/utility/source/video_coder.h @@ -20,43 +20,42 @@ namespace webrtc { class VideoCoder : public VCMPacketizationCallback, public VCMReceiveCallback { public: - VideoCoder(WebRtc_UWord32 instanceID); + VideoCoder(uint32_t instanceID); ~VideoCoder(); - WebRtc_Word32 ResetDecoder(); + int32_t ResetDecoder(); - WebRtc_Word32 SetEncodeCodec(VideoCodec& videoCodecInst, - WebRtc_UWord32 numberOfCores, - WebRtc_UWord32 maxPayloadSize); + int32_t SetEncodeCodec(VideoCodec& videoCodecInst, + uint32_t numberOfCores, + uint32_t maxPayloadSize); // Select the codec that should be used for decoding. videoCodecInst.plType // will be set to the codec's default payload type. - WebRtc_Word32 SetDecodeCodec(VideoCodec& videoCodecInst, - WebRtc_Word32 numberOfCores); + int32_t SetDecodeCodec(VideoCodec& videoCodecInst, int32_t numberOfCores); - WebRtc_Word32 Decode(I420VideoFrame& decodedVideo, - const EncodedVideoData& encodedData); + int32_t Decode(I420VideoFrame& decodedVideo, + const EncodedVideoData& encodedData); - WebRtc_Word32 Encode(const I420VideoFrame& videoFrame, - EncodedVideoData& videoEncodedData); + int32_t Encode(const I420VideoFrame& videoFrame, + EncodedVideoData& videoEncodedData); - WebRtc_Word8 DefaultPayloadType(const char* plName); + int8_t DefaultPayloadType(const char* plName); private: // VCMReceiveCallback function. // Note: called by VideoCodingModule when decoding finished. - WebRtc_Word32 FrameToRender(I420VideoFrame& videoFrame); + int32_t FrameToRender(I420VideoFrame& videoFrame); // VCMPacketizationCallback function. // Note: called by VideoCodingModule when encoding finished. - WebRtc_Word32 SendData( + int32_t SendData( FrameType /*frameType*/, - WebRtc_UWord8 /*payloadType*/, - WebRtc_UWord32 /*timeStamp*/, + uint8_t /*payloadType*/, + uint32_t /*timeStamp*/, int64_t capture_time_ms, - const WebRtc_UWord8* payloadData, - WebRtc_UWord32 payloadSize, + const uint8_t* payloadData, + uint32_t payloadSize, const RTPFragmentationHeader& /* fragmentationHeader*/, const RTPVideoHeader* rtpTypeHdr); diff --git a/webrtc/modules/utility/source/video_frames_queue.cc b/webrtc/modules/utility/source/video_frames_queue.cc index 5a1ea596f..039c9e83e 100644 --- a/webrtc/modules/utility/source/video_frames_queue.cc +++ b/webrtc/modules/utility/source/video_frames_queue.cc @@ -47,7 +47,7 @@ VideoFramesQueue::~VideoFramesQueue() { } } -WebRtc_Word32 VideoFramesQueue::AddFrame(const I420VideoFrame& newFrame) { +int32_t VideoFramesQueue::AddFrame(const I420VideoFrame& newFrame) { I420VideoFrame* ptrFrameToAdd = NULL; // Try to re-use a VideoFrame. Only allocate new memory if it is necessary. if (!_emptyFrames.Empty()) { @@ -112,7 +112,7 @@ I420VideoFrame* VideoFramesQueue::FrameToRecord() { return ptrRenderFrame; } -WebRtc_Word32 VideoFramesQueue::ReturnFrame(I420VideoFrame* ptrOldFrame) { +int32_t VideoFramesQueue::ReturnFrame(I420VideoFrame* ptrOldFrame) { ptrOldFrame->set_timestamp(0); ptrOldFrame->set_width(0); ptrOldFrame->set_height(0); @@ -122,7 +122,7 @@ WebRtc_Word32 VideoFramesQueue::ReturnFrame(I420VideoFrame* ptrOldFrame) { return 0; } -WebRtc_Word32 VideoFramesQueue::SetRenderDelay(WebRtc_UWord32 renderDelay) { +int32_t VideoFramesQueue::SetRenderDelay(uint32_t renderDelay) { _renderDelayMs = renderDelay; return 0; } diff --git a/webrtc/modules/utility/source/video_frames_queue.h b/webrtc/modules/utility/source/video_frames_queue.h index 3f63f6575..af6821ee1 100644 --- a/webrtc/modules/utility/source/video_frames_queue.h +++ b/webrtc/modules/utility/source/video_frames_queue.h @@ -26,7 +26,7 @@ class VideoFramesQueue { ~VideoFramesQueue(); // Put newFrame (last) in the queue. - WebRtc_Word32 AddFrame(const I420VideoFrame& newFrame); + int32_t AddFrame(const I420VideoFrame& newFrame); // Return the most current frame. I.e. the frame with the highest // VideoFrame::RenderTimeMs() that is lower than @@ -34,12 +34,12 @@ class VideoFramesQueue { I420VideoFrame* FrameToRecord(); // Set the render delay estimate to renderDelay ms. - WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 renderDelay); + int32_t SetRenderDelay(uint32_t renderDelay); protected: // Make ptrOldFrame available for re-use. I.e. put it in the empty frames // queue. - WebRtc_Word32 ReturnFrame(I420VideoFrame* ptrOldFrame); + int32_t ReturnFrame(I420VideoFrame* ptrOldFrame); private: // Don't allow the buffer to expand beyond KMaxNumberOfFrames VideoFrames. @@ -54,7 +54,7 @@ class VideoFramesQueue { ListWrapper _emptyFrames; // Estimated render delay. - WebRtc_UWord32 _renderDelayMs; + uint32_t _renderDelayMs; }; } // namespace webrtc #endif // WEBRTC_MODULE_UTILITY_VIDEO diff --git a/webrtc/modules/utility/test/testAPI.cc b/webrtc/modules/utility/test/testAPI.cc index 3408a86a6..31e103058 100644 --- a/webrtc/modules/utility/test/testAPI.cc +++ b/webrtc/modules/utility/test/testAPI.cc @@ -38,29 +38,29 @@ bool notify = false, playing = false, recording = false; class MyFileModuleCallback : public FileCallback { public: - virtual void PlayNotification( const WebRtc_Word32 id, - const WebRtc_UWord32 durationMs ) + virtual void PlayNotification( const int32_t id, + const uint32_t durationMs ) { printf("\tReceived PlayNotification from module %ld, durationMs = %ld\n", id, durationMs); notify = true; }; - virtual void RecordNotification( const WebRtc_Word32 id, - const WebRtc_UWord32 durationMs ) + virtual void RecordNotification( const int32_t id, + const uint32_t durationMs ) { printf("\tReceived RecordNotification from module %ld, durationMs = %ld\n", id, durationMs); notify = true; }; - virtual void PlayFileEnded(const WebRtc_Word32 id) + virtual void PlayFileEnded(const int32_t id) { printf("\tReceived PlayFileEnded notification from module %ld.\n", id); playing = false; }; - virtual void RecordFileEnded(const WebRtc_Word32 id) + virtual void RecordFileEnded(const int32_t id) { printf("\tReceived RecordFileEnded notification from module %ld.\n", id); recording = false; @@ -151,14 +151,14 @@ int main(int /*argc*/, char** /*argv*/) ::Sleep(10); } } - WebRtc_UWord32 decodedDataLengthInSamples; + uint32_t decodedDataLengthInSamples; if( 0 != filePlayer.Get10msAudioFromFile( audioFrame.data_, decodedDataLengthInSamples, audioCodec.plfreq)) { audioNotDone = false; } else { audioFrame.sample_rate_hz_ = filePlayer.Frequency(); - audioFrame.samples_per_channel_ = (WebRtc_UWord16)decodedDataLengthInSamples; + audioFrame.samples_per_channel_ = (uint16_t)decodedDataLengthInSamples; fileRecorder.RecordAudioToFile(audioFrame, &TickTime::Now()); } } @@ -210,7 +210,7 @@ int main(int /*argc*/, char** /*argv*/) assert(fileRecorder.IsRecording()); - const WebRtc_UWord32 KVideoWriteSize = static_cast< WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3) / 2); + const uint32_t KVideoWriteSize = static_cast< uint32_t>( (videoCodec.width * videoCodec.height * 3) / 2); webrtc::VideoFrame videoFrame; // 10 ms @@ -339,7 +339,7 @@ int main(int /*argc*/, char** /*argv*/) } } - WebRtc_UWord32 decodedDataLengthInSamples; + uint32_t decodedDataLengthInSamples; if( 0 != filePlayer.Get10msAudioFromFile( audioFrame.data_, decodedDataLengthInSamples, audioCodec.plfreq)) { audioNotDone = false; @@ -348,7 +348,7 @@ int main(int /*argc*/, char** /*argv*/) { ::Sleep(5); audioFrame.sample_rate_hz_ = filePlayer.Frequency(); - audioFrame.samples_per_channel_ = (WebRtc_UWord16)decodedDataLengthInSamples; + audioFrame.samples_per_channel_ = (uint16_t)decodedDataLengthInSamples; assert(0 == fileRecorder.RecordAudioToFile(audioFrame)); audioFrameCount++;