Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index a9cf2b26dc8edea51bb8d56b57d13a52275e58b2..db5b2169b29b41c1284cb98a36d48dd29073a12a 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -460,7 +460,7 @@ bool Channel::SendRtp(const uint8_t* data, |
rtc::CritScope cs(&_callbackCritSect); |
- if (_transportPtr == NULL) { |
+ if (_transportPtr == nullptr) { |
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::SendPacket() failed to send RTP packet due to" |
" invalid transport object"); |
@@ -486,7 +486,7 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) { |
"Channel::SendRtcp(len=%" PRIuS ")", len); |
rtc::CritScope cs(&_callbackCritSect); |
- if (_transportPtr == NULL) { |
+ if (_transportPtr == nullptr) { |
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::SendRtcp() failed to send RTCP packet" |
" due to invalid transport object"); |
@@ -589,8 +589,8 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData, |
} |
int64_t round_trip_time = 0; |
- _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL, |
- NULL); |
+ _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, nullptr, nullptr, |
+ nullptr); |
std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time); |
if (!nack_list.empty()) { |
@@ -826,7 +826,7 @@ int32_t Channel::CreateChannel( |
instanceId); |
channel = new Channel(channelId, instanceId, config); |
- if (channel == NULL) { |
+ if (channel == nullptr) { |
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId), |
"Channel::CreateChannel() unable to allocate memory for" |
" channel"); |
@@ -908,8 +908,8 @@ Channel::Channel(int32_t channelId, |
_outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026), |
_outputFileRecording(false), |
_outputExternalMedia(false), |
- _inputExternalMediaCallbackPtr(NULL), |
- _outputExternalMediaCallbackPtr(NULL), |
+ _inputExternalMediaCallbackPtr(nullptr), |
+ _outputExternalMediaCallbackPtr(nullptr), |
_timeStamp(0), // This is just an offset, RTP module will add it's own |
// random offset |
ntp_estimator_(Clock::GetRealTimeClock()), |
@@ -921,14 +921,14 @@ Channel::Channel(int32_t channelId, |
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()), |
capture_start_rtp_time_stamp_(-1), |
capture_start_ntp_time_ms_(-1), |
- _engineStatisticsPtr(NULL), |
- _outputMixerPtr(NULL), |
- _transmitMixerPtr(NULL), |
- _moduleProcessThreadPtr(NULL), |
- _audioDeviceModulePtr(NULL), |
- _voiceEngineObserverPtr(NULL), |
- _callbackCritSectPtr(NULL), |
- _transportPtr(NULL), |
+ _engineStatisticsPtr(nullptr), |
+ _outputMixerPtr(nullptr), |
+ _transmitMixerPtr(nullptr), |
+ _moduleProcessThreadPtr(nullptr), |
+ _audioDeviceModulePtr(nullptr), |
+ _voiceEngineObserverPtr(nullptr), |
+ _callbackCritSectPtr(nullptr), |
+ _transportPtr(nullptr), |
_sendFrameType(0), |
_externalMixing(false), |
_mixFileWithMicrophone(false), |
@@ -988,7 +988,7 @@ Channel::Channel(int32_t channelId, |
} |
Channel::~Channel() { |
- rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL); |
+ rtp_receive_statistics_->RegisterRtcpStatisticsCallback(nullptr); |
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::~Channel() - dtor"); |
@@ -1004,15 +1004,15 @@ Channel::~Channel() { |
{ |
rtc::CritScope cs(&_fileCritSect); |
if (input_file_player_) { |
- input_file_player_->RegisterModuleFileCallback(NULL); |
+ input_file_player_->RegisterModuleFileCallback(nullptr); |
input_file_player_->StopPlayingFile(); |
} |
if (output_file_player_) { |
- output_file_player_->RegisterModuleFileCallback(NULL); |
+ output_file_player_->RegisterModuleFileCallback(nullptr); |
output_file_player_->StopPlayingFile(); |
} |
if (output_file_recorder_) { |
- output_file_recorder_->RegisterModuleFileCallback(NULL); |
+ output_file_recorder_->RegisterModuleFileCallback(nullptr); |
output_file_recorder_->StopRecording(); |
} |
} |
@@ -1021,12 +1021,12 @@ Channel::~Channel() { |
// 1. De-register callbacks in modules |
// 2. De-register modules in process thread |
// 3. Destroy modules |
- if (audio_coding_->RegisterTransportCallback(NULL) == -1) { |
+ if (audio_coding_->RegisterTransportCallback(nullptr) == -1) { |
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), |
"~Channel() failed to de-register transport callback" |
" (Audio coding module)"); |
} |
- if (audio_coding_->RegisterVADCallback(NULL) == -1) { |
+ if (audio_coding_->RegisterVADCallback(nullptr) == -1) { |
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), |
"~Channel() failed to de-register VAD callback" |
" (Audio coding module)"); |
@@ -1045,7 +1045,8 @@ int32_t Channel::Init() { |
// --- Initial sanity |
- if ((_engineStatisticsPtr == NULL) || (_moduleProcessThreadPtr == NULL)) { |
+ if ((_engineStatisticsPtr == nullptr) || |
+ (_moduleProcessThreadPtr == nullptr)) { |
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::Init() must call SetEngineInformation() first"); |
return -1; |
@@ -1311,7 +1312,7 @@ int32_t Channel::DeRegisterVoiceEngineObserver() { |
"DeRegisterVoiceEngineObserver() observer already disabled"); |
return 0; |
} |
- _voiceEngineObserverPtr = NULL; |
+ _voiceEngineObserverPtr = nullptr; |
return 0; |
} |
@@ -1630,7 +1631,7 @@ int32_t Channel::DeRegisterExternalTransport() { |
"disabled"); |
} |
_externalTransport = false; |
- _transportPtr = NULL; |
+ _transportPtr = nullptr; |
return 0; |
} |
@@ -1728,7 +1729,8 @@ bool Channel::IsPacketRetransmitted(const RTPHeader& header, |
return false; |
// Check if this is a retransmission. |
int64_t min_rtt = 0; |
- _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); |
+ _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, |
+ nullptr); |
return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt); |
} |
@@ -1769,7 +1771,7 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) { |
uint32_t ntp_frac = 0; |
uint32_t rtp_timestamp = 0; |
if (0 != |
- _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, |
+ _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr, |
&rtp_timestamp)) { |
// Waiting for RTCP. |
return 0; |
@@ -1807,7 +1809,7 @@ int Channel::StartPlayingFileLocally(const char* fileName, |
rtc::CritScope cs(&_fileCritSect); |
if (output_file_player_) { |
- output_file_player_->RegisterModuleFileCallback(NULL); |
+ output_file_player_->RegisterModuleFileCallback(nullptr); |
output_file_player_.reset(); |
} |
@@ -1854,10 +1856,10 @@ int Channel::StartPlayingFileLocally(InStream* stream, |
" volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
format, volumeScaling, startPosition, stopPosition); |
- if (stream == NULL) { |
+ if (stream == nullptr) { |
_engineStatisticsPtr->SetLastError( |
VE_BAD_FILE, kTraceError, |
- "StartPlayingFileLocally() NULL as input stream"); |
+ "StartPlayingFileLocally() null as input stream"); |
return -1; |
} |
@@ -1873,7 +1875,7 @@ int Channel::StartPlayingFileLocally(InStream* stream, |
// Destroy the old instance |
if (output_file_player_) { |
- output_file_player_->RegisterModuleFileCallback(NULL); |
+ output_file_player_->RegisterModuleFileCallback(nullptr); |
output_file_player_.reset(); |
} |
@@ -1927,7 +1929,7 @@ int Channel::StopPlayingFileLocally() { |
"StopPlayingFile() could not stop playing"); |
return -1; |
} |
- output_file_player_->RegisterModuleFileCallback(NULL); |
+ output_file_player_->RegisterModuleFileCallback(nullptr); |
output_file_player_.reset(); |
channel_state_.SetOutputFilePlaying(false); |
} |
@@ -2001,7 +2003,7 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName, |
// Destroy the old instance |
if (input_file_player_) { |
- input_file_player_->RegisterModuleFileCallback(NULL); |
+ input_file_player_->RegisterModuleFileCallback(nullptr); |
input_file_player_.reset(); |
} |
@@ -2045,10 +2047,10 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream, |
"volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
format, volumeScaling, startPosition, stopPosition); |
- if (stream == NULL) { |
+ if (stream == nullptr) { |
_engineStatisticsPtr->SetLastError( |
VE_BAD_FILE, kTraceError, |
- "StartPlayingFileAsMicrophone NULL as input stream"); |
+ "StartPlayingFileAsMicrophone null as input stream"); |
return -1; |
} |
@@ -2063,7 +2065,7 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream, |
// Destroy the old instance |
if (input_file_player_) { |
- input_file_player_->RegisterModuleFileCallback(NULL); |
+ input_file_player_->RegisterModuleFileCallback(nullptr); |
input_file_player_.reset(); |
} |
@@ -2113,7 +2115,7 @@ int Channel::StopPlayingFileAsMicrophone() { |
"StopPlayingFile() could not stop playing"); |
return -1; |
} |
- input_file_player_->RegisterModuleFileCallback(NULL); |
+ input_file_player_->RegisterModuleFileCallback(nullptr); |
input_file_player_.reset(); |
channel_state_.SetInputFilePlaying(false); |
@@ -2139,14 +2141,14 @@ int Channel::StartRecordingPlayout(const char* fileName, |
const uint32_t notificationTime(0); // Not supported in VoE |
CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000}; |
- if ((codecInst != NULL) && |
+ if ((codecInst != nullptr) && |
((codecInst->channels < 1) || (codecInst->channels > 2))) { |
_engineStatisticsPtr->SetLastError( |
VE_BAD_ARGUMENT, kTraceError, |
"StartRecordingPlayout() invalid compression"); |
return (-1); |
} |
- if (codecInst == NULL) { |
+ if (codecInst == nullptr) { |
format = kFileFormatPcm16kHzFile; |
codecInst = &dummyCodec; |
} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) || |
@@ -2161,7 +2163,7 @@ int Channel::StartRecordingPlayout(const char* fileName, |
// Destroy the old instance |
if (output_file_recorder_) { |
- output_file_recorder_->RegisterModuleFileCallback(NULL); |
+ output_file_recorder_->RegisterModuleFileCallback(nullptr); |
output_file_recorder_.reset(); |
} |
@@ -2204,13 +2206,13 @@ int Channel::StartRecordingPlayout(OutStream* stream, |
const uint32_t notificationTime(0); // Not supported in VoE |
CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000}; |
- if (codecInst != NULL && codecInst->channels != 1) { |
+ if (codecInst != nullptr && codecInst->channels != 1) { |
_engineStatisticsPtr->SetLastError( |
VE_BAD_ARGUMENT, kTraceError, |
"StartRecordingPlayout() invalid compression"); |
return (-1); |
} |
- if (codecInst == NULL) { |
+ if (codecInst == nullptr) { |
format = kFileFormatPcm16kHzFile; |
codecInst = &dummyCodec; |
} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) || |
@@ -2225,7 +2227,7 @@ int Channel::StartRecordingPlayout(OutStream* stream, |
// Destroy the old instance |
if (output_file_recorder_) { |
- output_file_recorder_->RegisterModuleFileCallback(NULL); |
+ output_file_recorder_->RegisterModuleFileCallback(nullptr); |
output_file_recorder_.reset(); |
} |
@@ -2272,7 +2274,7 @@ int Channel::StopRecordingPlayout() { |
"StopRecording() could not stop recording"); |
return (-1); |
} |
- output_file_recorder_->RegisterModuleFileCallback(NULL); |
+ output_file_recorder_->RegisterModuleFileCallback(nullptr); |
output_file_recorder_.reset(); |
_outputFileRecording = false; |
@@ -2500,7 +2502,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) { |
} |
int Channel::GetRemoteRTCP_CNAME(char cName[256]) { |
- if (cName == NULL) { |
+ if (cName == nullptr) { |
_engineStatisticsPtr->SetLastError( |
VE_INVALID_ARGUMENT, kTraceError, |
"GetRemoteRTCP_CNAME() invalid CNAME input buffer"); |
@@ -2547,7 +2549,7 @@ int Channel::GetRemoteRTCPData(unsigned int& NTPHigh, |
// has been received) |
playoutTimestamp = playout_timestamp_rtcp_; |
- if (NULL != jitter || NULL != fractionLost) { |
+ if (nullptr != jitter || nullptr != fractionLost) { |
// Get all RTCP receiver report blocks that have been received on this |
// channel. If we receive RTP packets from a remote source we know the |
// remote SSRC and use the report block from him. |
@@ -2600,7 +2602,7 @@ int Channel::SendApplicationDefinedRTCPPacket( |
"SendApplicationDefinedRTCPPacket() not sending"); |
return -1; |
} |
- if (NULL == data) { |
+ if (nullptr == data) { |
_engineStatisticsPtr->SetLastError( |
VE_INVALID_ARGUMENT, kTraceError, |
"SendApplicationDefinedRTCPPacket() invalid data value"); |
@@ -2664,7 +2666,7 @@ int Channel::GetRTPStatistics(unsigned int& averageJitterMs, |
int Channel::GetRemoteRTCPReportBlocks( |
std::vector<ReportBlock>* report_blocks) { |
- if (report_blocks == NULL) { |
+ if (report_blocks == nullptr) { |
_engineStatisticsPtr->SetLastError( |
VE_INVALID_ARGUMENT, kTraceError, |
"GetRemoteRTCPReportBlock()s invalid report_blocks."); |
@@ -2894,7 +2896,7 @@ void Channel::DisassociateSendChannel(int channel_id) { |
if (channel && channel->ChannelId() == channel_id) { |
// If this channel is associated with a send channel of the specified |
// Channel ID, disassociate with it. |
- ChannelOwner ref(NULL); |
+ ChannelOwner ref(nullptr); |
associate_send_channel_ = ref; |
} |
} |
@@ -2972,7 +2974,7 @@ int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type) { |
return 0; |
} |
_outputExternalMedia = false; |
- _outputExternalMediaCallbackPtr = NULL; |
+ _outputExternalMediaCallbackPtr = nullptr; |
} else if (kRecordingPerChannel == type) { |
if (!_inputExternalMediaCallbackPtr) { |
_engineStatisticsPtr->SetLastError( |
@@ -2982,7 +2984,7 @@ int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type) { |
return 0; |
} |
channel_state_.SetInputExternalMedia(false); |
- _inputExternalMediaCallbackPtr = NULL; |
+ _inputExternalMediaCallbackPtr = nullptr; |
} |
return 0; |