This CL changes all the freq relevant variables to be int type. So it will take away the VoE "comparison between signed and unsigned integer expressions" warnings.

BR,
/SX
Review URL: http://webrtc-codereview.appspot.com/89014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@320 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@google.com 2011-08-08 08:18:44 +00:00
parent 81520b7955
commit 0b0665acc1
12 changed files with 36 additions and 42 deletions

View File

@ -256,8 +256,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return kNullPointerError;
}
if (frame->_frequencyInHz !=
static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
if (frame->_frequencyInHz != sample_rate_hz_) {
return kBadSampleRateError;
}
@ -382,8 +381,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return kNullPointerError;
}
if (frame->_frequencyInHz !=
static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
if (frame->_frequencyInHz != sample_rate_hz_) {
return kBadSampleRateError;
}

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@ -697,7 +697,7 @@ public:
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
const WebRtc_UWord32 frequencyInHz,
const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel = 1,
@ -719,7 +719,7 @@ public:
// Supporting Stereo, stereo samples are interleaved
mutable WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
WebRtc_UWord16 _payloadDataLengthInSamples;
WebRtc_UWord32 _frequencyInHz;
int _frequencyInHz;
WebRtc_UWord8 _audioChannel;
SpeechType _speechType;
VADActivity _vadActivity;
@ -756,7 +756,7 @@ AudioFrame::UpdateFrame(
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
const WebRtc_UWord32 frequencyInHz,
const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel,

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@ -159,7 +159,7 @@ public:
virtual WebRtc_Word32 OnInitializeDecoder(const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) = 0;

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@ -3142,7 +3142,7 @@ void ViEChannel::OnApplicationDataReceived(const WebRtc_Word32 id,
WebRtc_Word32 ViEChannel::OnInitializeDecoder(
const WebRtc_Word32 id, const WebRtc_Word8 payloadType,
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency, const WebRtc_UWord8 channels,
const int frequency, const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate)
{
WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(_engineId, _channelId),

View File

@ -193,7 +193,7 @@ public:
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate);

View File

@ -630,7 +630,7 @@ Channel::OnInitializeDecoder(
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate)
{
@ -5860,7 +5860,7 @@ Channel::Demultiplex(const AudioFrame& audioFrame,
}
WebRtc_UWord32
Channel::PrepareEncodeAndSend(WebRtc_UWord32 mixingFrequency)
Channel::PrepareEncodeAndSend(int mixingFrequency)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::PrepareEncodeAndSend()");
@ -6216,7 +6216,7 @@ Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
}
WebRtc_Word32
Channel::MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency)
Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
{
WebRtc_Word16 fileBuffer[320];
WebRtc_UWord32 fileSamples(0);
@ -6278,7 +6278,7 @@ Channel::MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency)
WebRtc_Word32
Channel::MixAudioWithFile(AudioFrame& audioFrame,
const WebRtc_UWord32 mixingFrequency)
const int mixingFrequency)
{
assert(mixingFrequency <= 32000);

View File

@ -389,7 +389,7 @@ public:
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate);
@ -533,15 +533,15 @@ public:
#endif
WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame,
const WebRtc_UWord8 audioLevel_dBov);
WebRtc_UWord32 PrepareEncodeAndSend(WebRtc_UWord32 mixingFrequency);
WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);
WebRtc_UWord32 EncodeAndSend();
private:
int InsertInbandDtmfTone();
WebRtc_Word32
MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency);
MixOrReplaceAudioWithFile(const int mixingFrequency);
WebRtc_Word32 MixAudioWithFile(AudioFrame& audioFrame,
const WebRtc_UWord32 mixingFrequency);
const int mixingFrequency);
WebRtc_Word32 GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp);
void UpdateDeadOrAliveCounters(bool alive);
WebRtc_Word32 SendPacketRaw(const void *data, int len, bool RTCP);
@ -576,9 +576,9 @@ private:
FilePlayer* _inputFilePlayerPtr;
FilePlayer* _outputFilePlayerPtr;
FileRecorder* _outputFileRecorderPtr;
WebRtc_UWord32 _inputFilePlayerId;
WebRtc_UWord32 _outputFilePlayerId;
WebRtc_UWord32 _outputFileRecorderId;
int _inputFilePlayerId;
int _outputFilePlayerId;
int _outputFileRecorderId;
bool _inputFilePlaying;
bool _outputFilePlaying;
bool _outputFileRecording;

View File

@ -142,12 +142,12 @@ private: // owns
Resampler _apmResampler; // converts mixed audio to fit APM rate
AudioLevel _audioLevel; // measures audio level for the combined signal
DtmfInband _dtmfGenerator;
WebRtc_UWord32 _instanceId;
int _instanceId;
VoEMediaProcess* _externalMediaCallbackPtr;
bool _externalMedia;
float _panLeft;
float _panRight;
WebRtc_UWord32 _mixingFrequencyHz;
int _mixingFrequencyHz;
FileRecorder* _outputFileRecorderPtr;
bool _outputFileRecording;
};

View File

@ -314,7 +314,7 @@ TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
totalDelayMS, clockDrift, currentMicLevel);
const WebRtc_UWord32 mixingFrequency = _mixingFrequency;
const int mixingFrequency = _mixingFrequency;
ScopedChannel sc(*_channelManagerPtr);
void* iterator(NULL);
@ -326,7 +326,7 @@ TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
{
CodecInst tmpCdc;
channelPtr->GetSendCodec(tmpCdc);
if ((WebRtc_UWord32) tmpCdc.plfreq > _mixingFrequency)
if (tmpCdc.plfreq > _mixingFrequency)
_mixingFrequency = tmpCdc.plfreq;
}
channelPtr = sc.GetNextChannel(iterator);
@ -1151,7 +1151,7 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
const WebRtc_UWord32 nSamples,
const WebRtc_UWord8 nChannels,
const WebRtc_UWord32 samplesPerSec,
const WebRtc_UWord32 mixingFrequency)
const int mixingFrequency)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::GenerateAudioFrame(nSamples=%u,"
@ -1214,12 +1214,11 @@ WebRtc_Word32 TransmitMixer::RecordAudioToFile(
}
WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
const WebRtc_UWord32 mixingFrequency)
const int mixingFrequency)
{
WebRtc_Word16 fileBuffer[320];
WebRtc_UWord32 fileSamples(0);
WebRtc_Word32 outSamples(0);
{
CriticalSectionScoped cs(_critSect);
@ -1387,7 +1386,6 @@ int TransmitMixer::TypingDetection()
{
return (-1);
}
bool vad = (_audioFrame._vadActivity == AudioFrame::kVadActive);
if (_audioFrame._vadActivity == AudioFrame::kVadActive)
_timeActive++;
@ -1423,7 +1421,7 @@ int TransmitMixer::TypingDetection()
}
#endif
WebRtc_UWord32 TransmitMixer::GetMixingFrequency()
int TransmitMixer::GetMixingFrequency()
{
assert(_mixingFrequency!=0);
return (_mixingFrequency);

View File

@ -81,7 +81,7 @@ public:
int DeRegisterExternalMediaProcessing();
WebRtc_UWord32 GetMixingFrequency();
int GetMixingFrequency();
// VoEVolumeControl
int SetMute(const bool enable);
@ -160,11 +160,11 @@ private:
const WebRtc_UWord32 nSamples,
const WebRtc_UWord8 nChannels,
const WebRtc_UWord32 samplesPerSec,
const WebRtc_UWord32 mixingFrequency);
const int mixingFrequency);
WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency);
WebRtc_Word32 MixOrReplaceAudioWithFile(
const WebRtc_UWord32 mixingFrequency);
const int mixingFrequency);
WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS,
const WebRtc_Word32 clockDrift,
@ -188,9 +188,9 @@ private: // owns
FilePlayer* _filePlayerPtr;
FileRecorder* _fileRecorderPtr;
FileRecorder* _fileCallRecorderPtr;
WebRtc_UWord32 _filePlayerId;
WebRtc_UWord32 _fileRecorderId;
WebRtc_UWord32 _fileCallRecorderId;
int _filePlayerId;
int _fileRecorderId;
int _fileCallRecorderId;
bool _filePlaying;
bool _fileRecording;
bool _fileCallRecording;
@ -208,14 +208,14 @@ private: // owns
WebRtc_UWord32 _noiseWarning;
private:
WebRtc_UWord32 _instanceId;
int _instanceId;
bool _mixFileWithMicrophone;
WebRtc_UWord32 _captureLevel;
bool _externalMedia;
VoEMediaProcess* _externalMediaCallbackPtr;
bool _mute;
WebRtc_Word32 _remainingMuteMicTimeMs;
WebRtc_UWord32 _mixingFrequency;
int _mixingFrequency;
bool _includeAudioLevelIndication;
WebRtc_UWord8 _audioLevel_dBov;
};

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@ -282,7 +282,8 @@ WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
_outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, audioFrame);
assert(nSamples == audioFrame._payloadDataLengthInSamples);
assert(samplesPerSec == audioFrame._frequencyInHz);
assert(samplesPerSec ==
static_cast<WebRtc_UWord32>(audioFrame._frequencyInHz));
// Deliver audio (PCM) samples to the ADM
memcpy(
@ -585,7 +586,6 @@ int VoEBaseImpl::Init()
}
bool available(false);
WebRtc_Word32 ret(0);
// --------------------
// Reinitialize the ADM

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@ -85,8 +85,6 @@ int VoECallReportImpl::ResetCallReportStatistics(int channel)
}
assert(_audioProcessingModulePtr != NULL);
int res1(0);
int res2(0);
bool levelMode =
_audioProcessingModulePtr->level_estimator()->is_enabled();
bool echoMode =