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Remove saturation warning support from TransmitMixer.

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BUG=none

Review-Url: https://codereview.webrtc.org/2720253002
Cr-Commit-Position: refs/heads/master@{#16913}
This commit is contained in:
tommi 2017-02-28 08:25:11 -08:00 committed by Commit bot
parent e3e902eef9
commit ba08a140da
4 changed files with 17 additions and 41 deletions
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1
webrtc/voice_engine/include/voe_errors.h
View File

@ -80,7 +80,6 @@
#define VE_RECEIVE_SOCKETS_CONFLICT 8105
#define VE_SEND_SOCKETS_CONFLICT 8106
#define VE_TYPING_NOISE_WARNING 8107
#define VE_SATURATION_WARNING 8108
#define VE_NOISE_WARNING 8109
#define VE_CANNOT_GET_SEND_CODEC 8110
#define VE_CANNOT_GET_REC_CODEC 8111

2
webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
View File

@ -81,8 +81,6 @@ void MyObserver::CallbackOnError(int channel, int err_code) {
printf(" RUNTIME PLAY WARNING \n");
} else if (err_code == VE_RUNTIME_REC_WARNING) {
printf(" RUNTIME RECORD WARNING \n");
} else if (err_code == VE_SATURATION_WARNING) {
printf(" SATURATION WARNING \n");
} else if (err_code == VE_RUNTIME_PLAY_ERROR) {
printf(" RUNTIME PLAY ERROR \n");
} else if (err_code == VE_RUNTIME_REC_ERROR) {

46
webrtc/voice_engine/transmit_mixer.cc
View File

@ -26,14 +26,13 @@
namespace webrtc {
namespace voe {
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
// TODO(ajm): The thread safety of this is dubious...
void
TransmitMixer::OnPeriodicProcess()
void TransmitMixer::OnPeriodicProcess()
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::OnPeriodicProcess()");
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
bool send_typing_noise_warning = false;
bool typing_noise_detected = false;
{
@ -64,32 +63,8 @@ TransmitMixer::OnPeriodicProcess()
}
}
}
#endif // WEBRTC_VOICE_ENGINE_TYPING_DETECTION
bool saturationWarning = false;
{
// Modify |_saturationWarning| under lock to avoid conflict with write op
// in ProcessAudio and also ensure that we don't hold the lock during the
// callback.
rtc::CritScope cs(&_critSect);
saturationWarning = _saturationWarning;
if (_saturationWarning)
_saturationWarning = false;
}
if (saturationWarning)
{
rtc::CritScope cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::OnPeriodicProcess() =>"
" CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
}
}
}
#endif // WEBRTC_VOICE_ENGINE_TYPING_DETECTION
void TransmitMixer::PlayNotification(int32_t id,
uint32_t durationMs)
@ -176,12 +151,14 @@ TransmitMixer::Destroy(TransmitMixer*& mixer)
}
TransmitMixer::TransmitMixer(uint32_t instanceId) :
_monitorModule(this),
// Avoid conflict with other channels by adding 1024 - 1026,
// won't use as much as 1024 channels.
_filePlayerId(instanceId + 1024),
_fileRecorderId(instanceId + 1025),
_fileCallRecorderId(instanceId + 1026),
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
_monitorModule(this),
#endif
_instanceId(instanceId)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
@ -192,10 +169,10 @@ TransmitMixer::~TransmitMixer()
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::~TransmitMixer() - dtor");
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
if (_processThreadPtr)
{
_processThreadPtr->DeRegisterModule(&_monitorModule);
}
#endif
{
rtc::CritScope cs(&_critSect);
if (file_recorder_) {
@ -225,8 +202,9 @@ TransmitMixer::SetEngineInformation(ProcessThread& processThread,
_engineStatisticsPtr = &engineStatistics;
_channelManagerPtr = &channelManager;
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
_processThreadPtr->RegisterModule(&_monitorModule);
#endif
return 0;
}
@ -1077,10 +1055,6 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
// Store new capture level. Only updated when analog AGC is enabled.
_captureLevel = agc->stream_analog_level();
rtc::CritScope cs(&_critSect);
// Triggers a callback in OnPeriodicProcess().
_saturationWarning |= agc->stream_is_saturated();
}
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION

9
webrtc/voice_engine/transmit_mixer.h
View File

@ -132,8 +132,10 @@ public:
virtual ~TransmitMixer();
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
// Periodic callback from the MonitorModule.
void OnPeriodicProcess();
#endif
// FileCallback
void PlayNotification(const int32_t id,
@ -161,7 +163,11 @@ public:
bool IsStereoChannelSwappingEnabled();
protected:
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
TransmitMixer() : _monitorModule(this) {}
#else
TransmitMixer() = default;
#endif
private:
TransmitMixer(uint32_t instanceId);
@ -194,7 +200,6 @@ private:
ProcessThread* _processThreadPtr = nullptr;
// owns
MonitorModule<TransmitMixer> _monitorModule;
AudioFrame _audioFrame;
PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
std::unique_ptr<FilePlayer> file_player_;
@ -212,11 +217,11 @@ private:
rtc::CriticalSection _callbackCritSect;
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
MonitorModule<TransmitMixer> _monitorModule;
webrtc::TypingDetection _typingDetection;
bool _typingNoiseWarningPending = false;
bool _typingNoiseDetected = false;
#endif
bool _saturationWarning = false;
int _instanceId = 0;
bool _mixFileWithMicrophone = false;