Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
index c85a19781dab9f36a0a037491292c528982449d6..6d0f7a4627bc49cad1680b11f34fb26090b9ff1e 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
@@ -25,7 +25,6 @@ static const int kDtmfFrequencyHz = 8000; |
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender) |
: _clock(clock), |
_rtpSender(rtpSender), |
- _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
_packetSizeSamples(160), |
_dtmfEventIsOn(false), |
_dtmfEventFirstPacketSent(false), |
@@ -54,7 +53,7 @@ int RTPSenderAudio::AudioFrequency() const { |
// set audio packet size, used to determine when it's time to send a DTMF packet |
// in silence (CNG) |
int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) { |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
_packetSizeSamples = packetSizeSamples; |
return 0; |
@@ -68,7 +67,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload( |
const uint32_t rate, |
RtpUtility::Payload** payload) { |
if (RtpUtility::StringCompare(payloadName, "cn", 2)) { |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
// we can have multiple CNG payload types |
switch (frequency) { |
case 8000: |
@@ -87,7 +86,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload( |
return -1; |
} |
} else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) { |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
// Don't add it to the list |
// we dont want to allow send with a DTMF payloadtype |
_dtmfPayloadType = payloadType; |
@@ -105,7 +104,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload( |
} |
bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) { |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
// for audio true for first packet in a speech burst |
bool markerBit = false; |
if (_lastPayloadType != payload_type) { |
@@ -163,7 +162,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType, |
int8_t dtmf_payload_type; |
uint16_t packet_size_samples; |
{ |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
red_payload_type = _REDPayloadType; |
audio_level_dbov = _audioLevel_dBov; |
dtmf_payload_type = _dtmfPayloadType; |
@@ -336,7 +335,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType, |
} |
{ |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
_lastPayloadType = payloadType; |
} |
// Update audio level extension, if included. |
@@ -365,7 +364,7 @@ int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) { |
if (level_dBov > 127) { |
return -1; |
} |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
_audioLevel_dBov = level_dBov; |
return 0; |
} |
@@ -375,14 +374,14 @@ int32_t RTPSenderAudio::SetRED(int8_t payloadType) { |
if (payloadType < -1) { |
return -1; |
} |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
_REDPayloadType = payloadType; |
return 0; |
} |
// Get payload type for Redundant Audio Data RFC 2198 |
int32_t RTPSenderAudio::RED(int8_t* payloadType) const { |
- CriticalSectionScoped cs(_sendAudioCritsect.get()); |
+ rtc::CritScope cs(&_sendAudioCritsect); |
if (_REDPayloadType == -1) { |
// not configured |
return -1; |
@@ -396,7 +395,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key, |
uint16_t time_ms, |
uint8_t level) { |
{ |
- CriticalSectionScoped lock(_sendAudioCritsect.get()); |
+ rtc::CritScope lock(&_sendAudioCritsect); |
if (_dtmfPayloadType < 0) { |
// TelephoneEvent payloadtype not configured |
return -1; |