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 de728f0860511491cdf5c4ace62dabad751aa451..2f3faf5d739231aa9ba1bcaa43a1a32592533fa4 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
@@ -20,34 +20,31 @@ namespace webrtc { |
static const int kDtmfFrequencyHz = 8000; |
-RTPSenderAudio::RTPSenderAudio(const int32_t id, |
- Clock* clock, |
+RTPSenderAudio::RTPSenderAudio(Clock* clock, |
RTPSender* rtpSender, |
- RtpAudioFeedback* audio_feedback) : |
- _id(id), |
- _clock(clock), |
- _rtpSender(rtpSender), |
- _audioFeedback(audio_feedback), |
- _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
- _packetSizeSamples(160), |
- _dtmfEventIsOn(false), |
- _dtmfEventFirstPacketSent(false), |
- _dtmfPayloadType(-1), |
- _dtmfTimestamp(0), |
- _dtmfKey(0), |
- _dtmfLengthSamples(0), |
- _dtmfLevel(0), |
- _dtmfTimeLastSent(0), |
- _dtmfTimestampLastSent(0), |
- _REDPayloadType(-1), |
- _inbandVADactive(false), |
- _cngNBPayloadType(-1), |
- _cngWBPayloadType(-1), |
- _cngSWBPayloadType(-1), |
- _cngFBPayloadType(-1), |
- _lastPayloadType(-1), |
- _audioLevel_dBov(0) { |
-} |
+ RtpAudioFeedback* audio_feedback) |
+ : _clock(clock), |
+ _rtpSender(rtpSender), |
+ _audioFeedback(audio_feedback), |
+ _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
+ _packetSizeSamples(160), |
+ _dtmfEventIsOn(false), |
+ _dtmfEventFirstPacketSent(false), |
+ _dtmfPayloadType(-1), |
+ _dtmfTimestamp(0), |
+ _dtmfKey(0), |
+ _dtmfLengthSamples(0), |
+ _dtmfLevel(0), |
+ _dtmfTimeLastSent(0), |
+ _dtmfTimestampLastSent(0), |
+ _REDPayloadType(-1), |
+ _inbandVADactive(false), |
+ _cngNBPayloadType(-1), |
+ _cngWBPayloadType(-1), |
+ _cngSWBPayloadType(-1), |
+ _cngFBPayloadType(-1), |
+ _lastPayloadType(-1), |
+ _audioLevel_dBov(0) {} |
RTPSenderAudio::~RTPSenderAudio() { |
} |
@@ -204,7 +201,7 @@ int32_t RTPSenderAudio::SendAudio( |
} |
if (dtmfToneStarted) { |
if (_audioFeedback) |
- _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel); |
+ _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel); |
} |
// A source MAY send events and coded audio packets for the same time |