| 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),
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| - _cngWBPayloadType(-1),
|
| - _cngSWBPayloadType(-1),
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| - _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
|
|
|