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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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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
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