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 c0f827e80f8cebb3f9f96c4ffd1fa852659ef21b..5dc82fc8fce2dfb6aeb504d537ada76088d4f8fc 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
@@ -169,14 +169,14 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type, |
} |
// Check if we have pending DTMFs to send |
- if (!dtmf_event_is_on_ && PendingDTMF()) { |
+ if (!dtmf_event_is_on_ && dtmf_queue_.PendingDTMF()) { |
int64_t delaySinceLastDTMF = |
clock_->TimeInMilliseconds() - dtmf_time_last_sent_; |
if (delaySinceLastDTMF > 100) { |
// New tone to play |
dtmf_timestamp_ = rtp_timestamp; |
- if (NextDTMF(&key, &dtmf_length_ms, &dtmf_level_) >= 0) { |
+ if (dtmf_queue_.NextDTMF(&key, &dtmf_length_ms, &dtmf_level_) >= 0) { |
dtmf_event_first_packet_sent_ = false; |
dtmf_key_ = key; |
dtmf_length_samples_ = (kDtmfFrequencyHz / 1000) * dtmf_length_ms; |
@@ -310,7 +310,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key, |
return -1; |
} |
} |
- return AddDTMF(key, time_ms, level); |
+ return dtmf_queue_.AddDTMF(key, time_ms, level); |
} |
bool RTPSenderAudio::SendTelephoneEventPacket(bool ended, |