Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(203)

Unified Diff: webrtc/voice_engine/channel.h

Issue 1776243003: Revert of - Clean up unused voice engine DTMF code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_1
Patch Set: Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/voice_engine/BUILD.gn ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index c89b0e0c8bb9a36efbc4dda32aa6e211cc5c1f22..75c4fd87cb96467aed48a6ba0c27202b4641dc84 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -25,6 +25,8 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/dtmf_inband.h"
+#include "webrtc/voice_engine/dtmf_inband_queue.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/level_indicator.h"
@@ -294,9 +296,17 @@
// VoEVideoSyncExtended
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
- // DTMF
- int SendTelephoneEventOutband(int event, int duration_ms);
- int SetSendTelephoneEventPayloadType(int payload_type);
+ // VoEDtmf
+ int SendTelephoneEventOutband(unsigned char eventCode,
+ int lengthMs,
+ int attenuationDb,
+ bool playDtmfEvent);
+ int SendTelephoneEventInband(unsigned char eventCode,
+ int lengthMs,
+ int attenuationDb,
+ bool playDtmfEvent);
+ int SetSendTelephoneEventPayloadType(unsigned char type);
+ int GetSendTelephoneEventPayloadType(unsigned char& type);
// VoEAudioProcessingImpl
int UpdateRxVadDetection(AudioFrame& audioFrame);
@@ -454,6 +464,7 @@
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
+ int InsertInbandDtmfTone();
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
void UpdatePlayoutTimestamp(bool rtcp);
@@ -499,10 +510,13 @@
int _outputFilePlayerId;
int _outputFileRecorderId;
bool _outputFileRecording;
+ DtmfInbandQueue _inbandDtmfQueue;
+ DtmfInband _inbandDtmfGenerator;
bool _outputExternalMedia;
VoEMediaProcess* _inputExternalMediaCallbackPtr;
VoEMediaProcess* _outputExternalMediaCallbackPtr;
uint32_t _timeStamp;
+ uint8_t _sendTelephoneEventPayloadType;
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
@@ -546,6 +560,9 @@
float _panLeft;
float _panRight;
float _outputGain;
+ // VoEDtmf
+ bool _playOutbandDtmfEvent;
+ bool _playInbandDtmfEvent;
// VoeRTP_RTCP
uint32_t _lastLocalTimeStamp;
int8_t _lastPayloadType;
« no previous file with comments | « webrtc/voice_engine/BUILD.gn ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698