diff options
| author | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-03-12 22:27:27 +0000 |
|---|---|---|
| committer | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-03-12 22:27:27 +0000 |
| commit | b7edd065306329309dac6767fe4914c185f941f8 (patch) | |
| tree | 35df823fe64f79a68ba6f1c645588c9eded92683 /webrtc/modules/rtp_rtcp/interface | |
| parent | 728b7ea245739ae3ab628c39d8f9913fee4ee788 (diff) | |
| download | webrtc-b7edd065306329309dac6767fe4914c185f941f8.tar.gz | |
Remove DTMF detection. Talk team has been in the loop and there is no need for
DTMF detection at the receiver side.
test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'webrtc/modules/rtp_rtcp/interface')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h | 16 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h | 7 |
2 files changed, 4 insertions, 19 deletions
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h index 7fcef2099c..b436e3ff99 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h @@ -13,8 +13,8 @@ #include <vector> -#include "modules/interface/module.h" -#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" namespace webrtc { // Forward declarations. @@ -796,19 +796,11 @@ class RtpRtcp : public Module { const WebRtc_UWord16 packetSizeSamples) = 0; /* - * Outband TelephoneEvent(DTMF) detection + * Forward DTMF to decoder for playout. * * return -1 on failure else 0 */ - virtual WebRtc_Word32 SetTelephoneEventStatus( - const bool enable, - const bool forwardToDecoder, - const bool detectEndOfTone = false) = 0; - - /* - * Is outband TelephoneEvent(DTMF) turned on/off? - */ - virtual bool TelephoneEvent() const = 0; + virtual int SetTelephoneEventForwardToDecoder(bool forwardToDecoder) = 0; /* * Returns true if received DTMF events are forwarded to the decoder using diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index 38891999dd..2b13d39518 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -207,9 +207,6 @@ protected: class RtpAudioFeedback { public: - virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id, - const WebRtc_UWord8 event, - const bool endOfEvent) = 0; virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id, const WebRtc_UWord8 event, @@ -304,10 +301,6 @@ class NullRtpAudioFeedback : public RtpAudioFeedback { public: virtual ~NullRtpAudioFeedback() {} - virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id, - const WebRtc_UWord8 event, - const bool endOfEvent) {} - virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id, const WebRtc_UWord8 event, const WebRtc_UWord16 lengthMs, |