Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index 92098634fcdba8bbf2a51715eb572dc38a0bd30f..65c3debdf076eceab6421468a7c13e09b335cf4a 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -145,20 +145,11 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { |
class FakeWebRtcVoiceEngine |
: public webrtc::VoEAudioProcessing, |
- public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf, |
+ public webrtc::VoEBase, public webrtc::VoECodec, |
public webrtc::VoEHardware, |
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP, |
public webrtc::VoEVolumeControl { |
public: |
- struct DtmfInfo { |
- DtmfInfo() |
- : dtmf_event_code(-1), |
- dtmf_out_of_band(false), |
- dtmf_length_ms(-1) {} |
- int dtmf_event_code; |
- bool dtmf_out_of_band; |
- int dtmf_length_ms; |
- }; |
struct Channel { |
explicit Channel() |
: external_transport(false), |
@@ -173,7 +164,6 @@ class FakeWebRtcVoiceEngine |
nack(false), |
cn8_type(13), |
cn16_type(105), |
- dtmf_type(106), |
red_type(117), |
nack_max_packets(0), |
send_ssrc(0), |
@@ -195,12 +185,10 @@ class FakeWebRtcVoiceEngine |
bool nack; |
int cn8_type; |
int cn16_type; |
- int dtmf_type; |
int red_type; |
int nack_max_packets; |
uint32_t send_ssrc; |
int associate_send_channel; |
- DtmfInfo dtmf_info; |
std::vector<webrtc::CodecInst> recv_codecs; |
webrtc::CodecInst send_codec; |
webrtc::PacketTime last_rtp_packet_time; |
@@ -281,9 +269,6 @@ class FakeWebRtcVoiceEngine |
channels_[channel]->cn16_type : |
channels_[channel]->cn8_type; |
} |
- int GetSendTelephoneEventPayloadType(int channel) { |
- return channels_[channel]->dtmf_type; |
- } |
int GetSendREDPayloadType(int channel) { |
return channels_[channel]->red_type; |
} |
@@ -552,26 +537,6 @@ class FakeWebRtcVoiceEngine |
return 0; |
} |
- // webrtc::VoEDtmf |
- WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code, |
- bool out_of_band = true, int length_ms = 160, int attenuation_db = 10)) { |
- channels_[channel]->dtmf_info.dtmf_event_code = event_code; |
- channels_[channel]->dtmf_info.dtmf_out_of_band = out_of_band; |
- channels_[channel]->dtmf_info.dtmf_length_ms = length_ms; |
- return 0; |
- } |
- WEBRTC_FUNC(SetSendTelephoneEventPayloadType, |
- (int channel, unsigned char type)) { |
- channels_[channel]->dtmf_type = type; |
- return 0; |
- }; |
- WEBRTC_STUB(GetSendTelephoneEventPayloadType, |
- (int channel, unsigned char& type)); |
- WEBRTC_STUB(SetDtmfFeedbackStatus, (bool enable, bool directFeedback)); |
- WEBRTC_STUB(GetDtmfFeedbackStatus, (bool& enabled, bool& directFeedback)); |
- WEBRTC_STUB(PlayDtmfTone, |
- (int event_code, int length_ms = 200, int attenuation_db = 10)); |
- |
// webrtc::VoEHardware |
WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) { |
return GetNumDevices(num); |
@@ -831,15 +796,6 @@ class FakeWebRtcVoiceEngine |
void EnableStereoChannelSwapping(bool enable) { |
stereo_swapping_enabled_ = enable; |
} |
- bool WasSendTelephoneEventCalled(int channel, int event_code, int length_ms) { |
- return (channels_[channel]->dtmf_info.dtmf_event_code == event_code && |
- channels_[channel]->dtmf_info.dtmf_out_of_band == true && |
- channels_[channel]->dtmf_info.dtmf_length_ms == length_ms); |
- } |
- bool WasPlayDtmfToneCalled(int event_code, int length_ms) { |
- return (dtmf_info_.dtmf_event_code == event_code && |
- dtmf_info_.dtmf_length_ms == length_ms); |
- } |
int GetNetEqCapacity() const { |
auto ch = channels_.find(last_channel_); |
ASSERT(ch != channels_.end()); |
@@ -910,7 +866,6 @@ class FakeWebRtcVoiceEngine |
int send_fail_channel_; |
int recording_sample_rate_; |
int playout_sample_rate_; |
- DtmfInfo dtmf_info_; |
FakeAudioProcessing audio_processing_; |
}; |