| 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_;
|
| };
|
|
|
|
|