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Unified Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1491743004: Refactor WVoE DTMF handling #2 (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@wvoe_dtmf
Patch Set: rebase Created 5 years ago
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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_;
};
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