WebRtcVoiceEngine: virtual to override + git cl format.
BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54369004
Cr-Commit-Position: refs/heads/master@{#9154}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5bf54d4..b509d17 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -375,7 +375,7 @@
engine_->RegisterSoundclip(this);
}
- virtual ~WebRtcSoundclipMedia() {
+ ~WebRtcSoundclipMedia() override {
engine_->UnregisterSoundclip(this);
if (webrtc_channel_ != -1) {
// We shouldn't have to call Disable() here. DeleteChannel() should call
@@ -419,7 +419,7 @@
return true;
}
- virtual bool PlaySound(const char *buf, int len, int flags) {
+ bool PlaySound(const char* buf, int len, int flags) override {
// The voe file api is not available in chrome.
if (!engine_->voe_sc()->file()) {
return false;
@@ -1796,9 +1796,7 @@
voe_audio_transport_(voe_audio_transport),
renderer_(NULL) {
}
- virtual ~WebRtcVoiceChannelRenderer() {
- Stop();
- }
+ ~WebRtcVoiceChannelRenderer() override { Stop(); }
// Starts the rendering by setting a sink to the renderer to get data
// callback.
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 30f5717..f938806 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -337,56 +337,58 @@
: public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> {
public:
explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
- virtual ~WebRtcVoiceMediaChannel();
- virtual bool SetOptions(const AudioOptions& options);
- virtual bool GetOptions(AudioOptions* options) const {
+ ~WebRtcVoiceMediaChannel() override;
+ bool SetOptions(const AudioOptions& options) override;
+ bool GetOptions(AudioOptions* options) const override {
*options = options_;
return true;
}
- virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs);
- virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs);
- virtual bool SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
- virtual bool SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
- virtual bool SetPlayout(bool playout);
+ bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
+ bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override;
+ bool SetRecvRtpHeaderExtensions(
+ const std::vector<RtpHeaderExtension>& extensions) override;
+ bool SetSendRtpHeaderExtensions(
+ const std::vector<RtpHeaderExtension>& extensions) override;
+ bool SetPlayout(bool playout) override;
bool PausePlayout();
bool ResumePlayout();
- virtual bool SetSend(SendFlags send);
+ bool SetSend(SendFlags send) override;
bool PauseSend();
bool ResumeSend();
- virtual bool AddSendStream(const StreamParams& sp);
- virtual bool RemoveSendStream(uint32 ssrc);
- virtual bool AddRecvStream(const StreamParams& sp);
- virtual bool RemoveRecvStream(uint32 ssrc);
- virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
- virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
- virtual bool GetActiveStreams(AudioInfo::StreamList* actives);
- virtual int GetOutputLevel();
- virtual int GetTimeSinceLastTyping();
- virtual void SetTypingDetectionParameters(int time_window,
- int cost_per_typing, int reporting_threshold, int penalty_decay,
- int type_event_delay);
- virtual bool SetOutputScaling(uint32 ssrc, double left, double right);
- virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right);
+ bool AddSendStream(const StreamParams& sp) override;
+ bool RemoveSendStream(uint32 ssrc) override;
+ bool AddRecvStream(const StreamParams& sp) override;
+ bool RemoveRecvStream(uint32 ssrc) override;
+ bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
+ bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
+ bool GetActiveStreams(AudioInfo::StreamList* actives) override;
+ int GetOutputLevel() override;
+ int GetTimeSinceLastTyping() override;
+ void SetTypingDetectionParameters(int time_window,
+ int cost_per_typing,
+ int reporting_threshold,
+ int penalty_decay,
+ int type_event_delay) override;
+ bool SetOutputScaling(uint32 ssrc, double left, double right) override;
+ bool GetOutputScaling(uint32 ssrc, double* left, double* right) override;
- virtual bool SetRingbackTone(const char *buf, int len);
- virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
- virtual bool CanInsertDtmf();
- virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags);
+ bool SetRingbackTone(const char* buf, int len) override;
+ bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) override;
+ bool CanInsertDtmf() override;
+ bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
- virtual void OnPacketReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time);
- virtual void OnRtcpReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time);
- virtual void OnReadyToSend(bool ready) {}
- virtual bool MuteStream(uint32 ssrc, bool on);
- virtual bool SetMaxSendBandwidth(int bps);
- virtual bool GetStats(VoiceMediaInfo* info);
+ void OnPacketReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time) override;
+ void OnRtcpReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time) override;
+ void OnReadyToSend(bool ready) override {}
+ bool MuteStream(uint32 ssrc, bool on) override;
+ bool SetMaxSendBandwidth(int bps) override;
+ bool GetStats(VoiceMediaInfo* info) override;
// Gets last reported error from WebRtc voice engine. This should be only
// called in response a failure.
- virtual void GetLastMediaError(uint32* ssrc,
- VoiceMediaChannel::Error* error);
+ void GetLastMediaError(uint32* ssrc,
+ VoiceMediaChannel::Error* error) override;
bool FindSsrc(int channel_num, uint32* ssrc);
void OnError(uint32 ssrc, int error);
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 0ea260c..93d8c51 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -87,16 +87,12 @@
class FakeVoETraceWrapper : public cricket::VoETraceWrapper {
public:
- virtual int SetTraceFilter(const unsigned int filter) {
+ int SetTraceFilter(const unsigned int filter) override {
filter_ = filter;
return 0;
}
- virtual int SetTraceFile(const char* fileNameUTF8) {
- return 0;
- }
- virtual int SetTraceCallback(webrtc::TraceCallback* callback) {
- return 0;
- }
+ int SetTraceFile(const char* fileNameUTF8) override { return 0; }
+ int SetTraceCallback(webrtc::TraceCallback* callback) override { return 0; }
unsigned int filter_;
};
@@ -172,7 +168,7 @@
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
channel_->OnPacketReceived(&packet, rtc::PacketTime());
}
- virtual void TearDown() {
+ void TearDown() override {
delete soundclip_;
delete channel_;
engine_.Terminate();
