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WebRtcVoiceEngine: virtual to override + git cl format.

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BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54369004

Cr-Commit-Position: refs/heads/master@{#9154}
This commit is contained in:
Fredrik Solenberg
2015-05-07 16:05:53 +02:00
parent 6179b89e53
commit aaf8ff2e45
3 changed files with 48 additions and 52 deletions
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8
talk/media/webrtc/webrtcvoiceengine.cc
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@@ -375,7 +375,7 @@ class WebRtcSoundclipMedia : public SoundclipMedia {
engine_->RegisterSoundclip(this); engine_->RegisterSoundclip(this);
} }
virtual ~WebRtcSoundclipMedia() { ~WebRtcSoundclipMedia() override {
engine_->UnregisterSoundclip(this); engine_->UnregisterSoundclip(this);
if (webrtc_channel_ != -1) { if (webrtc_channel_ != -1) {
// We shouldn't have to call Disable() here. DeleteChannel() should call // We shouldn't have to call Disable() here. DeleteChannel() should call
@@ -419,7 +419,7 @@ class WebRtcSoundclipMedia : public SoundclipMedia {
return true; 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. // The voe file api is not available in chrome.
if (!engine_->voe_sc()->file()) { if (!engine_->voe_sc()->file()) {
return false; return false;
@@ -1796,9 +1796,7 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
voe_audio_transport_(voe_audio_transport), voe_audio_transport_(voe_audio_transport),
renderer_(NULL) { renderer_(NULL) {
} }
virtual ~WebRtcVoiceChannelRenderer() { ~WebRtcVoiceChannelRenderer() override { Stop(); }
Stop();
}
// Starts the rendering by setting a sink to the renderer to get data // Starts the rendering by setting a sink to the renderer to get data
// callback. // callback.

80
talk/media/webrtc/webrtcvoiceengine.h
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@@ -337,56 +337,58 @@ class WebRtcVoiceMediaChannel
: public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> { : public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> {
public: public:
explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine); explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
virtual ~WebRtcVoiceMediaChannel(); ~WebRtcVoiceMediaChannel() override;
virtual bool SetOptions(const AudioOptions& options); bool SetOptions(const AudioOptions& options) override;
virtual bool GetOptions(AudioOptions* options) const { bool GetOptions(AudioOptions* options) const override {
*options = options_; *options = options_;
return true; return true;
} }
virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs); bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs); bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override;
virtual bool SetRecvRtpHeaderExtensions( bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions); const std::vector<RtpHeaderExtension>& extensions) override;
virtual bool SetSendRtpHeaderExtensions( bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions); const std::vector<RtpHeaderExtension>& extensions) override;
virtual bool SetPlayout(bool playout); bool SetPlayout(bool playout) override;
bool PausePlayout(); bool PausePlayout();
bool ResumePlayout(); bool ResumePlayout();
virtual bool SetSend(SendFlags send); bool SetSend(SendFlags send) override;
bool PauseSend(); bool PauseSend();
bool ResumeSend(); bool ResumeSend();
virtual bool AddSendStream(const StreamParams& sp); bool AddSendStream(const StreamParams& sp) override;
virtual bool RemoveSendStream(uint32 ssrc); bool RemoveSendStream(uint32 ssrc) override;
virtual bool AddRecvStream(const StreamParams& sp); bool AddRecvStream(const StreamParams& sp) override;
virtual bool RemoveRecvStream(uint32 ssrc); bool RemoveRecvStream(uint32 ssrc) override;
virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer); bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer); bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
virtual bool GetActiveStreams(AudioInfo::StreamList* actives); bool GetActiveStreams(AudioInfo::StreamList* actives) override;
virtual int GetOutputLevel(); int GetOutputLevel() override;
virtual int GetTimeSinceLastTyping(); int GetTimeSinceLastTyping() override;
virtual void SetTypingDetectionParameters(int time_window, void SetTypingDetectionParameters(int time_window,
int cost_per_typing, int reporting_threshold, int penalty_decay, int cost_per_typing,
int type_event_delay); int reporting_threshold,
virtual bool SetOutputScaling(uint32 ssrc, double left, double right); int penalty_decay,
virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right); 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); bool SetRingbackTone(const char* buf, int len) override;
virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop); bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) override;
virtual bool CanInsertDtmf(); bool CanInsertDtmf() override;
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags); bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
virtual void OnPacketReceived(rtc::Buffer* packet, void OnPacketReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time); const rtc::PacketTime& packet_time) override;
virtual void OnRtcpReceived(rtc::Buffer* packet, void OnRtcpReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time); const rtc::PacketTime& packet_time) override;
virtual void OnReadyToSend(bool ready) {} void OnReadyToSend(bool ready) override {}
virtual bool MuteStream(uint32 ssrc, bool on); bool MuteStream(uint32 ssrc, bool on) override;
virtual bool SetMaxSendBandwidth(int bps); bool SetMaxSendBandwidth(int bps) override;
virtual bool GetStats(VoiceMediaInfo* info); bool GetStats(VoiceMediaInfo* info) override;
// Gets last reported error from WebRtc voice engine. This should be only // Gets last reported error from WebRtc voice engine. This should be only
// called in response a failure. // called in response a failure.
virtual void GetLastMediaError(uint32* ssrc, void GetLastMediaError(uint32* ssrc,
VoiceMediaChannel::Error* error); VoiceMediaChannel::Error* error) override;
bool FindSsrc(int channel_num, uint32* ssrc); bool FindSsrc(int channel_num, uint32* ssrc);
void OnError(uint32 ssrc, int error); void OnError(uint32 ssrc, int error);

12
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@@ -87,16 +87,12 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
class FakeVoETraceWrapper : public cricket::VoETraceWrapper { class FakeVoETraceWrapper : public cricket::VoETraceWrapper {
public: public:
virtual int SetTraceFilter(const unsigned int filter) { int SetTraceFilter(const unsigned int filter) override {
filter_ = filter; filter_ = filter;
return 0; return 0;
} }
virtual int SetTraceFile(const char* fileNameUTF8) { int SetTraceFile(const char* fileNameUTF8) override { return 0; }
return 0; int SetTraceCallback(webrtc::TraceCallback* callback) override { return 0; }
}
virtual int SetTraceCallback(webrtc::TraceCallback* callback) {
return 0;
}
unsigned int filter_; unsigned int filter_;
}; };
@@ -172,7 +168,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len); rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
channel_->OnPacketReceived(&packet, rtc::PacketTime()); channel_->OnPacketReceived(&packet, rtc::PacketTime());
} }
virtual void TearDown() { void TearDown() override {
delete soundclip_; delete soundclip_;
delete channel_; delete channel_;
engine_.Terminate(); engine_.Terminate();