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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
View File

@ -375,7 +375,7 @@ class WebRtcSoundclipMedia : public SoundclipMedia {
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 @@ class WebRtcSoundclipMedia : public SoundclipMedia {
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 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
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.

80
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -337,56 +337,58 @@ class WebRtcVoiceMediaChannel
: 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);

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

@ -87,16 +87,12 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
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 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
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();