Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/
BUG=163 TBR=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1900004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
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@@ -33,10 +33,10 @@ class DelayManager {
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// object to the DelayManager.
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DelayManager(int max_packets_in_buffer, DelayPeakDetector* peak_detector);
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virtual ~DelayManager() {}
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virtual ~DelayManager();
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// Read the inter-arrival time histogram. Mainly for testing purposes.
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virtual const IATVector& iat_vector() const { return iat_vector_; }
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virtual const IATVector& iat_vector() const;
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// Updates the delay manager with a new incoming packet, with
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// |sequence_number| and |timestamp| from the RTP header. This updates the
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@@ -79,7 +79,7 @@ class DelayManager {
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virtual void UpdateCounters(int elapsed_time_ms);
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// Reset the inter-arrival time counter to 0.
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virtual void ResetPacketIatCount() { packet_iat_count_ms_ = 0; }
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virtual void ResetPacketIatCount();
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// Writes the lower and higher limits which the buffer level should stay
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// within to the corresponding pointers. The values are in (fractions of)
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@@ -93,13 +93,11 @@ class DelayManager {
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virtual void LastDecoderType(NetEqDecoder decoder_type);
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// Accessors and mutators.
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virtual void set_extra_delay_ms(int16_t delay) { extra_delay_ms_ = delay; }
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virtual int base_target_level() const { return base_target_level_; }
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virtual void set_streaming_mode(bool value) { streaming_mode_ = value; }
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virtual int last_pack_cng_or_dtmf() const { return last_pack_cng_or_dtmf_; }
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virtual void set_last_pack_cng_or_dtmf(int value) {
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last_pack_cng_or_dtmf_ = value;
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}
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virtual void set_extra_delay_ms(int16_t delay);
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virtual int base_target_level() const;
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virtual void set_streaming_mode(bool value);
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virtual int last_pack_cng_or_dtmf() const;
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virtual void set_last_pack_cng_or_dtmf(int value);
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private:
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static const int kLimitProbability = 53687091; // 1/20 in Q30.
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