| Index: webrtc/audio_receive_stream.h
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| diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
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| index 70d6480b10c1cc3e9d140b93a90b614d1008413e..3e5a518a7d5efce9f9f0194a5dc98e40ec95d733 100644
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| --- a/webrtc/audio_receive_stream.h
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| +++ b/webrtc/audio_receive_stream.h
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| @@ -26,7 +26,32 @@ class AudioDecoder;
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|  
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|  class AudioReceiveStream : public ReceiveStream {
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|   public:
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| -  struct Stats {};
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| +  struct Stats {
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| +    uint32_t remote_ssrc = 0;
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| +    int64_t bytes_rcvd = 0;
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| +    uint32_t packets_rcvd = 0;
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| +    uint32_t packets_lost = 0;
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| +    float fraction_lost = 0.0f;
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| +    std::string codec_name;
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| +    uint32_t ext_seqnum = 0;
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| +    uint32_t jitter_ms = 0;
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| +    uint32_t jitter_buffer_ms = 0;
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| +    uint32_t jitter_buffer_preferred_ms = 0;
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| +    uint32_t delay_estimate_ms = 0;
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| +    int32_t audio_level = -1;
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| +    float expand_rate = 0.0f;
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| +    float speech_expand_rate = 0.0f;
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| +    float secondary_decoded_rate = 0.0f;
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| +    float accelerate_rate = 0.0f;
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| +    float preemptive_expand_rate = 0.0f;
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| +    int32_t decoding_calls_to_silence_generator = 0;
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| +    int32_t decoding_calls_to_neteq = 0;
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| +    int32_t decoding_normal = 0;
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| +    int32_t decoding_plc = 0;
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| +    int32_t decoding_cng = 0;
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| +    int32_t decoding_plc_cng = 0;
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| +    int64_t capture_start_ntp_time_ms = 0;
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| +  };
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|  
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|    struct Config {
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|      std::string ToString() const;
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| 
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