Move NetworkStatistics and AudioDecodingCallStats from common_types.h

Bug: webrtc:7626
Change-Id: I1b933b8be7acbca1f1043a374a7cafb95aa9ffde
Reviewed-on: https://webrtc-review.googlesource.com/c/111249
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25688}

modules/audio_coding/BUILD.gn

@@ -67,6 +67,7 @@ rtc_source_set("audio_coding_module_typedefs") {
   ]
   deps = [
     "../..:webrtc_common",
+    "../../rtc_base:deprecation",
   ]
 }
 

modules/audio_coding/acm2/call_statistics.h

@@ -12,7 +12,7 @@
 #define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 
 #include "api/audio/audio_frame.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API

modules/audio_coding/include/audio_coding_module_typedefs.h

@@ -13,6 +13,8 @@
 
 #include <map>
 
+#include "rtc_base/deprecation.h"
+
 namespace webrtc {
 
 ///////////////////////////////////////////////////////////////////////////
@@ -43,6 +45,80 @@ enum OpusApplicationMode {
   kAudio = 1,
 };
 
+// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
+struct AudioDecodingCallStats {
+  AudioDecodingCallStats()
+      : calls_to_silence_generator(0),
+        calls_to_neteq(0),
+        decoded_normal(0),
+        decoded_plc(0),
+        decoded_cng(0),
+        decoded_plc_cng(0),
+        decoded_muted_output(0) {}
+
+  int calls_to_silence_generator;  // Number of calls where silence generated,
+                                   // and NetEq was disengaged from decoding.
+  int calls_to_neteq;              // Number of calls to NetEq.
+  int decoded_normal;  // Number of calls where audio RTP packet decoded.
+  int decoded_plc;     // Number of calls resulted in PLC.
+  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
+  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
+  int decoded_muted_output;  // Number of calls returning a muted state output.
+};
+
+// NETEQ statistics.
+struct NetworkStatistics {
+  // current jitter buffer size in ms
+  uint16_t currentBufferSize;
+  // preferred (optimal) buffer size in ms
+  uint16_t preferredBufferSize;
+  // adding extra delay due to "peaky jitter"
+  bool jitterPeaksFound;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+  uint64_t totalSamplesReceived;
+  uint64_t concealedSamples;
+  uint64_t concealmentEvents;
+  uint64_t jitterBufferDelayMs;
+  // Stats below DO NOT correspond directly to anything in the WebRTC stats
+  // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
+  uint16_t currentPacketLossRate;
+  // Late loss rate; fraction between 0 and 1, scaled to Q14.
+  union {
+    RTC_DEPRECATED uint16_t currentDiscardRate;
+  };
+  // fraction (of original stream) of synthesized audio inserted through
+  // expansion (in Q14)
+  uint16_t currentExpandRate;
+  // fraction (of original stream) of synthesized speech inserted through
+  // expansion (in Q14)
+  uint16_t currentSpeechExpandRate;
+  // fraction of synthesized speech inserted through pre-emptive expansion
+  // (in Q14)
+  uint16_t currentPreemptiveRate;
+  // fraction of data removed through acceleration (in Q14)
+  uint16_t currentAccelerateRate;
+  // fraction of data coming from secondary decoding (in Q14)
+  uint16_t currentSecondaryDecodedRate;
+  // Fraction of secondary data, including FEC and RED, that is discarded (in
+  // Q14). Discarding of secondary data can be caused by the reception of the
+  // primary data, obsoleting the secondary data. It can also be caused by early
+  // or late arrival of secondary data.
+  uint16_t currentSecondaryDiscardedRate;
+  // clock-drift in parts-per-million (negative or positive)
+  int32_t clockDriftPPM;
+  // average packet waiting time in the jitter buffer (ms)
+  int meanWaitingTimeMs;
+  // median packet waiting time in the jitter buffer (ms)
+  int medianWaitingTimeMs;
+  // min packet waiting time in the jitter buffer (ms)
+  int minWaitingTimeMs;
+  // max packet waiting time in the jitter buffer (ms)
+  int maxWaitingTimeMs;
+  // added samples in off mode due to packet loss
+  size_t addedSamples;
+};
+
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_