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}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index bc82751..5163d16 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -67,6 +67,7 @@
]
deps = [
"../..:webrtc_common",
+ "../../rtc_base:deprecation",
]
}
diff --git a/modules/audio_coding/acm2/call_statistics.h b/modules/audio_coding/acm2/call_statistics.h
index 9dced64..5d94ac4 100644
--- a/modules/audio_coding/acm2/call_statistics.h
+++ b/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
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index cd4351b..8946afd 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/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 @@
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_