Implement RTCMediaSourceStats and friends in standard getStats().
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond
This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.
Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
If the same track is attached multiple times this results in
multiple media-source objects, but the spec says it should be on a
per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
sender with SSRC), but if collected on a per-source basis the source
should be able to tell us the FPS whether or not we are sending it.
Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h
index bb23490..4361163 100644
--- a/api/stats/rtc_stats.h
+++ b/api/stats/rtc_stats.h
@@ -179,6 +179,24 @@
return local_var_members_vec; \
}
+// A version of WEBRTC_RTCSTATS_IMPL() where "..." is omitted, used to avoid a
+// compile error on windows. This is used if the stats dictionary does not
+// declare any members of its own (but perhaps its parent dictionary does).
+#define WEBRTC_RTCSTATS_IMPL_NO_MEMBERS(this_class, parent_class, type_str) \
+ const char this_class::kType[] = type_str; \
+ \
+ std::unique_ptr<webrtc::RTCStats> this_class::copy() const { \
+ return std::unique_ptr<webrtc::RTCStats>(new this_class(*this)); \
+ } \
+ \
+ const char* this_class::type() const { return this_class::kType; } \
+ \
+ std::vector<const webrtc::RTCStatsMemberInterface*> \
+ this_class::MembersOfThisObjectAndAncestors( \
+ size_t local_var_additional_capacity) const { \
+ return parent_class::MembersOfThisObjectAndAncestors(0); \
+ }
+
// Non-standard stats members can be exposed to the JavaScript API in Chrome
// e.g. through origin trials. The group ID can be used by the blink layer to
// determine if a stats member should be exposed or not. Multiple non-standard
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 2e6cd3b..cbcc8fa 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -279,6 +279,7 @@
~RTCMediaStreamTrackStats() override;
RTCStatsMember<std::string> track_identifier;
+ RTCStatsMember<std::string> media_source_id;
RTCStatsMember<bool> remote_source;
RTCStatsMember<bool> ended;
// TODO(hbos): |RTCStatsCollector| does not return stats for detached tracks.
@@ -450,6 +451,7 @@
RTCOutboundRTPStreamStats(const RTCOutboundRTPStreamStats& other);
~RTCOutboundRTPStreamStats() override;
+ RTCStatsMember<std::string> media_source_id;
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
@@ -466,6 +468,50 @@
RTCStatsMember<std::string> content_type;
};
+// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
+class RTC_EXPORT RTCMediaSourceStats : public RTCStats {
+ public:
+ WEBRTC_RTCSTATS_DECL();
+
+ RTCMediaSourceStats(const RTCMediaSourceStats& other);
+ ~RTCMediaSourceStats() override;
+
+ RTCStatsMember<std::string> track_identifier;
+ RTCStatsMember<std::string> kind;
+
+ protected:
+ RTCMediaSourceStats(const std::string& id, int64_t timestamp_us);
+ RTCMediaSourceStats(std::string&& id, int64_t timestamp_us);
+};
+
+// https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats
+class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
+ public:
+ WEBRTC_RTCSTATS_DECL();
+
+ RTCAudioSourceStats(const std::string& id, int64_t timestamp_us);
+ RTCAudioSourceStats(std::string&& id, int64_t timestamp_us);
+ RTCAudioSourceStats(const RTCAudioSourceStats& other);
+ ~RTCAudioSourceStats() override;
+};
+
+// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
+class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats {
+ public:
+ WEBRTC_RTCSTATS_DECL();
+
+ RTCVideoSourceStats(const std::string& id, int64_t timestamp_us);
+ RTCVideoSourceStats(std::string&& id, int64_t timestamp_us);
+ RTCVideoSourceStats(const RTCVideoSourceStats& other);
+ ~RTCVideoSourceStats() override;
+
+ RTCStatsMember<uint32_t> width;
+ RTCStatsMember<uint32_t> height;
+ // TODO(hbos): Implement this metric.
+ RTCStatsMember<uint32_t> frames;
+ RTCStatsMember<uint32_t> frames_per_second;
+};
+
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
class RTC_EXPORT RTCTransportStats final : public RTCStats {
public: