Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 98 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/voip/voip_statistics.h b/third_party/libwebrtc/api/voip/voip_statistics.h
new file mode 100644
index 0000000000..2d1ab8d5e8
--- /dev/null
+++ b/third_party/libwebrtc/api/voip/voip_statistics.h
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VOIP_VOIP_STATISTICS_H_
+#define API_VOIP_VOIP_STATISTICS_H_
+
+#include "api/neteq/neteq.h"
+#include "api/voip/voip_base.h"
+
+namespace webrtc {
+
+struct IngressStatistics {
+  // Stats included from api/neteq/neteq.h.
+  NetEqLifetimeStatistics neteq_stats;
+
+  // Represents the total duration in seconds of all samples that have been
+  // received.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsamplesduration
+  double total_duration = 0.0;
+};
+
+// Remote statistics obtained via remote RTCP SR/RR report received.
+struct RemoteRtcpStatistics {
+  // Jitter as defined in RFC 3550 [6.4.1] expressed in seconds.
+  double jitter = 0.0;
+
+  // Cumulative packets lost as defined in RFC 3550 [6.4.1]
+  int64_t packets_lost = 0;
+
+  // Fraction lost as defined in RFC 3550 [6.4.1] expressed as a floating
+  // pointer number.
+  double fraction_lost = 0.0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats-roundtriptime
+  absl::optional<double> round_trip_time;
+
+  // Last time (not RTP timestamp) when RTCP report received in milliseconds.
+  int64_t last_report_received_timestamp_ms;
+};
+
+struct ChannelStatistics {
+  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-packetssent
+  uint64_t packets_sent = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent
+  uint64_t bytes_sent = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceived
+  uint64_t packets_received = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived
+  uint64_t bytes_received = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter
+  double jitter = 0.0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost
+  int64_t packets_lost = 0;
+
+  // SSRC from remote media endpoint as indicated either by RTP header in RFC
+  // 3550 [5.1] or RTCP SSRC of sender in RFC 3550 [6.4.1].
+  absl::optional<uint32_t> remote_ssrc;
+
+  absl::optional<RemoteRtcpStatistics> remote_rtcp;
+};
+
+// VoipStatistics interface provides the interfaces for querying metrics around
+// the jitter buffer (NetEq) performance.
+class VoipStatistics {
+ public:
+  // Gets the audio ingress statistics by `ingress_stats` reference.
+  // Returns following VoipResult;
+  //  kOk - successfully set provided IngressStatistics reference.
+  //  kInvalidArgument - `channel_id` is invalid.
+  virtual VoipResult GetIngressStatistics(ChannelId channel_id,
+                                          IngressStatistics& ingress_stats) = 0;
+
+  // Gets the channel statistics by `channel_stats` reference.
+  // Returns following VoipResult;
+  //  kOk - successfully set provided ChannelStatistics reference.
+  //  kInvalidArgument - `channel_id` is invalid.
+  virtual VoipResult GetChannelStatistics(ChannelId channel_id,
+                                          ChannelStatistics& channel_stats) = 0;
+
+ protected:
+  virtual ~VoipStatistics() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VOIP_VOIP_STATISTICS_H_