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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
commit43a97878ce14b72f0981164f87f2e35e14151312 (patch)
tree620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h
parentInitial commit. (diff)
downloadfirefox-upstream.tar.xz
firefox-upstream.zip
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h')
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1 files changed, 74 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h b/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h
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+/*
+ * Copyright (c) 2019 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 MODULES_PACING_RTP_PACKET_PACER_H_
+#define MODULES_PACING_RTP_PACKET_PACER_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+
+namespace webrtc {
+
+class RtpPacketPacer {
+ public:
+ virtual ~RtpPacketPacer() = default;
+
+ virtual void CreateProbeClusters(
+ std::vector<ProbeClusterConfig> probe_cluster_configs) = 0;
+
+ // Temporarily pause all sending.
+ virtual void Pause() = 0;
+
+ // Resume sending packets.
+ virtual void Resume() = 0;
+
+ virtual void SetCongested(bool congested) = 0;
+
+ // Sets the pacing rates. Must be called once before packets can be sent.
+ virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0;
+
+ // Time since the oldest packet currently in the queue was added.
+ virtual TimeDelta OldestPacketWaitTime() const = 0;
+
+ // Sum of payload + padding bytes of all packets currently in the pacer queue.
+ virtual DataSize QueueSizeData() const = 0;
+
+ // Returns the time when the first packet was sent.
+ virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
+
+ // Returns the expected number of milliseconds it will take to send the
+ // current packets in the queue, given the current size and bitrate, ignoring
+ // priority.
+ virtual TimeDelta ExpectedQueueTime() const = 0;
+
+ // Set the average upper bound on pacer queuing delay. The pacer may send at
+ // a higher rate than what was configured via SetPacingRates() in order to
+ // keep ExpectedQueueTimeMs() below `limit_ms` on average.
+ virtual void SetQueueTimeLimit(TimeDelta limit) = 0;
+
+ // Currently audio traffic is not accounted by pacer and passed through.
+ // With the introduction of audio BWE audio traffic will be accounted for
+ // the pacer budget calculation. The audio traffic still will be injected
+ // at high priority.
+ virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
+ virtual void SetIncludeOverhead() = 0;
+ virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
+};
+
+} // namespace webrtc
+#endif // MODULES_PACING_RTP_PACKET_PACER_H_