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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h | |
parent | Initial commit. (diff) | |
download | firefox-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')
-rw-r--r-- | third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h | 74 |
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 new file mode 100644 index 0000000000..e2cf806385 --- /dev/null +++ b/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h @@ -0,0 +1,74 @@ +/* + * 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_ |