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This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets. The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter. The usage and parameters can be controlled via field trial. Bug: webrtc:10178 Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34954}
61 lines
1.8 KiB
C++
61 lines
1.8 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_RELATIVE_ARRIVAL_DELAY_TRACKER_H_
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#define MODULES_AUDIO_CODING_NETEQ_RELATIVE_ARRIVAL_DELAY_TRACKER_H_
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#include <deque>
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#include <memory>
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#include "absl/types/optional.h"
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#include "api/neteq/tick_timer.h"
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namespace webrtc {
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class RelativeArrivalDelayTracker {
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public:
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RelativeArrivalDelayTracker(const TickTimer* tick_timer, int max_history_ms)
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: tick_timer_(tick_timer), max_history_ms_(max_history_ms) {}
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absl::optional<int> Update(uint32_t timestamp, int sample_rate_hz);
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void Reset();
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absl::optional<uint32_t> newest_timestamp() const {
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return newest_timestamp_;
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}
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private:
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// Updates `delay_history_`.
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void UpdateDelayHistory(int iat_delay_ms,
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uint32_t timestamp,
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int sample_rate_hz);
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// Calculate relative packet arrival delay from `delay_history_`.
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int CalculateRelativePacketArrivalDelay() const;
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const TickTimer* tick_timer_;
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const int max_history_ms_;
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struct PacketDelay {
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int iat_delay_ms;
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uint32_t timestamp;
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};
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std::deque<PacketDelay> delay_history_;
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absl::optional<uint32_t> newest_timestamp_;
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absl::optional<uint32_t> last_timestamp_;
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std::unique_ptr<TickTimer::Stopwatch>
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packet_iat_stopwatch_; // Time elapsed since last packet.
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_RELATIVE_ARRIVAL_DELAY_TRACKER_H_
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