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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h | 82 |
1 files changed, 82 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h new file mode 100644 index 0000000000..cad362b469 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h @@ -0,0 +1,82 @@ +/* + * Copyright (c) 2022 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_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_ +#define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_ + +#include <cstdint> +#include <deque> + +#include "absl/types/optional.h" +#include "api/neteq/tick_timer.h" +#include "rtc_base/numerics/sequence_number_unwrapper.h" + +namespace webrtc { + +// Stores timing information about previously received packets. +// The history has a fixed window size beyond which old data is automatically +// pruned. +class PacketArrivalHistory { + public: + explicit PacketArrivalHistory(int window_size_ms); + + // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history. + void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms); + + // The delay for `rtp_timestamp` at `time_ms` is calculated as + // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)` + // where `p` is chosen as the packet arrival in the history that maximizes the + // delay. + int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const; + + // Get the maximum packet arrival delay observed in the history. + int GetMaxDelayMs() const; + + bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const; + + void Reset(); + + void set_sample_rate(int sample_rate) { + sample_rate_khz_ = sample_rate / 1000; + } + + size_t size() const { return history_.size(); } + + private: + struct PacketArrival { + PacketArrival(int64_t rtp_timestamp_ms, int64_t arrival_time_ms) + : rtp_timestamp_ms(rtp_timestamp_ms), + arrival_time_ms(arrival_time_ms) {} + int64_t rtp_timestamp_ms; + int64_t arrival_time_ms; + bool operator<=(const PacketArrival& other) const { + return arrival_time_ms - rtp_timestamp_ms <= + other.arrival_time_ms - other.rtp_timestamp_ms; + } + bool operator>=(const PacketArrival& other) const { + return arrival_time_ms - rtp_timestamp_ms >= + other.arrival_time_ms - other.rtp_timestamp_ms; + } + }; + std::deque<PacketArrival> history_; + int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const; + // Updates `min_packet_arrival_` and `max_packet_arrival_`. + void MaybeUpdateCachedArrivals(const PacketArrival& packet); + const PacketArrival* min_packet_arrival_ = nullptr; + const PacketArrival* max_packet_arrival_ = nullptr; + const int window_size_ms_; + RtpTimestampUnwrapper timestamp_unwrapper_; + absl::optional<int64_t> newest_rtp_timestamp_; + int sample_rate_khz_ = 0; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_ |