mirror of
https://github.com/mollyim/webrtc.git
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6634c91194
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023}
107 lines
4.2 KiB
C++
107 lines
4.2 KiB
C++
/*
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* Copyright (c) 2019 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_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
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#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
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#include "api/array_view.h"
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#include "api/rtp_headers.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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//
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// Helper class for sending the `AbsoluteCaptureTime` header extension.
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//
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// Supports the "timestamp interpolation" optimization:
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// A sender SHOULD save bandwidth by not sending abs-capture-time with every
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// RTP packet. It SHOULD still send them at regular intervals (e.g. every
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// second) to help mitigate the impact of clock drift and packet loss. Mixers
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// SHOULD always send abs-capture-time with the first RTP packet after
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// changing capture system.
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//
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// Timestamp interpolation works fine as long as there’s reasonably low
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// NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
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// between its NTP and RTP clock mappings SHOULD send abs-capture-time with
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// the first RTP packet after such a thing happening.
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//
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// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
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//
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class AbsoluteCaptureTimeSender {
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public:
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static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(1);
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static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
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explicit AbsoluteCaptureTimeSender(Clock* clock);
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// Returns the source (i.e. SSRC or CSRC) of the capture system.
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static uint32_t GetSource(uint32_t ssrc,
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rtc::ArrayView<const uint32_t> csrcs);
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// Returns value to write into AbsoluteCaptureTime RTP header extension to be
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// sent, or `absl::nullopt` if the header extension shouldn't be attached to
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// the outgoing packet.
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//
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// - `source` - id of the capture system.
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// - `rtp_timestamp` - capture time represented as rtp timestamp in the
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// outgoing packet
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// - `rtp_clock_frequency_hz` - description of the `rtp_timestamp` units -
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// `rtp_timetamp` delta of `rtp_clock_freqnecy_hz` represents 1 second.
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// - `absolute_capture_time` - time when a frame was captured by the capture
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// system.
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// - `estimated_capture_clock_offset` - estimated offset between capture
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// system clock and local `clock` passed as the AbsoluteCaptureTimeSender
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// construction paramter. Uses the same units as `absolute_capture_time`,
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// i.e. delta of 2^32 represents 1 second. See AbsoluteCaptureTime type
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// comments for more details.
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// - `force` - when set to true, OnSendPacket is forced to return non-nullopt.
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absl::optional<AbsoluteCaptureTime> OnSendPacket(
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uint32_t source,
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz,
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NtpTime absolute_capture_time,
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absl::optional<int64_t> estimated_capture_clock_offset,
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bool force = false);
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// Returns a header extension to be sent, or `absl::nullopt` if the header
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// extension shouldn't be sent.
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[[deprecated]] absl::optional<AbsoluteCaptureTime> OnSendPacket(
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uint32_t source,
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uint32_t rtp_timestamp,
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uint32_t rtp_clock_frequency,
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uint64_t absolute_capture_timestamp,
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absl::optional<int64_t> estimated_capture_clock_offset);
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private:
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bool ShouldSendExtension(
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Timestamp send_time,
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uint32_t source,
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz,
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NtpTime absolute_capture_time,
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absl::optional<int64_t> estimated_capture_clock_offset) const;
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Clock* const clock_;
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Timestamp last_send_time_ = Timestamp::MinusInfinity();
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uint32_t last_source_;
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uint32_t last_rtp_timestamp_;
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int last_rtp_clock_frequency_hz_;
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NtpTime last_absolute_capture_time_;
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absl::optional<int64_t> last_estimated_capture_clock_offset_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
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