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56e611bbda
This is a reland of 4f68f5398d
Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}
TBR=stefan@webrtc.org
Bug: webrtc:11340
Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30479}
106 lines
3.1 KiB
C++
106 lines
3.1 KiB
C++
/*
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* Copyright (c) 2012 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 COMMON_TYPES_H_
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#define COMMON_TYPES_H_
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#include <stddef.h> // For size_t
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#include <cstdint>
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namespace webrtc {
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struct FrameCounts {
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FrameCounts() : key_frames(0), delta_frames(0) {}
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int key_frames;
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int delta_frames;
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};
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// Callback, used to notify an observer whenever frame counts have been updated.
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class FrameCountObserver {
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public:
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virtual ~FrameCountObserver() {}
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virtual void FrameCountUpdated(const FrameCounts& frame_counts,
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uint32_t ssrc) = 0;
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};
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// Callback, used to notify an observer when the overhead per packet
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// has changed.
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class OverheadObserver {
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public:
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virtual ~OverheadObserver() = default;
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virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0;
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};
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// ==================================================================
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// Video specific types
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// ==================================================================
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// TODO(magjed): Move this and other H264 related classes out to their own file.
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namespace H264 {
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enum Profile {
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kProfileConstrainedBaseline,
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kProfileBaseline,
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kProfileMain,
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kProfileConstrainedHigh,
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kProfileHigh,
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};
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} // namespace H264
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struct SpatialLayer {
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bool operator==(const SpatialLayer& other) const;
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bool operator!=(const SpatialLayer& other) const { return !(*this == other); }
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unsigned short width;
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unsigned short height;
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float maxFramerate; // fps.
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unsigned char numberOfTemporalLayers;
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unsigned int maxBitrate; // kilobits/sec.
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unsigned int targetBitrate; // kilobits/sec.
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unsigned int minBitrate; // kilobits/sec.
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unsigned int qpMax; // minimum quality
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bool active; // encoded and sent.
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};
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// Simulcast is when the same stream is encoded multiple times with different
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// settings such as resolution.
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typedef SpatialLayer SimulcastStream;
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// Minimum and maximum playout delay values from capture to render.
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// These are best effort values.
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//
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// A value < 0 indicates no change from previous valid value.
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//
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// min = max = 0 indicates that the receiver should try and render
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// frame as soon as possible.
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//
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// min = x, max = y indicates that the receiver is free to adapt
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// in the range (x, y) based on network jitter.
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//
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// Note: Given that this gets embedded in a union, it is up-to the owner to
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// initialize these values.
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struct PlayoutDelay {
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PlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
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int min_ms;
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int max_ms;
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static PlayoutDelay Noop() { return PlayoutDelay(-1, -1); }
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bool IsNoop() const { return min_ms == -1 && max_ms == -1; }
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bool operator==(const PlayoutDelay& rhs) const {
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return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
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}
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};
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} // namespace webrtc
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#endif // COMMON_TYPES_H_
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