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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
stefan@webrtc.org29794612012-02-08 08:58:55 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_VIDEO_CODING_JITTER_BUFFER_H_
12#define MODULES_VIDEO_CODING_JITTER_BUFFER_H_
niklase@google.com470e71d2011-07-07 08:21:25 +000013
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000014#include <list>
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000015#include <map>
kwiberg3f55dea2016-02-29 05:51:59 -080016#include <memory>
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000017#include <set>
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +000018#include <vector>
stefan@webrtc.org29794612012-02-08 08:58:55 +000019
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020020#include "modules/include/module_common_types.h"
21#include "modules/utility/include/process_thread.h"
22#include "modules/video_coding/decoding_state.h"
23#include "modules/video_coding/include/video_coding.h"
24#include "modules/video_coding/include/video_coding_defines.h"
25#include "modules/video_coding/inter_frame_delay.h"
26#include "modules/video_coding/jitter_buffer_common.h"
27#include "modules/video_coding/jitter_estimator.h"
Steve Anton10542f22019-01-11 09:11:00 -080028#include "rtc_base/constructor_magic.h"
29#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#include "rtc_base/thread_annotations.h"
Niels Möllerebad1772018-12-04 13:15:37 +010031#include "system_wrappers/include/event_wrapper.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000032
stefan@webrtc.org912981f2012-10-12 07:04:52 +000033namespace webrtc {
niklase@google.com470e71d2011-07-07 08:21:25 +000034
niklase@google.com470e71d2011-07-07 08:21:25 +000035// forward declarations
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000036class Clock;
niklase@google.com470e71d2011-07-07 08:21:25 +000037class VCMFrameBuffer;
38class VCMPacket;
39class VCMEncodedFrame;
40
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000041typedef std::list<VCMFrameBuffer*> UnorderedFrameList;
42
stefan@webrtc.org912981f2012-10-12 07:04:52 +000043struct VCMJitterSample {
44 VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
45 uint32_t timestamp;
46 uint32_t frame_size;
47 int64_t latest_packet_time;
niklase@google.com470e71d2011-07-07 08:21:25 +000048};
49
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000050class TimestampLessThan {
51 public:
philipel9d3ab612015-12-21 04:12:39 -080052 bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000053 return IsNewerTimestamp(timestamp2, timestamp1);
54 }
55};
56
agalusza@google.comd818dcb2013-07-29 21:48:11 +000057class FrameList
58 : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000059 public:
60 void InsertFrame(VCMFrameBuffer* frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000061 VCMFrameBuffer* PopFrame(uint32_t timestamp);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000062 VCMFrameBuffer* Front() const;
63 VCMFrameBuffer* Back() const;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000064 int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
philipel9d3ab612015-12-21 04:12:39 -080065 UnorderedFrameList* free_frames);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000066 void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
67 UnorderedFrameList* free_frames);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000068 void Reset(UnorderedFrameList* free_frames);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000069};
70
stefan@webrtc.org912981f2012-10-12 07:04:52 +000071class VCMJitterBuffer {
72 public:
Niels Möllerdb64d992019-03-29 14:30:53 +010073 VCMJitterBuffer(Clock* clock, std::unique_ptr<EventWrapper> event);
Qiang Chend4cec152015-06-19 09:17:00 -070074
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -070075 ~VCMJitterBuffer();
niklase@google.com470e71d2011-07-07 08:21:25 +000076
stefan@webrtc.org912981f2012-10-12 07:04:52 +000077 // Initializes and starts jitter buffer.
78 void Start();
niklase@google.com470e71d2011-07-07 08:21:25 +000079
stefan@webrtc.org912981f2012-10-12 07:04:52 +000080 // Signals all internal events and stops the jitter buffer.
81 void Stop();
niklase@google.com470e71d2011-07-07 08:21:25 +000082
stefan@webrtc.org912981f2012-10-12 07:04:52 +000083 // Returns true if the jitter buffer is running.
84 bool Running() const;
niklase@google.com470e71d2011-07-07 08:21:25 +000085
stefan@webrtc.org912981f2012-10-12 07:04:52 +000086 // Empty the jitter buffer of all its data.
87 void Flush();
stefan@webrtc.org791eec72011-10-11 07:53:43 +000088
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +000089 // Gets number of packets received.
90 int num_packets() const;
91
92 // Gets number of duplicated packets received.
93 int num_duplicated_packets() const;
94
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000095 // Wait |max_wait_time_ms| for a complete frame to arrive.
isheriff6b4b5f32016-06-08 00:24:21 -070096 // If found, a pointer to the frame is returned. Returns nullptr otherwise.
97 VCMEncodedFrame* NextCompleteFrame(uint32_t max_wait_time_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +000098
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000099 // Extract frame corresponding to input timestamp.
100 // Frame will be set to a decoding state.
101 VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000102
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000103 // Releases a frame returned from the jitter buffer, should be called when
104 // done with decoding.
105 void ReleaseFrame(VCMEncodedFrame* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000106
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000107 // Returns the time in ms when the latest packet was inserted into the frame.
108 // Retransmitted is set to true if any of the packets belonging to the frame
109 // has been retransmitted.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000110 int64_t LastPacketTime(const VCMEncodedFrame* frame,
111 bool* retransmitted) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000112
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000113 // Inserts a packet into a frame returned from GetFrame().
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000114 // If the return value is <= 0, |frame| is invalidated and the pointer must
115 // be dropped after this function returns.
philipel9d3ab612015-12-21 04:12:39 -0800116 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
niklase@google.com470e71d2011-07-07 08:21:25 +0000117
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000118 // Returns the estimated jitter in milliseconds.
119 uint32_t EstimatedJitterMs();
niklase@google.com470e71d2011-07-07 08:21:25 +0000120
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000121 // Updates the round-trip time estimate.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000122 void UpdateRtt(int64_t rtt_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000123
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000124 void SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000125 int max_packet_age_to_nack,
126 int max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000127
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000128 // Returns a list of the sequence numbers currently missing.
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700129 std::vector<uint16_t> GetNackList(bool* request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000130
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000131 private:
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000132 class SequenceNumberLessThan {
133 public:
philipel9d3ab612015-12-21 04:12:39 -0800134 bool operator()(const uint16_t& sequence_number1,
135 const uint16_t& sequence_number2) const {
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +0000136 return IsNewerSequenceNumber(sequence_number2, sequence_number1);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000137 }
138 };
139 typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
140
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000141 // Gets the frame assigned to the timestamp of the packet. May recycle
142 // existing frames if no free frames are available. Returns an error code if
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000143 // failing, or kNoError on success. |frame_list| contains which list the
144 // packet was in, or NULL if it was not in a FrameList (a new frame).
145 VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
146 VCMFrameBuffer** frame,
147 FrameList** frame_list)
danilchap56359be2017-09-07 07:53:45 -0700148 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000149
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000150 // Returns true if |frame| is continuous in |decoding_state|, not taking
151 // decodable frames into account.
152 bool IsContinuousInState(const VCMFrameBuffer& frame,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000153 const VCMDecodingState& decoding_state) const
danilchap56359be2017-09-07 07:53:45 -0700154 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000155 // Returns true if |frame| is continuous in the |last_decoded_state_|, taking
156 // all decodable frames into account.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000157 bool IsContinuous(const VCMFrameBuffer& frame) const
danilchap56359be2017-09-07 07:53:45 -0700158 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700159 // Looks for frames in |incomplete_frames_| which are continuous in the
160 // provided |decoded_state|. Starts the search from the timestamp of
161 // |decoded_state|.
162 void FindAndInsertContinuousFramesWithState(
163 const VCMDecodingState& decoded_state)
danilchap56359be2017-09-07 07:53:45 -0700164 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000165 // Looks for frames in |incomplete_frames_| which are continuous in
166 // |last_decoded_state_| taking all decodable frames into account. Starts
167 // the search from |new_frame|.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000168 void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
danilchap56359be2017-09-07 07:53:45 -0700169 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
170 VCMFrameBuffer* NextFrame() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000171 // Returns true if the NACK list was updated to cover sequence numbers up to
172 // |sequence_number|. If false a key frame is needed to get into a state where
173 // we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000174 bool UpdateNackList(uint16_t sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700175 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000176 bool TooLargeNackList() const;
177 // Returns true if the NACK list was reduced without problem. If false a key
178 // frame is needed to get into a state where we can continue decoding.
danilchap56359be2017-09-07 07:53:45 -0700179 bool HandleTooLargeNackList() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000180 bool MissingTooOldPacket(uint16_t latest_sequence_number) const
danilchap56359be2017-09-07 07:53:45 -0700181 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000182 // Returns true if the too old packets was successfully removed from the NACK
183 // list. If false, a key frame is needed to get into a state where we can
184 // continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000185 bool HandleTooOldPackets(uint16_t latest_sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700186 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000187 // Drops all packets in the NACK list up until |last_decoded_sequence_number|.
188 void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
189
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000190 // Gets an empty frame, creating a new frame if necessary (i.e. increases
191 // jitter buffer size).
danilchap56359be2017-09-07 07:53:45 -0700192 VCMFrameBuffer* GetEmptyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000193
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000194 // Attempts to increase the size of the jitter buffer. Returns true on
195 // success, false otherwise.
danilchap56359be2017-09-07 07:53:45 -0700196 bool TryToIncreaseJitterBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000197
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000198 // Recycles oldest frames until a key frame is found. Used if jitter buffer is
199 // completely full. Returns true if a key frame was found.
danilchap56359be2017-09-07 07:53:45 -0700200 bool RecycleFramesUntilKeyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000201
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000202 // Update rolling average of packets per frame.
203 void UpdateAveragePacketsPerFrame(int current_number_packets_);
204
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000205 // Cleans the frame list in the JB from old/empty frames.
206 // Should only be called prior to actual use.
danilchap56359be2017-09-07 07:53:45 -0700207 void CleanUpOldOrEmptyFrames() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000208
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000209 // Returns true if |packet| is likely to have been retransmitted.
210 bool IsPacketRetransmitted(const VCMPacket& packet) const;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +0000211
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000212 // The following three functions update the jitter estimate with the
213 // payload size, receive time and RTP timestamp of a frame.
214 void UpdateJitterEstimate(const VCMJitterSample& sample,
215 bool incomplete_frame);
216 void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
217 void UpdateJitterEstimate(int64_t latest_packet_time_ms,
218 uint32_t timestamp,
219 unsigned int frame_size,
220 bool incomplete_frame);
221
danilchap56359be2017-09-07 07:53:45 -0700222 int NonContinuousOrIncompleteDuration()
223 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000224
225 uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
226
sprang22691e02016-07-13 10:57:07 -0700227 // Reset frame buffer and return it to free_frames_.
228 void RecycleFrameBuffer(VCMFrameBuffer* frame)
danilchap56359be2017-09-07 07:53:45 -0700229 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
sprang22691e02016-07-13 10:57:07 -0700230
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000231 Clock* clock_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000232 // If we are running (have started) or not.
233 bool running_;
kthelgasonff046c72017-03-31 02:03:55 -0700234 rtc::CriticalSection crit_sect_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000235 // Event to signal when we have a frame ready for decoder.
kwiberg3f55dea2016-02-29 05:51:59 -0800236 std::unique_ptr<EventWrapper> frame_event_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000237 // Number of allocated frames.
238 int max_number_of_frames_;
danilchap56359be2017-09-07 07:53:45 -0700239 UnorderedFrameList free_frames_ RTC_GUARDED_BY(crit_sect_);
240 FrameList decodable_frames_ RTC_GUARDED_BY(crit_sect_);
241 FrameList incomplete_frames_ RTC_GUARDED_BY(crit_sect_);
242 VCMDecodingState last_decoded_state_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000243 bool first_packet_since_reset_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000244
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000245 // Number of packets in a row that have been too old.
246 int num_consecutive_old_packets_;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000247 // Number of packets received.
danilchap56359be2017-09-07 07:53:45 -0700248 int num_packets_ RTC_GUARDED_BY(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000249 // Number of duplicated packets received.
danilchap56359be2017-09-07 07:53:45 -0700250 int num_duplicated_packets_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000251
252 // Jitter estimation.
253 // Filter for estimating jitter.
254 VCMJitterEstimator jitter_estimate_;
255 // Calculates network delays used for jitter calculations.
256 VCMInterFrameDelay inter_frame_delay_;
257 VCMJitterSample waiting_for_completion_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000258 int64_t rtt_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000259
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000260 // Holds the internal NACK list (the missing sequence numbers).
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000261 SequenceNumberSet missing_sequence_numbers_;
262 uint16_t latest_received_sequence_number_;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000263 size_t max_nack_list_size_;
264 int max_packet_age_to_nack_; // Measured in sequence numbers.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000265 int max_incomplete_time_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000266
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000267 // Estimated rolling average of packets per frame
268 float average_packets_per_frame_;
269 // average_packets_per_frame converges fast if we have fewer than this many
270 // frames.
271 int frame_counter_;
philipel83f831a2016-03-12 03:30:23 -0800272
henrikg3c089d72015-09-16 05:37:44 -0700273 RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
niklase@google.com470e71d2011-07-07 08:21:25 +0000274};
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000275} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000276
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200277#endif // MODULES_VIDEO_CODING_JITTER_BUFFER_H_