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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"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020028#include "rtc_base/constructormagic.h"
29#include "rtc_base/criticalsection.h"
30#include "rtc_base/thread_annotations.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000031
stefan@webrtc.org912981f2012-10-12 07:04:52 +000032namespace webrtc {
niklase@google.com470e71d2011-07-07 08:21:25 +000033
philipel9d3ab612015-12-21 04:12:39 -080034enum VCMNackMode { kNack, kNoNack };
niklase@google.com470e71d2011-07-07 08:21:25 +000035
36// forward declarations
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000037class Clock;
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000038class EventFactory;
39class EventWrapper;
niklase@google.com470e71d2011-07-07 08:21:25 +000040class VCMFrameBuffer;
41class VCMPacket;
42class VCMEncodedFrame;
43
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000044typedef std::list<VCMFrameBuffer*> UnorderedFrameList;
45
stefan@webrtc.org912981f2012-10-12 07:04:52 +000046struct VCMJitterSample {
47 VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
48 uint32_t timestamp;
49 uint32_t frame_size;
50 int64_t latest_packet_time;
niklase@google.com470e71d2011-07-07 08:21:25 +000051};
52
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000053class TimestampLessThan {
54 public:
philipel9d3ab612015-12-21 04:12:39 -080055 bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000056 return IsNewerTimestamp(timestamp2, timestamp1);
57 }
58};
59
agalusza@google.comd818dcb2013-07-29 21:48:11 +000060class FrameList
61 : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000062 public:
63 void InsertFrame(VCMFrameBuffer* frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000064 VCMFrameBuffer* PopFrame(uint32_t timestamp);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000065 VCMFrameBuffer* Front() const;
66 VCMFrameBuffer* Back() const;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000067 int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
philipel9d3ab612015-12-21 04:12:39 -080068 UnorderedFrameList* free_frames);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000069 void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
70 UnorderedFrameList* free_frames);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000071 void Reset(UnorderedFrameList* free_frames);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000072};
73
asapersson9a4cd872015-10-23 00:27:14 -070074class Vp9SsMap {
75 public:
76 typedef std::map<uint32_t, GofInfoVP9, TimestampLessThan> SsMap;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020077 Vp9SsMap();
78 ~Vp9SsMap();
79
asapersson9a4cd872015-10-23 00:27:14 -070080 bool Insert(const VCMPacket& packet);
81 void Reset();
82
83 // Removes SS data that are older than |timestamp|.
84 // The |timestamp| should be an old timestamp, i.e. packets with older
85 // timestamps should no longer be inserted.
86 void RemoveOld(uint32_t timestamp);
87
88 bool UpdatePacket(VCMPacket* packet);
89 void UpdateFrames(FrameList* frames);
90
91 // Public for testing.
92 // Returns an iterator to the corresponding SS data for the input |timestamp|.
93 bool Find(uint32_t timestamp, SsMap::iterator* it);
94
95 private:
96 // These two functions are called by RemoveOld.
97 // Checks if it is time to do a clean up (done each kSsCleanupIntervalSec).
98 bool TimeForCleanup(uint32_t timestamp) const;
99
100 // Advances the oldest SS data to handle timestamp wrap in cases where SS data
101 // are received very seldom (e.g. only once in beginning, second when
102 // IsNewerTimestamp is not true).
103 void AdvanceFront(uint32_t timestamp);
104
105 SsMap ss_map_;
106};
107
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000108class VCMJitterBuffer {
109 public:
philipel83f831a2016-03-12 03:30:23 -0800110 VCMJitterBuffer(Clock* clock,
111 std::unique_ptr<EventWrapper> event,
112 NackSender* nack_sender = nullptr,
113 KeyFrameRequestSender* keyframe_request_sender = nullptr);
Qiang Chend4cec152015-06-19 09:17:00 -0700114
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -0700115 ~VCMJitterBuffer();
niklase@google.com470e71d2011-07-07 08:21:25 +0000116
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000117 // Initializes and starts jitter buffer.
118 void Start();
niklase@google.com470e71d2011-07-07 08:21:25 +0000119
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000120 // Signals all internal events and stops the jitter buffer.
121 void Stop();
niklase@google.com470e71d2011-07-07 08:21:25 +0000122
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000123 // Returns true if the jitter buffer is running.
124 bool Running() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000125
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000126 // Empty the jitter buffer of all its data.
127 void Flush();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000128
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000129 // Get the number of received frames, by type, since the jitter buffer
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000130 // was started.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000131 FrameCounts FrameStatistics() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000132
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000133 // Gets number of packets received.
134 int num_packets() const;
135
136 // Gets number of duplicated packets received.
137 int num_duplicated_packets() const;
138
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000139 // Gets number of packets discarded by the jitter buffer.
140 int num_discarded_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000141
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000142 // Statistics, Calculate frame and bit rates.
philipel9d3ab612015-12-21 04:12:39 -0800143 void IncomingRateStatistics(unsigned int* framerate, unsigned int* bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000144
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000145 // Wait |max_wait_time_ms| for a complete frame to arrive.
isheriff6b4b5f32016-06-08 00:24:21 -0700146 // If found, a pointer to the frame is returned. Returns nullptr otherwise.
147 VCMEncodedFrame* NextCompleteFrame(uint32_t max_wait_time_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000148
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000149 // Locates a frame for decoding (even an incomplete) without delay.
150 // The function returns true once such a frame is found, its corresponding
151 // timestamp is returned. Otherwise, returns false.
152 bool NextMaybeIncompleteTimestamp(uint32_t* timestamp);
153
154 // Extract frame corresponding to input timestamp.
155 // Frame will be set to a decoding state.
156 VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000157
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000158 // Releases a frame returned from the jitter buffer, should be called when
159 // done with decoding.
160 void ReleaseFrame(VCMEncodedFrame* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000161
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000162 // Returns the time in ms when the latest packet was inserted into the frame.
163 // Retransmitted is set to true if any of the packets belonging to the frame
164 // has been retransmitted.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000165 int64_t LastPacketTime(const VCMEncodedFrame* frame,
166 bool* retransmitted) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000167
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000168 // Inserts a packet into a frame returned from GetFrame().
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000169 // If the return value is <= 0, |frame| is invalidated and the pointer must
170 // be dropped after this function returns.
philipel9d3ab612015-12-21 04:12:39 -0800171 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
niklase@google.com470e71d2011-07-07 08:21:25 +0000172
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000173 // Returns the estimated jitter in milliseconds.
174 uint32_t EstimatedJitterMs();
niklase@google.com470e71d2011-07-07 08:21:25 +0000175
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000176 // Updates the round-trip time estimate.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000177 void UpdateRtt(int64_t rtt_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000178
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700179 // Set the NACK mode. |high_rtt_nack_threshold_ms| is an RTT threshold in ms
Wan-Teh Changf2912872015-06-05 13:16:45 -0700180 // above which NACK will be disabled if the NACK mode is |kNack|, -1 meaning
181 // that NACK is always enabled in the |kNack| mode.
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700182 // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
183 // to rely on NACK only, and therefore are using larger buffers to have time
184 // to wait for retransmissions.
philipel9d3ab612015-12-21 04:12:39 -0800185 void SetNackMode(VCMNackMode mode,
186 int64_t low_rtt_nack_threshold_ms,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000187 int64_t high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000188
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000189 void SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000190 int max_packet_age_to_nack,
191 int max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000192
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000193 // Returns the current NACK mode.
194 VCMNackMode nack_mode() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000195
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000196 // Returns a list of the sequence numbers currently missing.
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700197 std::vector<uint16_t> GetNackList(bool* request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000198
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000199 // Set decode error mode - Should not be changed in the middle of the
200 // session. Changes will not influence frames already in the buffer.
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000201 void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
philipel9d3ab612015-12-21 04:12:39 -0800202 VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
stefan@webrtc.org4c059d82011-10-13 07:35:37 +0000203
pbos@webrtc.org55707692014-12-19 15:45:03 +0000204 void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000205
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000206 private:
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000207 class SequenceNumberLessThan {
208 public:
philipel9d3ab612015-12-21 04:12:39 -0800209 bool operator()(const uint16_t& sequence_number1,
210 const uint16_t& sequence_number2) const {
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +0000211 return IsNewerSequenceNumber(sequence_number2, sequence_number1);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000212 }
213 };
214 typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
215
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000216 // Gets the frame assigned to the timestamp of the packet. May recycle
217 // existing frames if no free frames are available. Returns an error code if
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000218 // failing, or kNoError on success. |frame_list| contains which list the
219 // packet was in, or NULL if it was not in a FrameList (a new frame).
220 VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
221 VCMFrameBuffer** frame,
222 FrameList** frame_list)
danilchap56359be2017-09-07 07:53:45 -0700223 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000224
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000225 // Returns true if |frame| is continuous in |decoding_state|, not taking
226 // decodable frames into account.
227 bool IsContinuousInState(const VCMFrameBuffer& frame,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000228 const VCMDecodingState& decoding_state) const
danilchap56359be2017-09-07 07:53:45 -0700229 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000230 // Returns true if |frame| is continuous in the |last_decoded_state_|, taking
231 // all decodable frames into account.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000232 bool IsContinuous(const VCMFrameBuffer& frame) const
danilchap56359be2017-09-07 07:53:45 -0700233 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700234 // Looks for frames in |incomplete_frames_| which are continuous in the
235 // provided |decoded_state|. Starts the search from the timestamp of
236 // |decoded_state|.
237 void FindAndInsertContinuousFramesWithState(
238 const VCMDecodingState& decoded_state)
danilchap56359be2017-09-07 07:53:45 -0700239 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000240 // Looks for frames in |incomplete_frames_| which are continuous in
241 // |last_decoded_state_| taking all decodable frames into account. Starts
242 // the search from |new_frame|.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000243 void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
danilchap56359be2017-09-07 07:53:45 -0700244 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
245 VCMFrameBuffer* NextFrame() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000246 // Returns true if the NACK list was updated to cover sequence numbers up to
247 // |sequence_number|. If false a key frame is needed to get into a state where
248 // we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000249 bool UpdateNackList(uint16_t sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700250 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000251 bool TooLargeNackList() const;
252 // Returns true if the NACK list was reduced without problem. If false a key
253 // frame is needed to get into a state where we can continue decoding.
danilchap56359be2017-09-07 07:53:45 -0700254 bool HandleTooLargeNackList() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000255 bool MissingTooOldPacket(uint16_t latest_sequence_number) const
danilchap56359be2017-09-07 07:53:45 -0700256 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000257 // Returns true if the too old packets was successfully removed from the NACK
258 // list. If false, a key frame is needed to get into a state where we can
259 // continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000260 bool HandleTooOldPackets(uint16_t latest_sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700261 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000262 // Drops all packets in the NACK list up until |last_decoded_sequence_number|.
263 void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
264
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000265 // Gets an empty frame, creating a new frame if necessary (i.e. increases
266 // jitter buffer size).
danilchap56359be2017-09-07 07:53:45 -0700267 VCMFrameBuffer* GetEmptyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000268
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000269 // Attempts to increase the size of the jitter buffer. Returns true on
270 // success, false otherwise.
danilchap56359be2017-09-07 07:53:45 -0700271 bool TryToIncreaseJitterBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000272
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000273 // Recycles oldest frames until a key frame is found. Used if jitter buffer is
274 // completely full. Returns true if a key frame was found.
danilchap56359be2017-09-07 07:53:45 -0700275 bool RecycleFramesUntilKeyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000276
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000277 // Updates the frame statistics.
agalusza@google.comd177c102013-08-08 01:12:33 +0000278 // Counts only complete frames, so decodable incomplete frames will not be
279 // counted.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000280 void CountFrame(const VCMFrameBuffer& frame)
danilchap56359be2017-09-07 07:53:45 -0700281 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000282
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000283 // Update rolling average of packets per frame.
284 void UpdateAveragePacketsPerFrame(int current_number_packets_);
285
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000286 // Cleans the frame list in the JB from old/empty frames.
287 // Should only be called prior to actual use.
danilchap56359be2017-09-07 07:53:45 -0700288 void CleanUpOldOrEmptyFrames() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000289
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000290 // Returns true if |packet| is likely to have been retransmitted.
291 bool IsPacketRetransmitted(const VCMPacket& packet) const;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +0000292
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000293 // The following three functions update the jitter estimate with the
294 // payload size, receive time and RTP timestamp of a frame.
295 void UpdateJitterEstimate(const VCMJitterSample& sample,
296 bool incomplete_frame);
297 void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
298 void UpdateJitterEstimate(int64_t latest_packet_time_ms,
299 uint32_t timestamp,
300 unsigned int frame_size,
301 bool incomplete_frame);
302
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000303 // Returns true if we should wait for retransmissions, false otherwise.
304 bool WaitForRetransmissions();
305
danilchap56359be2017-09-07 07:53:45 -0700306 int NonContinuousOrIncompleteDuration()
307 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000308
309 uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
310
danilchap56359be2017-09-07 07:53:45 -0700311 void UpdateHistograms() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000312
sprang22691e02016-07-13 10:57:07 -0700313 // Reset frame buffer and return it to free_frames_.
314 void RecycleFrameBuffer(VCMFrameBuffer* frame)
danilchap56359be2017-09-07 07:53:45 -0700315 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
sprang22691e02016-07-13 10:57:07 -0700316
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000317 Clock* clock_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000318 // If we are running (have started) or not.
319 bool running_;
kthelgasonff046c72017-03-31 02:03:55 -0700320 rtc::CriticalSection crit_sect_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000321 // Event to signal when we have a frame ready for decoder.
kwiberg3f55dea2016-02-29 05:51:59 -0800322 std::unique_ptr<EventWrapper> frame_event_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000323 // Number of allocated frames.
324 int max_number_of_frames_;
danilchap56359be2017-09-07 07:53:45 -0700325 UnorderedFrameList free_frames_ RTC_GUARDED_BY(crit_sect_);
326 FrameList decodable_frames_ RTC_GUARDED_BY(crit_sect_);
327 FrameList incomplete_frames_ RTC_GUARDED_BY(crit_sect_);
328 VCMDecodingState last_decoded_state_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000329 bool first_packet_since_reset_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000330
331 // Statistics.
danilchap56359be2017-09-07 07:53:45 -0700332 VCMReceiveStatisticsCallback* stats_callback_ RTC_GUARDED_BY(crit_sect_);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000333 // Frame counts for each type (key, delta, ...)
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000334 FrameCounts receive_statistics_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000335 // Latest calculated frame rates of incoming stream.
336 unsigned int incoming_frame_rate_;
337 unsigned int incoming_frame_count_;
338 int64_t time_last_incoming_frame_count_;
339 unsigned int incoming_bit_count_;
340 unsigned int incoming_bit_rate_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000341 // Number of packets in a row that have been too old.
342 int num_consecutive_old_packets_;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000343 // Number of packets received.
danilchap56359be2017-09-07 07:53:45 -0700344 int num_packets_ RTC_GUARDED_BY(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000345 // Number of duplicated packets received.
danilchap56359be2017-09-07 07:53:45 -0700346 int num_duplicated_packets_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000347 // Number of packets discarded by the jitter buffer.
danilchap56359be2017-09-07 07:53:45 -0700348 int num_discarded_packets_ RTC_GUARDED_BY(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000349 // Time when first packet is received.
danilchap56359be2017-09-07 07:53:45 -0700350 int64_t time_first_packet_ms_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000351
352 // Jitter estimation.
353 // Filter for estimating jitter.
354 VCMJitterEstimator jitter_estimate_;
355 // Calculates network delays used for jitter calculations.
356 VCMInterFrameDelay inter_frame_delay_;
357 VCMJitterSample waiting_for_completion_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000358 int64_t rtt_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000359
360 // NACK and retransmissions.
361 VCMNackMode nack_mode_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000362 int64_t low_rtt_nack_threshold_ms_;
363 int64_t high_rtt_nack_threshold_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000364 // Holds the internal NACK list (the missing sequence numbers).
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000365 SequenceNumberSet missing_sequence_numbers_;
366 uint16_t latest_received_sequence_number_;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000367 size_t max_nack_list_size_;
368 int max_packet_age_to_nack_; // Measured in sequence numbers.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000369 int max_incomplete_time_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000370
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000371 VCMDecodeErrorMode decode_error_mode_;
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000372 // Estimated rolling average of packets per frame
373 float average_packets_per_frame_;
374 // average_packets_per_frame converges fast if we have fewer than this many
375 // frames.
376 int frame_counter_;
philipel83f831a2016-03-12 03:30:23 -0800377
henrikg3c089d72015-09-16 05:37:44 -0700378 RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
niklase@google.com470e71d2011-07-07 08:21:25 +0000379};
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000380} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000381
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200382#endif // MODULES_VIDEO_CODING_JITTER_BUFFER_H_