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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
Henrik Kjellander2557b862015-11-18 22:00:21 +010011#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
12#define WEBRTC_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
Henrik Kjellanderff761fb2015-11-04 08:31:52 +010020#include "webrtc/modules/include/module_common_types.h"
philipel83f831a2016-03-12 03:30:23 -080021#include "webrtc/modules/utility/include/process_thread.h"
Edward Lemurc20978e2017-07-06 19:44:34 +020022#include "webrtc/modules/video_coding/decoding_state.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010023#include "webrtc/modules/video_coding/include/video_coding.h"
24#include "webrtc/modules/video_coding/include/video_coding_defines.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010025#include "webrtc/modules/video_coding/inter_frame_delay.h"
26#include "webrtc/modules/video_coding/jitter_buffer_common.h"
27#include "webrtc/modules/video_coding/jitter_estimator.h"
philipel83f831a2016-03-12 03:30:23 -080028#include "webrtc/modules/video_coding/nack_module.h"
Edward Lemurc20978e2017-07-06 19:44:34 +020029#include "webrtc/rtc_base/constructormagic.h"
30#include "webrtc/rtc_base/criticalsection.h"
31#include "webrtc/rtc_base/thread_annotations.h"
mikhal@webrtc.org119c67d2013-01-31 17:18:02 +000032#include "webrtc/typedefs.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000033
stefan@webrtc.org912981f2012-10-12 07:04:52 +000034namespace webrtc {
niklase@google.com470e71d2011-07-07 08:21:25 +000035
philipel9d3ab612015-12-21 04:12:39 -080036enum VCMNackMode { kNack, kNoNack };
niklase@google.com470e71d2011-07-07 08:21:25 +000037
38// forward declarations
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000039class Clock;
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000040class EventFactory;
41class EventWrapper;
niklase@google.com470e71d2011-07-07 08:21:25 +000042class VCMFrameBuffer;
43class VCMPacket;
44class VCMEncodedFrame;
45
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000046typedef std::list<VCMFrameBuffer*> UnorderedFrameList;
47
stefan@webrtc.org912981f2012-10-12 07:04:52 +000048struct VCMJitterSample {
49 VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
50 uint32_t timestamp;
51 uint32_t frame_size;
52 int64_t latest_packet_time;
niklase@google.com470e71d2011-07-07 08:21:25 +000053};
54
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000055class TimestampLessThan {
56 public:
philipel9d3ab612015-12-21 04:12:39 -080057 bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000058 return IsNewerTimestamp(timestamp2, timestamp1);
59 }
60};
61
agalusza@google.comd818dcb2013-07-29 21:48:11 +000062class FrameList
63 : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000064 public:
65 void InsertFrame(VCMFrameBuffer* frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000066 VCMFrameBuffer* PopFrame(uint32_t timestamp);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000067 VCMFrameBuffer* Front() const;
68 VCMFrameBuffer* Back() const;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000069 int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
philipel9d3ab612015-12-21 04:12:39 -080070 UnorderedFrameList* free_frames);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000071 void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
72 UnorderedFrameList* free_frames);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000073 void Reset(UnorderedFrameList* free_frames);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000074};
75
asapersson9a4cd872015-10-23 00:27:14 -070076class Vp9SsMap {
77 public:
78 typedef std::map<uint32_t, GofInfoVP9, TimestampLessThan> SsMap;
79 bool Insert(const VCMPacket& packet);
80 void Reset();
81
82 // Removes SS data that are older than |timestamp|.
83 // The |timestamp| should be an old timestamp, i.e. packets with older
84 // timestamps should no longer be inserted.
85 void RemoveOld(uint32_t timestamp);
86
87 bool UpdatePacket(VCMPacket* packet);
88 void UpdateFrames(FrameList* frames);
89
90 // Public for testing.
91 // Returns an iterator to the corresponding SS data for the input |timestamp|.
92 bool Find(uint32_t timestamp, SsMap::iterator* it);
93
94 private:
95 // These two functions are called by RemoveOld.
96 // Checks if it is time to do a clean up (done each kSsCleanupIntervalSec).
97 bool TimeForCleanup(uint32_t timestamp) const;
98
99 // Advances the oldest SS data to handle timestamp wrap in cases where SS data
100 // are received very seldom (e.g. only once in beginning, second when
101 // IsNewerTimestamp is not true).
102 void AdvanceFront(uint32_t timestamp);
103
104 SsMap ss_map_;
105};
106
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000107class VCMJitterBuffer {
108 public:
philipel83f831a2016-03-12 03:30:23 -0800109 VCMJitterBuffer(Clock* clock,
110 std::unique_ptr<EventWrapper> event,
111 NackSender* nack_sender = nullptr,
112 KeyFrameRequestSender* keyframe_request_sender = nullptr);
Qiang Chend4cec152015-06-19 09:17:00 -0700113
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -0700114 ~VCMJitterBuffer();
niklase@google.com470e71d2011-07-07 08:21:25 +0000115
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000116 // Initializes and starts jitter buffer.
117 void Start();
niklase@google.com470e71d2011-07-07 08:21:25 +0000118
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000119 // Signals all internal events and stops the jitter buffer.
120 void Stop();
niklase@google.com470e71d2011-07-07 08:21:25 +0000121
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000122 // Returns true if the jitter buffer is running.
123 bool Running() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000124
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000125 // Empty the jitter buffer of all its data.
126 void Flush();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000127
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000128 // Get the number of received frames, by type, since the jitter buffer
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000129 // was started.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000130 FrameCounts FrameStatistics() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000131
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000132 // Gets number of packets received.
133 int num_packets() const;
134
135 // Gets number of duplicated packets received.
136 int num_duplicated_packets() const;
137
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000138 // Gets number of packets discarded by the jitter buffer.
139 int num_discarded_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000140
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000141 // Statistics, Calculate frame and bit rates.
philipel9d3ab612015-12-21 04:12:39 -0800142 void IncomingRateStatistics(unsigned int* framerate, unsigned int* bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000143
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000144 // Wait |max_wait_time_ms| for a complete frame to arrive.
isheriff6b4b5f32016-06-08 00:24:21 -0700145 // If found, a pointer to the frame is returned. Returns nullptr otherwise.
146 VCMEncodedFrame* NextCompleteFrame(uint32_t max_wait_time_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000147
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000148 // Locates a frame for decoding (even an incomplete) without delay.
149 // The function returns true once such a frame is found, its corresponding
150 // timestamp is returned. Otherwise, returns false.
151 bool NextMaybeIncompleteTimestamp(uint32_t* timestamp);
152
153 // Extract frame corresponding to input timestamp.
154 // Frame will be set to a decoding state.
155 VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000156
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000157 // Releases a frame returned from the jitter buffer, should be called when
158 // done with decoding.
159 void ReleaseFrame(VCMEncodedFrame* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000160
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000161 // Returns the time in ms when the latest packet was inserted into the frame.
162 // Retransmitted is set to true if any of the packets belonging to the frame
163 // has been retransmitted.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000164 int64_t LastPacketTime(const VCMEncodedFrame* frame,
165 bool* retransmitted) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000166
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000167 // Inserts a packet into a frame returned from GetFrame().
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000168 // If the return value is <= 0, |frame| is invalidated and the pointer must
169 // be dropped after this function returns.
philipel9d3ab612015-12-21 04:12:39 -0800170 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
niklase@google.com470e71d2011-07-07 08:21:25 +0000171
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000172 // Returns the estimated jitter in milliseconds.
173 uint32_t EstimatedJitterMs();
niklase@google.com470e71d2011-07-07 08:21:25 +0000174
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000175 // Updates the round-trip time estimate.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000176 void UpdateRtt(int64_t rtt_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000177
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700178 // Set the NACK mode. |high_rtt_nack_threshold_ms| is an RTT threshold in ms
Wan-Teh Changf2912872015-06-05 13:16:45 -0700179 // above which NACK will be disabled if the NACK mode is |kNack|, -1 meaning
180 // that NACK is always enabled in the |kNack| mode.
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700181 // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
182 // to rely on NACK only, and therefore are using larger buffers to have time
183 // to wait for retransmissions.
philipel9d3ab612015-12-21 04:12:39 -0800184 void SetNackMode(VCMNackMode mode,
185 int64_t low_rtt_nack_threshold_ms,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000186 int64_t high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000187
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000188 void SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000189 int max_packet_age_to_nack,
190 int max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000191
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000192 // Returns the current NACK mode.
193 VCMNackMode nack_mode() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000194
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000195 // Returns a list of the sequence numbers currently missing.
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700196 std::vector<uint16_t> GetNackList(bool* request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000197
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000198 // Set decode error mode - Should not be changed in the middle of the
199 // session. Changes will not influence frames already in the buffer.
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000200 void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
philipel9d3ab612015-12-21 04:12:39 -0800201 VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
stefan@webrtc.org4c059d82011-10-13 07:35:37 +0000202
pbos@webrtc.org55707692014-12-19 15:45:03 +0000203 void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000204
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000205 private:
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000206 class SequenceNumberLessThan {
207 public:
philipel9d3ab612015-12-21 04:12:39 -0800208 bool operator()(const uint16_t& sequence_number1,
209 const uint16_t& sequence_number2) const {
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +0000210 return IsNewerSequenceNumber(sequence_number2, sequence_number1);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000211 }
212 };
213 typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
214
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000215 // Gets the frame assigned to the timestamp of the packet. May recycle
216 // existing frames if no free frames are available. Returns an error code if
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000217 // failing, or kNoError on success. |frame_list| contains which list the
218 // packet was in, or NULL if it was not in a FrameList (a new frame).
219 VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
220 VCMFrameBuffer** frame,
221 FrameList** frame_list)
danilchap56359be2017-09-07 07:53:45 -0700222 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000223
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000224 // Returns true if |frame| is continuous in |decoding_state|, not taking
225 // decodable frames into account.
226 bool IsContinuousInState(const VCMFrameBuffer& frame,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000227 const VCMDecodingState& decoding_state) const
danilchap56359be2017-09-07 07:53:45 -0700228 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000229 // Returns true if |frame| is continuous in the |last_decoded_state_|, taking
230 // all decodable frames into account.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000231 bool IsContinuous(const VCMFrameBuffer& frame) const
danilchap56359be2017-09-07 07:53:45 -0700232 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700233 // Looks for frames in |incomplete_frames_| which are continuous in the
234 // provided |decoded_state|. Starts the search from the timestamp of
235 // |decoded_state|.
236 void FindAndInsertContinuousFramesWithState(
237 const VCMDecodingState& decoded_state)
danilchap56359be2017-09-07 07:53:45 -0700238 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000239 // Looks for frames in |incomplete_frames_| which are continuous in
240 // |last_decoded_state_| taking all decodable frames into account. Starts
241 // the search from |new_frame|.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000242 void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
danilchap56359be2017-09-07 07:53:45 -0700243 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
244 VCMFrameBuffer* NextFrame() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000245 // Returns true if the NACK list was updated to cover sequence numbers up to
246 // |sequence_number|. If false a key frame is needed to get into a state where
247 // we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000248 bool UpdateNackList(uint16_t sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700249 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000250 bool TooLargeNackList() const;
251 // Returns true if the NACK list was reduced without problem. If false a key
252 // frame is needed to get into a state where we can continue decoding.
danilchap56359be2017-09-07 07:53:45 -0700253 bool HandleTooLargeNackList() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000254 bool MissingTooOldPacket(uint16_t latest_sequence_number) const
danilchap56359be2017-09-07 07:53:45 -0700255 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000256 // Returns true if the too old packets was successfully removed from the NACK
257 // list. If false, a key frame is needed to get into a state where we can
258 // continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000259 bool HandleTooOldPackets(uint16_t latest_sequence_number)
danilchap56359be2017-09-07 07:53:45 -0700260 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000261 // Drops all packets in the NACK list up until |last_decoded_sequence_number|.
262 void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
263
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000264 // Gets an empty frame, creating a new frame if necessary (i.e. increases
265 // jitter buffer size).
danilchap56359be2017-09-07 07:53:45 -0700266 VCMFrameBuffer* GetEmptyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000267
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000268 // Attempts to increase the size of the jitter buffer. Returns true on
269 // success, false otherwise.
danilchap56359be2017-09-07 07:53:45 -0700270 bool TryToIncreaseJitterBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000271
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000272 // Recycles oldest frames until a key frame is found. Used if jitter buffer is
273 // completely full. Returns true if a key frame was found.
danilchap56359be2017-09-07 07:53:45 -0700274 bool RecycleFramesUntilKeyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000275
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000276 // Updates the frame statistics.
agalusza@google.comd177c102013-08-08 01:12:33 +0000277 // Counts only complete frames, so decodable incomplete frames will not be
278 // counted.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000279 void CountFrame(const VCMFrameBuffer& frame)
danilchap56359be2017-09-07 07:53:45 -0700280 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000281
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000282 // Update rolling average of packets per frame.
283 void UpdateAveragePacketsPerFrame(int current_number_packets_);
284
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000285 // Cleans the frame list in the JB from old/empty frames.
286 // Should only be called prior to actual use.
danilchap56359be2017-09-07 07:53:45 -0700287 void CleanUpOldOrEmptyFrames() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000288
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000289 // Returns true if |packet| is likely to have been retransmitted.
290 bool IsPacketRetransmitted(const VCMPacket& packet) const;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +0000291
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000292 // The following three functions update the jitter estimate with the
293 // payload size, receive time and RTP timestamp of a frame.
294 void UpdateJitterEstimate(const VCMJitterSample& sample,
295 bool incomplete_frame);
296 void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
297 void UpdateJitterEstimate(int64_t latest_packet_time_ms,
298 uint32_t timestamp,
299 unsigned int frame_size,
300 bool incomplete_frame);
301
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000302 // Returns true if we should wait for retransmissions, false otherwise.
303 bool WaitForRetransmissions();
304
danilchap56359be2017-09-07 07:53:45 -0700305 int NonContinuousOrIncompleteDuration()
306 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000307
308 uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
309
danilchap56359be2017-09-07 07:53:45 -0700310 void UpdateHistograms() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000311
sprang22691e02016-07-13 10:57:07 -0700312 // Reset frame buffer and return it to free_frames_.
313 void RecycleFrameBuffer(VCMFrameBuffer* frame)
danilchap56359be2017-09-07 07:53:45 -0700314 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
sprang22691e02016-07-13 10:57:07 -0700315
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000316 Clock* clock_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000317 // If we are running (have started) or not.
318 bool running_;
kthelgasonff046c72017-03-31 02:03:55 -0700319 rtc::CriticalSection crit_sect_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000320 // Event to signal when we have a frame ready for decoder.
kwiberg3f55dea2016-02-29 05:51:59 -0800321 std::unique_ptr<EventWrapper> frame_event_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000322 // Number of allocated frames.
323 int max_number_of_frames_;
danilchap56359be2017-09-07 07:53:45 -0700324 UnorderedFrameList free_frames_ RTC_GUARDED_BY(crit_sect_);
325 FrameList decodable_frames_ RTC_GUARDED_BY(crit_sect_);
326 FrameList incomplete_frames_ RTC_GUARDED_BY(crit_sect_);
327 VCMDecodingState last_decoded_state_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000328 bool first_packet_since_reset_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000329
330 // Statistics.
danilchap56359be2017-09-07 07:53:45 -0700331 VCMReceiveStatisticsCallback* stats_callback_ RTC_GUARDED_BY(crit_sect_);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000332 // Frame counts for each type (key, delta, ...)
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000333 FrameCounts receive_statistics_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000334 // Latest calculated frame rates of incoming stream.
335 unsigned int incoming_frame_rate_;
336 unsigned int incoming_frame_count_;
337 int64_t time_last_incoming_frame_count_;
338 unsigned int incoming_bit_count_;
339 unsigned int incoming_bit_rate_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000340 // Number of packets in a row that have been too old.
341 int num_consecutive_old_packets_;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000342 // Number of packets received.
danilchap56359be2017-09-07 07:53:45 -0700343 int num_packets_ RTC_GUARDED_BY(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000344 // Number of duplicated packets received.
danilchap56359be2017-09-07 07:53:45 -0700345 int num_duplicated_packets_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000346 // Number of packets discarded by the jitter buffer.
danilchap56359be2017-09-07 07:53:45 -0700347 int num_discarded_packets_ RTC_GUARDED_BY(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000348 // Time when first packet is received.
danilchap56359be2017-09-07 07:53:45 -0700349 int64_t time_first_packet_ms_ RTC_GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000350
351 // Jitter estimation.
352 // Filter for estimating jitter.
353 VCMJitterEstimator jitter_estimate_;
354 // Calculates network delays used for jitter calculations.
355 VCMInterFrameDelay inter_frame_delay_;
356 VCMJitterSample waiting_for_completion_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000357 int64_t rtt_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000358
359 // NACK and retransmissions.
360 VCMNackMode nack_mode_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000361 int64_t low_rtt_nack_threshold_ms_;
362 int64_t high_rtt_nack_threshold_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000363 // Holds the internal NACK list (the missing sequence numbers).
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000364 SequenceNumberSet missing_sequence_numbers_;
365 uint16_t latest_received_sequence_number_;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000366 size_t max_nack_list_size_;
367 int max_packet_age_to_nack_; // Measured in sequence numbers.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000368 int max_incomplete_time_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000369
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000370 VCMDecodeErrorMode decode_error_mode_;
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000371 // Estimated rolling average of packets per frame
372 float average_packets_per_frame_;
373 // average_packets_per_frame converges fast if we have fewer than this many
374 // frames.
375 int frame_counter_;
philipel83f831a2016-03-12 03:30:23 -0800376
henrikg3c089d72015-09-16 05:37:44 -0700377 RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
niklase@google.com470e71d2011-07-07 08:21:25 +0000378};
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000379} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000380
Henrik Kjellander2557b862015-11-18 22:00:21 +0100381#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_