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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>
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000016#include <set>
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +000017#include <vector>
stefan@webrtc.org29794612012-02-08 08:58:55 +000018
henrike@webrtc.org88fbb2d2014-05-21 21:18:46 +000019#include "webrtc/base/constructormagic.h"
asapersson@webrtc.org83b52002014-11-28 10:17:13 +000020#include "webrtc/base/thread_annotations.h"
Henrik Kjellanderff761fb2015-11-04 08:31:52 +010021#include "webrtc/modules/include/module_common_types.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010022#include "webrtc/modules/video_coding/include/video_coding.h"
23#include "webrtc/modules/video_coding/include/video_coding_defines.h"
24#include "webrtc/modules/video_coding/decoding_state.h"
25#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"
Henrik Kjellander98f53512015-10-28 18:17:40 +010028#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
mikhal@webrtc.org119c67d2013-01-31 17:18:02 +000029#include "webrtc/typedefs.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000030
stefan@webrtc.org912981f2012-10-12 07:04:52 +000031namespace webrtc {
niklase@google.com470e71d2011-07-07 08:21:25 +000032
philipel9d3ab612015-12-21 04:12:39 -080033enum VCMNackMode { kNack, kNoNack };
niklase@google.com470e71d2011-07-07 08:21:25 +000034
35// forward declarations
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000036class Clock;
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000037class EventFactory;
38class EventWrapper;
niklase@google.com470e71d2011-07-07 08:21:25 +000039class VCMFrameBuffer;
40class VCMPacket;
41class VCMEncodedFrame;
42
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000043typedef std::list<VCMFrameBuffer*> UnorderedFrameList;
44
stefan@webrtc.org912981f2012-10-12 07:04:52 +000045struct VCMJitterSample {
46 VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
47 uint32_t timestamp;
48 uint32_t frame_size;
49 int64_t latest_packet_time;
niklase@google.com470e71d2011-07-07 08:21:25 +000050};
51
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000052class TimestampLessThan {
53 public:
philipel9d3ab612015-12-21 04:12:39 -080054 bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000055 return IsNewerTimestamp(timestamp2, timestamp1);
56 }
57};
58
agalusza@google.comd818dcb2013-07-29 21:48:11 +000059class FrameList
60 : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000061 public:
62 void InsertFrame(VCMFrameBuffer* frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000063 VCMFrameBuffer* PopFrame(uint32_t timestamp);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000064 VCMFrameBuffer* Front() const;
65 VCMFrameBuffer* Back() const;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000066 int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
philipel9d3ab612015-12-21 04:12:39 -080067 UnorderedFrameList* free_frames);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000068 void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
69 UnorderedFrameList* free_frames);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000070 void Reset(UnorderedFrameList* free_frames);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000071};
72
asapersson9a4cd872015-10-23 00:27:14 -070073class Vp9SsMap {
74 public:
75 typedef std::map<uint32_t, GofInfoVP9, TimestampLessThan> SsMap;
76 bool Insert(const VCMPacket& packet);
77 void Reset();
78
79 // Removes SS data that are older than |timestamp|.
80 // The |timestamp| should be an old timestamp, i.e. packets with older
81 // timestamps should no longer be inserted.
82 void RemoveOld(uint32_t timestamp);
83
84 bool UpdatePacket(VCMPacket* packet);
85 void UpdateFrames(FrameList* frames);
86
87 // Public for testing.
88 // Returns an iterator to the corresponding SS data for the input |timestamp|.
89 bool Find(uint32_t timestamp, SsMap::iterator* it);
90
91 private:
92 // These two functions are called by RemoveOld.
93 // Checks if it is time to do a clean up (done each kSsCleanupIntervalSec).
94 bool TimeForCleanup(uint32_t timestamp) const;
95
96 // Advances the oldest SS data to handle timestamp wrap in cases where SS data
97 // are received very seldom (e.g. only once in beginning, second when
98 // IsNewerTimestamp is not true).
99 void AdvanceFront(uint32_t timestamp);
100
101 SsMap ss_map_;
102};
103
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000104class VCMJitterBuffer {
105 public:
Qiang Chend4cec152015-06-19 09:17:00 -0700106 VCMJitterBuffer(Clock* clock, rtc::scoped_ptr<EventWrapper> event);
107
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -0700108 ~VCMJitterBuffer();
niklase@google.com470e71d2011-07-07 08:21:25 +0000109
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000110 // Initializes and starts jitter buffer.
111 void Start();
niklase@google.com470e71d2011-07-07 08:21:25 +0000112
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000113 // Signals all internal events and stops the jitter buffer.
114 void Stop();
niklase@google.com470e71d2011-07-07 08:21:25 +0000115
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000116 // Returns true if the jitter buffer is running.
117 bool Running() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000118
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000119 // Empty the jitter buffer of all its data.
120 void Flush();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000121
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000122 // Get the number of received frames, by type, since the jitter buffer
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000123 // was started.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000124 FrameCounts FrameStatistics() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000125
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000126 // The number of packets discarded by the jitter buffer because the decoder
127 // won't be able to decode them.
128 int num_not_decodable_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000129
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000130 // Gets number of packets received.
131 int num_packets() const;
132
133 // Gets number of duplicated packets received.
134 int num_duplicated_packets() const;
135
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000136 // Gets number of packets discarded by the jitter buffer.
137 int num_discarded_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000138
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000139 // Statistics, Calculate frame and bit rates.
philipel9d3ab612015-12-21 04:12:39 -0800140 void IncomingRateStatistics(unsigned int* framerate, unsigned int* bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000141
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000142 // Checks if the packet sequence will be complete if the next frame would be
143 // grabbed for decoding. That is, if a frame has been lost between the
144 // last decoded frame and the next, or if the next frame is missing one
145 // or more packets.
146 bool CompleteSequenceWithNextFrame();
niklase@google.com470e71d2011-07-07 08:21:25 +0000147
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000148 // Wait |max_wait_time_ms| for a complete frame to arrive.
149 // The function returns true once such a frame is found, its corresponding
150 // timestamp is returned. Otherwise, returns false.
151 bool NextCompleteTimestamp(uint32_t max_wait_time_ms, uint32_t* timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000152
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000153 // Locates a frame for decoding (even an incomplete) without delay.
154 // The function returns true once such a frame is found, its corresponding
155 // timestamp is returned. Otherwise, returns false.
156 bool NextMaybeIncompleteTimestamp(uint32_t* timestamp);
157
158 // Extract frame corresponding to input timestamp.
159 // Frame will be set to a decoding state.
160 VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000161
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000162 // Releases a frame returned from the jitter buffer, should be called when
163 // done with decoding.
164 void ReleaseFrame(VCMEncodedFrame* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000165
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000166 // Returns the time in ms when the latest packet was inserted into the frame.
167 // Retransmitted is set to true if any of the packets belonging to the frame
168 // has been retransmitted.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000169 int64_t LastPacketTime(const VCMEncodedFrame* frame,
170 bool* retransmitted) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000171
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000172 // Inserts a packet into a frame returned from GetFrame().
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000173 // If the return value is <= 0, |frame| is invalidated and the pointer must
174 // be dropped after this function returns.
philipel9d3ab612015-12-21 04:12:39 -0800175 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
niklase@google.com470e71d2011-07-07 08:21:25 +0000176
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000177 // Returns the estimated jitter in milliseconds.
178 uint32_t EstimatedJitterMs();
niklase@google.com470e71d2011-07-07 08:21:25 +0000179
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000180 // Updates the round-trip time estimate.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000181 void UpdateRtt(int64_t rtt_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000182
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700183 // Set the NACK mode. |high_rtt_nack_threshold_ms| is an RTT threshold in ms
Wan-Teh Changf2912872015-06-05 13:16:45 -0700184 // above which NACK will be disabled if the NACK mode is |kNack|, -1 meaning
185 // that NACK is always enabled in the |kNack| mode.
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700186 // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
187 // to rely on NACK only, and therefore are using larger buffers to have time
188 // to wait for retransmissions.
philipel9d3ab612015-12-21 04:12:39 -0800189 void SetNackMode(VCMNackMode mode,
190 int64_t low_rtt_nack_threshold_ms,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000191 int64_t high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000192
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000193 void SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000194 int max_packet_age_to_nack,
195 int max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000196
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000197 // Returns the current NACK mode.
198 VCMNackMode nack_mode() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000199
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000200 // Returns a list of the sequence numbers currently missing.
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700201 std::vector<uint16_t> GetNackList(bool* request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000202
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000203 // Set decode error mode - Should not be changed in the middle of the
204 // session. Changes will not influence frames already in the buffer.
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000205 void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000206 int64_t LastDecodedTimestamp() const;
philipel9d3ab612015-12-21 04:12:39 -0800207 VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
stefan@webrtc.org4c059d82011-10-13 07:35:37 +0000208
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000209 // Used to compute time of complete continuous frames. Returns the timestamps
210 // corresponding to the start and end of the continuous complete buffer.
211 void RenderBufferSize(uint32_t* timestamp_start, uint32_t* timestamp_end);
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000212
pbos@webrtc.org55707692014-12-19 15:45:03 +0000213 void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000214
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000215 private:
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000216 class SequenceNumberLessThan {
217 public:
philipel9d3ab612015-12-21 04:12:39 -0800218 bool operator()(const uint16_t& sequence_number1,
219 const uint16_t& sequence_number2) const {
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +0000220 return IsNewerSequenceNumber(sequence_number2, sequence_number1);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000221 }
222 };
223 typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
224
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000225 // Gets the frame assigned to the timestamp of the packet. May recycle
226 // existing frames if no free frames are available. Returns an error code if
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000227 // failing, or kNoError on success. |frame_list| contains which list the
228 // packet was in, or NULL if it was not in a FrameList (a new frame).
229 VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
230 VCMFrameBuffer** frame,
231 FrameList** frame_list)
232 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000233
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000234 // Returns true if |frame| is continuous in |decoding_state|, not taking
235 // decodable frames into account.
236 bool IsContinuousInState(const VCMFrameBuffer& frame,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000237 const VCMDecodingState& decoding_state) const
238 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000239 // Returns true if |frame| is continuous in the |last_decoded_state_|, taking
240 // all decodable frames into account.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000241 bool IsContinuous(const VCMFrameBuffer& frame) const
242 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700243 // Looks for frames in |incomplete_frames_| which are continuous in the
244 // provided |decoded_state|. Starts the search from the timestamp of
245 // |decoded_state|.
246 void FindAndInsertContinuousFramesWithState(
247 const VCMDecodingState& decoded_state)
248 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000249 // Looks for frames in |incomplete_frames_| which are continuous in
250 // |last_decoded_state_| taking all decodable frames into account. Starts
251 // the search from |new_frame|.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000252 void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
253 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
254 VCMFrameBuffer* NextFrame() const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000255 // Returns true if the NACK list was updated to cover sequence numbers up to
256 // |sequence_number|. If false a key frame is needed to get into a state where
257 // we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000258 bool UpdateNackList(uint16_t sequence_number)
259 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000260 bool TooLargeNackList() const;
261 // Returns true if the NACK list was reduced without problem. If false a key
262 // frame is needed to get into a state where we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000263 bool HandleTooLargeNackList() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
264 bool MissingTooOldPacket(uint16_t latest_sequence_number) const
265 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000266 // Returns true if the too old packets was successfully removed from the NACK
267 // list. If false, a key frame is needed to get into a state where we can
268 // continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000269 bool HandleTooOldPackets(uint16_t latest_sequence_number)
270 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000271 // Drops all packets in the NACK list up until |last_decoded_sequence_number|.
272 void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
273
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000274 void ReleaseFrameIfNotDecoding(VCMFrameBuffer* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000275
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000276 // Gets an empty frame, creating a new frame if necessary (i.e. increases
277 // jitter buffer size).
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000278 VCMFrameBuffer* GetEmptyFrame() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000279
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000280 // Attempts to increase the size of the jitter buffer. Returns true on
281 // success, false otherwise.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000282 bool TryToIncreaseJitterBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000283
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000284 // Recycles oldest frames until a key frame is found. Used if jitter buffer is
285 // completely full. Returns true if a key frame was found.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000286 bool RecycleFramesUntilKeyFrame() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000287
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000288 // Updates the frame statistics.
agalusza@google.comd177c102013-08-08 01:12:33 +0000289 // Counts only complete frames, so decodable incomplete frames will not be
290 // counted.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000291 void CountFrame(const VCMFrameBuffer& frame)
292 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000293
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000294 // Update rolling average of packets per frame.
295 void UpdateAveragePacketsPerFrame(int current_number_packets_);
296
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000297 // Cleans the frame list in the JB from old/empty frames.
298 // Should only be called prior to actual use.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000299 void CleanUpOldOrEmptyFrames() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000300
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000301 // Returns true if |packet| is likely to have been retransmitted.
302 bool IsPacketRetransmitted(const VCMPacket& packet) const;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +0000303
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000304 // The following three functions update the jitter estimate with the
305 // payload size, receive time and RTP timestamp of a frame.
306 void UpdateJitterEstimate(const VCMJitterSample& sample,
307 bool incomplete_frame);
308 void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
309 void UpdateJitterEstimate(int64_t latest_packet_time_ms,
310 uint32_t timestamp,
311 unsigned int frame_size,
312 bool incomplete_frame);
313
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000314 // Returns true if we should wait for retransmissions, false otherwise.
315 bool WaitForRetransmissions();
316
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000317 int NonContinuousOrIncompleteDuration() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000318
319 uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
320
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000321 void UpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000322
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000323 Clock* clock_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000324 // If we are running (have started) or not.
325 bool running_;
326 CriticalSectionWrapper* crit_sect_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000327 // Event to signal when we have a frame ready for decoder.
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000328 rtc::scoped_ptr<EventWrapper> frame_event_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000329 // Number of allocated frames.
330 int max_number_of_frames_;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000331 UnorderedFrameList free_frames_ GUARDED_BY(crit_sect_);
332 FrameList decodable_frames_ GUARDED_BY(crit_sect_);
333 FrameList incomplete_frames_ GUARDED_BY(crit_sect_);
334 VCMDecodingState last_decoded_state_ GUARDED_BY(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000335 bool first_packet_since_reset_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000336
337 // Statistics.
pbos@webrtc.org55707692014-12-19 15:45:03 +0000338 VCMReceiveStatisticsCallback* stats_callback_ GUARDED_BY(crit_sect_);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000339 // Frame counts for each type (key, delta, ...)
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000340 FrameCounts receive_statistics_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000341 // Latest calculated frame rates of incoming stream.
342 unsigned int incoming_frame_rate_;
343 unsigned int incoming_frame_count_;
344 int64_t time_last_incoming_frame_count_;
345 unsigned int incoming_bit_count_;
346 unsigned int incoming_bit_rate_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000347 // Number of frames in a row that have been too old.
348 int num_consecutive_old_frames_;
349 // Number of packets in a row that have been too old.
350 int num_consecutive_old_packets_;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000351 // Number of packets received.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000352 int num_packets_ GUARDED_BY(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000353 // Number of duplicated packets received.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000354 int num_duplicated_packets_ GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000355 // Number of packets discarded by the jitter buffer.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000356 int num_discarded_packets_ GUARDED_BY(crit_sect_);
357 // Time when first packet is received.
358 int64_t time_first_packet_ms_ GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000359
360 // Jitter estimation.
361 // Filter for estimating jitter.
362 VCMJitterEstimator jitter_estimate_;
363 // Calculates network delays used for jitter calculations.
364 VCMInterFrameDelay inter_frame_delay_;
365 VCMJitterSample waiting_for_completion_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000366 int64_t rtt_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000367
368 // NACK and retransmissions.
369 VCMNackMode nack_mode_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000370 int64_t low_rtt_nack_threshold_ms_;
371 int64_t high_rtt_nack_threshold_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000372 // Holds the internal NACK list (the missing sequence numbers).
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000373 SequenceNumberSet missing_sequence_numbers_;
374 uint16_t latest_received_sequence_number_;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000375 size_t max_nack_list_size_;
376 int max_packet_age_to_nack_; // Measured in sequence numbers.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000377 int max_incomplete_time_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000378
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000379 VCMDecodeErrorMode decode_error_mode_;
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000380 // Estimated rolling average of packets per frame
381 float average_packets_per_frame_;
382 // average_packets_per_frame converges fast if we have fewer than this many
383 // frames.
384 int frame_counter_;
henrikg3c089d72015-09-16 05:37:44 -0700385 RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
niklase@google.com470e71d2011-07-07 08:21:25 +0000386};
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000387} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000388
Henrik Kjellander2557b862015-11-18 22:00:21 +0100389#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_