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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
stefan@webrtc.org912981f2012-10-12 07:04:52 +000033enum VCMNackMode {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000034 kNack,
stefan@webrtc.org912981f2012-10-12 07:04:52 +000035 kNoNack
niklase@google.com470e71d2011-07-07 08:21:25 +000036};
37
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:
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -070057 bool operator() (uint32_t timestamp1,
58 uint32_t timestamp2) const {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000059 return IsNewerTimestamp(timestamp2, timestamp1);
60 }
61};
62
agalusza@google.comd818dcb2013-07-29 21:48:11 +000063class FrameList
64 : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000065 public:
66 void InsertFrame(VCMFrameBuffer* frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000067 VCMFrameBuffer* PopFrame(uint32_t timestamp);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000068 VCMFrameBuffer* Front() const;
69 VCMFrameBuffer* Back() const;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000070 int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
71 UnorderedFrameList* free_frames);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000072 void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
73 UnorderedFrameList* free_frames);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000074 void Reset(UnorderedFrameList* free_frames);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000075};
76
asapersson9a4cd872015-10-23 00:27:14 -070077class Vp9SsMap {
78 public:
79 typedef std::map<uint32_t, GofInfoVP9, TimestampLessThan> SsMap;
80 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:
Qiang Chend4cec152015-06-19 09:17:00 -0700110 VCMJitterBuffer(Clock* clock, rtc::scoped_ptr<EventWrapper> event);
111
Wan-Teh Chang6a1ba8c2015-05-26 14:11:41 -0700112 ~VCMJitterBuffer();
niklase@google.com470e71d2011-07-07 08:21:25 +0000113
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000114 // Initializes and starts jitter buffer.
115 void Start();
niklase@google.com470e71d2011-07-07 08:21:25 +0000116
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000117 // Signals all internal events and stops the jitter buffer.
118 void Stop();
niklase@google.com470e71d2011-07-07 08:21:25 +0000119
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000120 // Returns true if the jitter buffer is running.
121 bool Running() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000122
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000123 // Empty the jitter buffer of all its data.
124 void Flush();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000125
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000126 // Get the number of received frames, by type, since the jitter buffer
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000127 // was started.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000128 FrameCounts FrameStatistics() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000129
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000130 // The number of packets discarded by the jitter buffer because the decoder
131 // won't be able to decode them.
132 int num_not_decodable_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000133
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000134 // Gets number of packets received.
135 int num_packets() const;
136
137 // Gets number of duplicated packets received.
138 int num_duplicated_packets() const;
139
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000140 // Gets number of packets discarded by the jitter buffer.
141 int num_discarded_packets() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000142
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000143 // Statistics, Calculate frame and bit rates.
144 void IncomingRateStatistics(unsigned int* framerate,
145 unsigned int* bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000146
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000147 // Checks if the packet sequence will be complete if the next frame would be
148 // grabbed for decoding. That is, if a frame has been lost between the
149 // last decoded frame and the next, or if the next frame is missing one
150 // or more packets.
151 bool CompleteSequenceWithNextFrame();
niklase@google.com470e71d2011-07-07 08:21:25 +0000152
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000153 // Wait |max_wait_time_ms| for a complete frame to arrive.
154 // The function returns true once such a frame is found, its corresponding
155 // timestamp is returned. Otherwise, returns false.
156 bool NextCompleteTimestamp(uint32_t max_wait_time_ms, uint32_t* timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000157
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000158 // Locates a frame for decoding (even an incomplete) without delay.
159 // The function returns true once such a frame is found, its corresponding
160 // timestamp is returned. Otherwise, returns false.
161 bool NextMaybeIncompleteTimestamp(uint32_t* timestamp);
162
163 // Extract frame corresponding to input timestamp.
164 // Frame will be set to a decoding state.
165 VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
niklase@google.com470e71d2011-07-07 08:21:25 +0000166
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000167 // Releases a frame returned from the jitter buffer, should be called when
168 // done with decoding.
169 void ReleaseFrame(VCMEncodedFrame* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000170
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000171 // Returns the time in ms when the latest packet was inserted into the frame.
172 // Retransmitted is set to true if any of the packets belonging to the frame
173 // has been retransmitted.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000174 int64_t LastPacketTime(const VCMEncodedFrame* frame,
175 bool* retransmitted) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000176
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000177 // Inserts a packet into a frame returned from GetFrame().
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000178 // If the return value is <= 0, |frame| is invalidated and the pointer must
179 // be dropped after this function returns.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000180 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
181 bool* retransmitted);
niklase@google.com470e71d2011-07-07 08:21:25 +0000182
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000183 // Returns the estimated jitter in milliseconds.
184 uint32_t EstimatedJitterMs();
niklase@google.com470e71d2011-07-07 08:21:25 +0000185
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000186 // Updates the round-trip time estimate.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000187 void UpdateRtt(int64_t rtt_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000188
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700189 // Set the NACK mode. |high_rtt_nack_threshold_ms| is an RTT threshold in ms
Wan-Teh Changf2912872015-06-05 13:16:45 -0700190 // above which NACK will be disabled if the NACK mode is |kNack|, -1 meaning
191 // that NACK is always enabled in the |kNack| mode.
Wan-Teh Chang603175a2015-05-28 14:10:14 -0700192 // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
193 // to rely on NACK only, and therefore are using larger buffers to have time
194 // to wait for retransmissions.
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000195 void SetNackMode(VCMNackMode mode, int64_t low_rtt_nack_threshold_ms,
196 int64_t high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000197
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000198 void SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000199 int max_packet_age_to_nack,
200 int max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000201
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000202 // Returns the current NACK mode.
203 VCMNackMode nack_mode() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000204
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000205 // Returns a list of the sequence numbers currently missing.
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700206 std::vector<uint16_t> GetNackList(bool* request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000207
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000208 // Set decode error mode - Should not be changed in the middle of the
209 // session. Changes will not influence frames already in the buffer.
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000210 void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000211 int64_t LastDecodedTimestamp() const;
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000212 VCMDecodeErrorMode decode_error_mode() const {return decode_error_mode_;}
stefan@webrtc.org4c059d82011-10-13 07:35:37 +0000213
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000214 // Used to compute time of complete continuous frames. Returns the timestamps
215 // corresponding to the start and end of the continuous complete buffer.
216 void RenderBufferSize(uint32_t* timestamp_start, uint32_t* timestamp_end);
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000217
pbos@webrtc.org55707692014-12-19 15:45:03 +0000218 void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000219
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000220 private:
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000221 class SequenceNumberLessThan {
222 public:
223 bool operator() (const uint16_t& sequence_number1,
224 const uint16_t& sequence_number2) const {
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +0000225 return IsNewerSequenceNumber(sequence_number2, sequence_number1);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000226 }
227 };
228 typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
229
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000230 // Gets the frame assigned to the timestamp of the packet. May recycle
231 // existing frames if no free frames are available. Returns an error code if
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000232 // failing, or kNoError on success. |frame_list| contains which list the
233 // packet was in, or NULL if it was not in a FrameList (a new frame).
234 VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
235 VCMFrameBuffer** frame,
236 FrameList** frame_list)
237 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000238
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000239 // Returns true if |frame| is continuous in |decoding_state|, not taking
240 // decodable frames into account.
241 bool IsContinuousInState(const VCMFrameBuffer& frame,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000242 const VCMDecodingState& decoding_state) const
243 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000244 // Returns true if |frame| is continuous in the |last_decoded_state_|, taking
245 // all decodable frames into account.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000246 bool IsContinuous(const VCMFrameBuffer& frame) const
247 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700248 // Looks for frames in |incomplete_frames_| which are continuous in the
249 // provided |decoded_state|. Starts the search from the timestamp of
250 // |decoded_state|.
251 void FindAndInsertContinuousFramesWithState(
252 const VCMDecodingState& decoded_state)
253 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000254 // Looks for frames in |incomplete_frames_| which are continuous in
255 // |last_decoded_state_| taking all decodable frames into account. Starts
256 // the search from |new_frame|.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000257 void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
258 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
259 VCMFrameBuffer* NextFrame() const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000260 // Returns true if the NACK list was updated to cover sequence numbers up to
261 // |sequence_number|. If false a key frame is needed to get into a state where
262 // we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000263 bool UpdateNackList(uint16_t sequence_number)
264 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000265 bool TooLargeNackList() const;
266 // Returns true if the NACK list was reduced without problem. If false a key
267 // frame is needed to get into a state where we can continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000268 bool HandleTooLargeNackList() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
269 bool MissingTooOldPacket(uint16_t latest_sequence_number) const
270 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000271 // Returns true if the too old packets was successfully removed from the NACK
272 // list. If false, a key frame is needed to get into a state where we can
273 // continue decoding.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000274 bool HandleTooOldPackets(uint16_t latest_sequence_number)
275 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000276 // Drops all packets in the NACK list up until |last_decoded_sequence_number|.
277 void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
278
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000279 void ReleaseFrameIfNotDecoding(VCMFrameBuffer* frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000280
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000281 // Gets an empty frame, creating a new frame if necessary (i.e. increases
282 // jitter buffer size).
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000283 VCMFrameBuffer* GetEmptyFrame() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000284
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000285 // Attempts to increase the size of the jitter buffer. Returns true on
286 // success, false otherwise.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000287 bool TryToIncreaseJitterBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000288
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000289 // Recycles oldest frames until a key frame is found. Used if jitter buffer is
290 // completely full. Returns true if a key frame was found.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000291 bool RecycleFramesUntilKeyFrame() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000292
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000293 // Updates the frame statistics.
agalusza@google.comd177c102013-08-08 01:12:33 +0000294 // Counts only complete frames, so decodable incomplete frames will not be
295 // counted.
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000296 void CountFrame(const VCMFrameBuffer& frame)
297 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000298
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000299 // Update rolling average of packets per frame.
300 void UpdateAveragePacketsPerFrame(int current_number_packets_);
301
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000302 // Cleans the frame list in the JB from old/empty frames.
303 // Should only be called prior to actual use.
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000304 void CleanUpOldOrEmptyFrames() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000305
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000306 // Returns true if |packet| is likely to have been retransmitted.
307 bool IsPacketRetransmitted(const VCMPacket& packet) const;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +0000308
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000309 // The following three functions update the jitter estimate with the
310 // payload size, receive time and RTP timestamp of a frame.
311 void UpdateJitterEstimate(const VCMJitterSample& sample,
312 bool incomplete_frame);
313 void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
314 void UpdateJitterEstimate(int64_t latest_packet_time_ms,
315 uint32_t timestamp,
316 unsigned int frame_size,
317 bool incomplete_frame);
318
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000319 // Returns true if we should wait for retransmissions, false otherwise.
320 bool WaitForRetransmissions();
321
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000322 int NonContinuousOrIncompleteDuration() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000323
324 uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
325
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000326 void UpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000327
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000328 Clock* clock_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000329 // If we are running (have started) or not.
330 bool running_;
331 CriticalSectionWrapper* crit_sect_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000332 // Event to signal when we have a frame ready for decoder.
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000333 rtc::scoped_ptr<EventWrapper> frame_event_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000334 // Number of allocated frames.
335 int max_number_of_frames_;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000336 UnorderedFrameList free_frames_ GUARDED_BY(crit_sect_);
337 FrameList decodable_frames_ GUARDED_BY(crit_sect_);
338 FrameList incomplete_frames_ GUARDED_BY(crit_sect_);
339 VCMDecodingState last_decoded_state_ GUARDED_BY(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000340 bool first_packet_since_reset_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000341
342 // Statistics.
pbos@webrtc.org55707692014-12-19 15:45:03 +0000343 VCMReceiveStatisticsCallback* stats_callback_ GUARDED_BY(crit_sect_);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000344 // Frame counts for each type (key, delta, ...)
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000345 FrameCounts receive_statistics_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000346 // Latest calculated frame rates of incoming stream.
347 unsigned int incoming_frame_rate_;
348 unsigned int incoming_frame_count_;
349 int64_t time_last_incoming_frame_count_;
350 unsigned int incoming_bit_count_;
351 unsigned int incoming_bit_rate_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000352 // Number of frames in a row that have been too old.
353 int num_consecutive_old_frames_;
354 // Number of packets in a row that have been too old.
355 int num_consecutive_old_packets_;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000356 // Number of packets received.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000357 int num_packets_ GUARDED_BY(crit_sect_);
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000358 // Number of duplicated packets received.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000359 int num_duplicated_packets_ GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000360 // Number of packets discarded by the jitter buffer.
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000361 int num_discarded_packets_ GUARDED_BY(crit_sect_);
362 // Time when first packet is received.
363 int64_t time_first_packet_ms_ GUARDED_BY(crit_sect_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000364
365 // Jitter estimation.
366 // Filter for estimating jitter.
367 VCMJitterEstimator jitter_estimate_;
368 // Calculates network delays used for jitter calculations.
369 VCMInterFrameDelay inter_frame_delay_;
370 VCMJitterSample waiting_for_completion_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000371 int64_t rtt_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000372
373 // NACK and retransmissions.
374 VCMNackMode nack_mode_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000375 int64_t low_rtt_nack_threshold_ms_;
376 int64_t high_rtt_nack_threshold_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000377 // Holds the internal NACK list (the missing sequence numbers).
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000378 SequenceNumberSet missing_sequence_numbers_;
379 uint16_t latest_received_sequence_number_;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000380 size_t max_nack_list_size_;
381 int max_packet_age_to_nack_; // Measured in sequence numbers.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000382 int max_incomplete_time_ms_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000383
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000384 VCMDecodeErrorMode decode_error_mode_;
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000385 // Estimated rolling average of packets per frame
386 float average_packets_per_frame_;
387 // average_packets_per_frame converges fast if we have fewer than this many
388 // frames.
389 int frame_counter_;
henrikg3c089d72015-09-16 05:37:44 -0700390 RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
niklase@google.com470e71d2011-07-07 08:21:25 +0000391};
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000392} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000393
Henrik Kjellander2557b862015-11-18 22:00:21 +0100394#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_