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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 */
Henrik Kjellander2557b862015-11-18 22:00:21 +010010#include "webrtc/modules/video_coding/jitter_buffer.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000011
pbos@webrtc.org12dc1a32013-08-05 16:22:53 +000012#include <assert.h>
13
stefan@webrtc.org29794612012-02-08 08:58:55 +000014#include <algorithm>
agalusza@google.comd818dcb2013-07-29 21:48:11 +000015#include <utility>
stefan@webrtc.org29794612012-02-08 08:58:55 +000016
asapersson9a4cd872015-10-23 00:27:14 -070017#include "webrtc/base/checks.h"
pbos854e84c2015-11-16 16:39:06 -080018#include "webrtc/base/logging.h"
tommie4f96502015-10-20 23:00:48 -070019#include "webrtc/base/trace_event.h"
Henrik Kjellanderff761fb2015-11-04 08:31:52 +010020#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010021#include "webrtc/modules/video_coding/include/video_coding.h"
22#include "webrtc/modules/video_coding/frame_buffer.h"
23#include "webrtc/modules/video_coding/inter_frame_delay.h"
24#include "webrtc/modules/video_coding/internal_defines.h"
25#include "webrtc/modules/video_coding/jitter_buffer_common.h"
26#include "webrtc/modules/video_coding/jitter_estimator.h"
27#include "webrtc/modules/video_coding/packet.h"
Henrik Kjellander98f53512015-10-28 18:17:40 +010028#include "webrtc/system_wrappers/include/clock.h"
29#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
30#include "webrtc/system_wrappers/include/event_wrapper.h"
Henrik Kjellander98f53512015-10-28 18:17:40 +010031#include "webrtc/system_wrappers/include/metrics.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000032
niklase@google.com470e71d2011-07-07 08:21:25 +000033namespace webrtc {
34
asapersson9a4cd872015-10-23 00:27:14 -070035// Interval for updating SS data.
36static const uint32_t kSsCleanupIntervalSec = 60;
37
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +000038// Use this rtt if no value has been reported.
pkasting@chromium.org16825b12015-01-12 21:51:21 +000039static const int64_t kDefaultRtt = 200;
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +000040
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000041typedef std::pair<uint32_t, VCMFrameBuffer*> FrameListPair;
niklase@google.com470e71d2011-07-07 08:21:25 +000042
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000043bool IsKeyFrame(FrameListPair pair) {
44 return pair.second->FrameType() == kVideoFrameKey;
45}
stefan@webrtc.org29794612012-02-08 08:58:55 +000046
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000047bool HasNonEmptyState(FrameListPair pair) {
48 return pair.second->GetState() != kStateEmpty;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000049}
50
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000051void FrameList::InsertFrame(VCMFrameBuffer* frame) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000052 insert(rbegin().base(), FrameListPair(frame->TimeStamp(), frame));
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000053}
54
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000055VCMFrameBuffer* FrameList::PopFrame(uint32_t timestamp) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000056 FrameList::iterator it = find(timestamp);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000057 if (it == end())
58 return NULL;
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000059 VCMFrameBuffer* frame = it->second;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000060 erase(it);
61 return frame;
62}
63
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000064VCMFrameBuffer* FrameList::Front() const {
65 return begin()->second;
66}
67
68VCMFrameBuffer* FrameList::Back() const {
69 return rbegin()->second;
70}
71
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000072int FrameList::RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
73 UnorderedFrameList* free_frames) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000074 int drop_count = 0;
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000075 FrameList::iterator it = begin();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000076 while (!empty()) {
77 // Throw at least one frame.
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000078 it->second->Reset();
79 free_frames->push_back(it->second);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000080 erase(it++);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000081 ++drop_count;
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000082 if (it != end() && it->second->FrameType() == kVideoFrameKey) {
83 *key_frame_it = it;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000084 return drop_count;
85 }
86 }
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000087 *key_frame_it = end();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000088 return drop_count;
89}
90
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000091void FrameList::CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +000092 UnorderedFrameList* free_frames) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000093 while (!empty()) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +000094 VCMFrameBuffer* oldest_frame = Front();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +000095 bool remove_frame = false;
96 if (oldest_frame->GetState() == kStateEmpty && size() > 1) {
97 // This frame is empty, try to update the last decoded state and drop it
98 // if successful.
99 remove_frame = decoding_state->UpdateEmptyFrame(oldest_frame);
100 } else {
101 remove_frame = decoding_state->IsOldFrame(oldest_frame);
102 }
103 if (!remove_frame) {
104 break;
105 }
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000106 free_frames->push_back(oldest_frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000107 TRACE_EVENT_INSTANT1("webrtc", "JB::OldOrEmptyFrameDropped", "timestamp",
108 oldest_frame->TimeStamp());
109 erase(begin());
110 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000111}
112
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000113void FrameList::Reset(UnorderedFrameList* free_frames) {
114 while (!empty()) {
115 begin()->second->Reset();
116 free_frames->push_back(begin()->second);
117 erase(begin());
118 }
119}
120
asapersson9a4cd872015-10-23 00:27:14 -0700121bool Vp9SsMap::Insert(const VCMPacket& packet) {
122 if (!packet.codecSpecificHeader.codecHeader.VP9.ss_data_available)
123 return false;
124
125 ss_map_[packet.timestamp] = packet.codecSpecificHeader.codecHeader.VP9.gof;
126 return true;
127}
128
129void Vp9SsMap::Reset() {
130 ss_map_.clear();
131}
132
133bool Vp9SsMap::Find(uint32_t timestamp, SsMap::iterator* it_out) {
134 bool found = false;
135 for (SsMap::iterator it = ss_map_.begin(); it != ss_map_.end(); ++it) {
136 if (it->first == timestamp || IsNewerTimestamp(timestamp, it->first)) {
137 *it_out = it;
138 found = true;
139 }
140 }
141 return found;
142}
143
144void Vp9SsMap::RemoveOld(uint32_t timestamp) {
145 if (!TimeForCleanup(timestamp))
146 return;
147
148 SsMap::iterator it;
149 if (!Find(timestamp, &it))
150 return;
151
152 ss_map_.erase(ss_map_.begin(), it);
153 AdvanceFront(timestamp);
154}
155
156bool Vp9SsMap::TimeForCleanup(uint32_t timestamp) const {
157 if (ss_map_.empty() || !IsNewerTimestamp(timestamp, ss_map_.begin()->first))
158 return false;
159
160 uint32_t diff = timestamp - ss_map_.begin()->first;
161 return diff / kVideoPayloadTypeFrequency >= kSsCleanupIntervalSec;
162}
163
164void Vp9SsMap::AdvanceFront(uint32_t timestamp) {
165 RTC_DCHECK(!ss_map_.empty());
166 GofInfoVP9 gof = ss_map_.begin()->second;
167 ss_map_.erase(ss_map_.begin());
168 ss_map_[timestamp] = gof;
169}
170
asaperssonc253a1c2015-11-06 00:12:01 -0800171// TODO(asapersson): Update according to updates in RTP payload profile.
asapersson9a4cd872015-10-23 00:27:14 -0700172bool Vp9SsMap::UpdatePacket(VCMPacket* packet) {
173 uint8_t gof_idx = packet->codecSpecificHeader.codecHeader.VP9.gof_idx;
174 if (gof_idx == kNoGofIdx)
175 return false; // No update needed.
176
177 SsMap::iterator it;
178 if (!Find(packet->timestamp, &it))
179 return false; // Corresponding SS not yet received.
180
181 if (gof_idx >= it->second.num_frames_in_gof)
182 return false; // Assume corresponding SS not yet received.
183
184 RTPVideoHeaderVP9* vp9 = &packet->codecSpecificHeader.codecHeader.VP9;
185 vp9->temporal_idx = it->second.temporal_idx[gof_idx];
186 vp9->temporal_up_switch = it->second.temporal_up_switch[gof_idx];
187
188 // TODO(asapersson): Set vp9.ref_picture_id[i] and add usage.
189 vp9->num_ref_pics = it->second.num_ref_pics[gof_idx];
asaperssonc253a1c2015-11-06 00:12:01 -0800190 for (uint8_t i = 0; i < it->second.num_ref_pics[gof_idx]; ++i) {
asapersson9a4cd872015-10-23 00:27:14 -0700191 vp9->pid_diff[i] = it->second.pid_diff[gof_idx][i];
192 }
193 return true;
194}
195
196void Vp9SsMap::UpdateFrames(FrameList* frames) {
197 for (const auto& frame_it : *frames) {
198 uint8_t gof_idx =
199 frame_it.second->CodecSpecific()->codecSpecific.VP9.gof_idx;
200 if (gof_idx == kNoGofIdx) {
201 continue;
202 }
203 SsMap::iterator ss_it;
204 if (Find(frame_it.second->TimeStamp(), &ss_it)) {
205 if (gof_idx >= ss_it->second.num_frames_in_gof) {
206 continue; // Assume corresponding SS not yet received.
207 }
208 frame_it.second->SetGofInfo(ss_it->second, gof_idx);
209 }
210 }
211}
212
Qiang Chend4cec152015-06-19 09:17:00 -0700213VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
214 rtc::scoped_ptr<EventWrapper> event)
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000215 : clock_(clock),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000216 running_(false),
217 crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
Qiang Chend4cec152015-06-19 09:17:00 -0700218 frame_event_(event.Pass()),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000219 max_number_of_frames_(kStartNumberOfFrames),
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000220 free_frames_(),
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000221 decodable_frames_(),
222 incomplete_frames_(),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000223 last_decoded_state_(),
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000224 first_packet_since_reset_(true),
pbos@webrtc.org55707692014-12-19 15:45:03 +0000225 stats_callback_(NULL),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000226 incoming_frame_rate_(0),
227 incoming_frame_count_(0),
228 time_last_incoming_frame_count_(0),
229 incoming_bit_count_(0),
230 incoming_bit_rate_(0),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000231 num_consecutive_old_packets_(0),
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000232 num_packets_(0),
233 num_duplicated_packets_(0),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000234 num_discarded_packets_(0),
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000235 time_first_packet_ms_(0),
sprang@webrtc.org70e2d112014-09-24 14:06:56 +0000236 jitter_estimate_(clock),
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000237 inter_frame_delay_(clock_->TimeInMilliseconds()),
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +0000238 rtt_ms_(kDefaultRtt),
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000239 nack_mode_(kNoNack),
240 low_rtt_nack_threshold_ms_(-1),
241 high_rtt_nack_threshold_ms_(-1),
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000242 missing_sequence_numbers_(SequenceNumberLessThan()),
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000243 max_nack_list_size_(0),
244 max_packet_age_to_nack_(0),
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000245 max_incomplete_time_ms_(0),
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000246 decode_error_mode_(kNoErrors),
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000247 average_packets_per_frame_(0.0f),
248 frame_counter_(0) {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000249 for (int i = 0; i < kStartNumberOfFrames; i++)
250 free_frames_.push_back(new VCMFrameBuffer());
niklase@google.com470e71d2011-07-07 08:21:25 +0000251}
252
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000253VCMJitterBuffer::~VCMJitterBuffer() {
254 Stop();
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000255 for (UnorderedFrameList::iterator it = free_frames_.begin();
256 it != free_frames_.end(); ++it) {
257 delete *it;
258 }
259 for (FrameList::iterator it = incomplete_frames_.begin();
260 it != incomplete_frames_.end(); ++it) {
261 delete it->second;
262 }
263 for (FrameList::iterator it = decodable_frames_.begin();
264 it != decodable_frames_.end(); ++it) {
265 delete it->second;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000266 }
267 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000268}
269
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000270void VCMJitterBuffer::UpdateHistograms() {
Ã…sa Perssona96f02b2015-04-24 08:52:11 +0200271 if (num_packets_ <= 0 || !running_) {
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000272 return;
273 }
274 int64_t elapsed_sec =
275 (clock_->TimeInMilliseconds() - time_first_packet_ms_) / 1000;
276 if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
277 return;
278 }
279
280 RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
281 num_discarded_packets_ * 100 / num_packets_);
282 RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
283 num_duplicated_packets_ * 100 / num_packets_);
284
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000285 int total_frames =
286 receive_statistics_.key_frames + receive_statistics_.delta_frames;
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000287 if (total_frames > 0) {
288 RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.CompleteFramesReceivedPerSecond",
289 static_cast<int>((total_frames / elapsed_sec) + 0.5f));
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000290 RTC_HISTOGRAM_COUNTS_1000(
291 "WebRTC.Video.KeyFramesReceivedInPermille",
292 static_cast<int>(
293 (receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000294 }
295}
296
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000297void VCMJitterBuffer::Start() {
298 CriticalSectionScoped cs(crit_sect_);
299 running_ = true;
300 incoming_frame_count_ = 0;
301 incoming_frame_rate_ = 0;
302 incoming_bit_count_ = 0;
303 incoming_bit_rate_ = 0;
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000304 time_last_incoming_frame_count_ = clock_->TimeInMilliseconds();
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000305 receive_statistics_ = FrameCounts();
niklase@google.com470e71d2011-07-07 08:21:25 +0000306
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000307 num_consecutive_old_packets_ = 0;
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000308 num_packets_ = 0;
309 num_duplicated_packets_ = 0;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000310 num_discarded_packets_ = 0;
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000311 time_first_packet_ms_ = 0;
niklase@google.com470e71d2011-07-07 08:21:25 +0000312
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000313 // Start in a non-signaled state.
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000314 waiting_for_completion_.frame_size = 0;
315 waiting_for_completion_.timestamp = 0;
316 waiting_for_completion_.latest_packet_time = -1;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000317 first_packet_since_reset_ = true;
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +0000318 rtt_ms_ = kDefaultRtt;
mikhal@webrtc.org8392cd92013-04-25 21:30:50 +0000319 last_decoded_state_.Reset();
niklase@google.com470e71d2011-07-07 08:21:25 +0000320}
321
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000322void VCMJitterBuffer::Stop() {
323 crit_sect_->Enter();
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000324 UpdateHistograms();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000325 running_ = false;
326 last_decoded_state_.Reset();
asaperssona9455ab2015-07-31 06:10:09 -0700327
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000328 // Make sure all frames are free and reset.
329 for (FrameList::iterator it = decodable_frames_.begin();
330 it != decodable_frames_.end(); ++it) {
331 free_frames_.push_back(it->second);
332 }
333 for (FrameList::iterator it = incomplete_frames_.begin();
334 it != incomplete_frames_.end(); ++it) {
335 free_frames_.push_back(it->second);
336 }
337 for (UnorderedFrameList::iterator it = free_frames_.begin();
338 it != free_frames_.end(); ++it) {
339 (*it)->Reset();
340 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000341 decodable_frames_.clear();
342 incomplete_frames_.clear();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000343 crit_sect_->Leave();
344 // Make sure we wake up any threads waiting on these events.
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +0000345 frame_event_->Set();
niklase@google.com470e71d2011-07-07 08:21:25 +0000346}
347
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000348bool VCMJitterBuffer::Running() const {
349 CriticalSectionScoped cs(crit_sect_);
350 return running_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000351}
352
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000353void VCMJitterBuffer::Flush() {
354 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000355 decodable_frames_.Reset(&free_frames_);
356 incomplete_frames_.Reset(&free_frames_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000357 last_decoded_state_.Reset(); // TODO(mikhal): sync reset.
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000358 num_consecutive_old_packets_ = 0;
359 // Also reset the jitter and delay estimates
360 jitter_estimate_.Reset();
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000361 inter_frame_delay_.Reset(clock_->TimeInMilliseconds());
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000362 waiting_for_completion_.frame_size = 0;
363 waiting_for_completion_.timestamp = 0;
364 waiting_for_completion_.latest_packet_time = -1;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000365 first_packet_since_reset_ = true;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000366 missing_sequence_numbers_.clear();
niklase@google.com470e71d2011-07-07 08:21:25 +0000367}
368
niklase@google.com470e71d2011-07-07 08:21:25 +0000369// Get received key and delta frames
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000370FrameCounts VCMJitterBuffer::FrameStatistics() const {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000371 CriticalSectionScoped cs(crit_sect_);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000372 return receive_statistics_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000373}
374
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000375int VCMJitterBuffer::num_packets() const {
376 CriticalSectionScoped cs(crit_sect_);
377 return num_packets_;
378}
379
380int VCMJitterBuffer::num_duplicated_packets() const {
381 CriticalSectionScoped cs(crit_sect_);
382 return num_duplicated_packets_;
383}
384
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000385int VCMJitterBuffer::num_discarded_packets() const {
386 CriticalSectionScoped cs(crit_sect_);
387 return num_discarded_packets_;
388}
389
390// Calculate framerate and bitrate.
391void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
392 unsigned int* bitrate) {
393 assert(framerate);
394 assert(bitrate);
395 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000396 const int64_t now = clock_->TimeInMilliseconds();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000397 int64_t diff = now - time_last_incoming_frame_count_;
398 if (diff < 1000 && incoming_frame_rate_ > 0 && incoming_bit_rate_ > 0) {
399 // Make sure we report something even though less than
400 // 1 second has passed since last update.
401 *framerate = incoming_frame_rate_;
402 *bitrate = incoming_bit_rate_;
403 } else if (incoming_frame_count_ != 0) {
404 // We have received frame(s) since last call to this function
405
406 // Prepare calculations
407 if (diff <= 0) {
408 diff = 1;
niklase@google.com470e71d2011-07-07 08:21:25 +0000409 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000410 // we add 0.5f for rounding
411 float rate = 0.5f + ((incoming_frame_count_ * 1000.0f) / diff);
412 if (rate < 1.0f) {
413 rate = 1.0f;
414 }
415
416 // Calculate frame rate
417 // Let r be rate.
418 // r(0) = 1000*framecount/delta_time.
419 // (I.e. frames per second since last calculation.)
420 // frame_rate = r(0)/2 + r(-1)/2
421 // (I.e. fr/s average this and the previous calculation.)
422 *framerate = (incoming_frame_rate_ + static_cast<unsigned int>(rate)) / 2;
423 incoming_frame_rate_ = static_cast<unsigned int>(rate);
424
425 // Calculate bit rate
426 if (incoming_bit_count_ == 0) {
427 *bitrate = 0;
428 } else {
429 *bitrate = 10 * ((100 * incoming_bit_count_) /
430 static_cast<unsigned int>(diff));
431 }
432 incoming_bit_rate_ = *bitrate;
433
434 // Reset count
435 incoming_frame_count_ = 0;
436 incoming_bit_count_ = 0;
437 time_last_incoming_frame_count_ = now;
438
439 } else {
440 // No frames since last call
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000441 time_last_incoming_frame_count_ = clock_->TimeInMilliseconds();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000442 *framerate = 0;
stefan@webrtc.org49806792013-04-30 22:05:07 +0000443 *bitrate = 0;
444 incoming_frame_rate_ = 0;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000445 incoming_bit_rate_ = 0;
446 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000447}
448
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000449// Answers the question:
450// Will the packet sequence be complete if the next frame is grabbed for
451// decoding right now? That is, have we lost a frame between the last decoded
452// frame and the next, or is the next
453// frame missing one or more packets?
454bool VCMJitterBuffer::CompleteSequenceWithNextFrame() {
455 CriticalSectionScoped cs(crit_sect_);
456 // Finding oldest frame ready for decoder, check sequence number and size
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000457 CleanUpOldOrEmptyFrames();
agalusza@google.comd177c102013-08-08 01:12:33 +0000458 if (!decodable_frames_.empty()) {
459 if (decodable_frames_.Front()->GetState() == kStateComplete) {
460 return true;
461 }
462 } else if (incomplete_frames_.size() <= 1) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000463 // Frame not ready to be decoded.
464 return true;
465 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000466 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000467}
468
469// Returns immediately or a |max_wait_time_ms| ms event hang waiting for a
470// complete frame, |max_wait_time_ms| decided by caller.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000471bool VCMJitterBuffer::NextCompleteTimestamp(
472 uint32_t max_wait_time_ms, uint32_t* timestamp) {
mikhal@webrtc.orgc1f243f2013-04-22 22:24:38 +0000473 crit_sect_->Enter();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000474 if (!running_) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000475 crit_sect_->Leave();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000476 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000477 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000478 CleanUpOldOrEmptyFrames();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000479
agalusza@google.comd177c102013-08-08 01:12:33 +0000480 if (decodable_frames_.empty() ||
481 decodable_frames_.Front()->GetState() != kStateComplete) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000482 const int64_t end_wait_time_ms = clock_->TimeInMilliseconds() +
483 max_wait_time_ms;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000484 int64_t wait_time_ms = max_wait_time_ms;
485 while (wait_time_ms > 0) {
486 crit_sect_->Leave();
487 const EventTypeWrapper ret =
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +0000488 frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000489 crit_sect_->Enter();
490 if (ret == kEventSignaled) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000491 // Are we shutting down the jitter buffer?
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000492 if (!running_) {
493 crit_sect_->Leave();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000494 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000495 }
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000496 // Finding oldest frame ready for decoder.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000497 CleanUpOldOrEmptyFrames();
agalusza@google.comd177c102013-08-08 01:12:33 +0000498 if (decodable_frames_.empty() ||
499 decodable_frames_.Front()->GetState() != kStateComplete) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000500 wait_time_ms = end_wait_time_ms - clock_->TimeInMilliseconds();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000501 } else {
502 break;
503 }
504 } else {
mikhal@webrtc.org9c7685f2013-05-07 16:07:52 +0000505 break;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000506 }
507 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000508 }
agalusza@google.comd177c102013-08-08 01:12:33 +0000509 if (decodable_frames_.empty() ||
510 decodable_frames_.Front()->GetState() != kStateComplete) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000511 crit_sect_->Leave();
512 return false;
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000513 }
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000514 *timestamp = decodable_frames_.Front()->TimeStamp();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000515 crit_sect_->Leave();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000516 return true;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000517}
518
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000519bool VCMJitterBuffer::NextMaybeIncompleteTimestamp(uint32_t* timestamp) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000520 CriticalSectionScoped cs(crit_sect_);
521 if (!running_) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000522 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000523 }
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000524 if (decode_error_mode_ == kNoErrors) {
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000525 // No point to continue, as we are not decoding with errors.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000526 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000527 }
528
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000529 CleanUpOldOrEmptyFrames();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000530
agalusza@google.comd177c102013-08-08 01:12:33 +0000531 if (decodable_frames_.empty()) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000532 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000533 }
agalusza@google.comd177c102013-08-08 01:12:33 +0000534 VCMFrameBuffer* oldest_frame = decodable_frames_.Front();
agalusza@google.comd177c102013-08-08 01:12:33 +0000535 // If we have exactly one frame in the buffer, release it only if it is
mikhal@webrtc.orgb2c28c32013-08-23 21:54:50 +0000536 // complete. We know decodable_frames_ is not empty due to the previous
agalusza@google.comd177c102013-08-08 01:12:33 +0000537 // check.
538 if (decodable_frames_.size() == 1 && incomplete_frames_.empty()
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000539 && oldest_frame->GetState() != kStateComplete) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000540 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000541 }
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000542
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000543 *timestamp = oldest_frame->TimeStamp();
544 return true;
545}
546
547VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000548 CriticalSectionScoped cs(crit_sect_);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000549 if (!running_) {
550 return NULL;
551 }
552 // Extract the frame with the desired timestamp.
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000553 VCMFrameBuffer* frame = decodable_frames_.PopFrame(timestamp);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000554 bool continuous = true;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000555 if (!frame) {
556 frame = incomplete_frames_.PopFrame(timestamp);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000557 if (frame)
558 continuous = last_decoded_state_.ContinuousFrame(frame);
559 else
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000560 return NULL;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000561 }
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000562 TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", timestamp, "Extract");
mikhal@webrtc.orgcb20a5b2013-05-15 17:10:44 +0000563 // Frame pulled out from jitter buffer, update the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000564 const bool retransmitted = (frame->GetNackCount() > 0);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000565 if (retransmitted) {
566 jitter_estimate_.FrameNacked();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000567 } else if (frame->Length() > 0) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000568 // Ignore retransmitted and empty frames.
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000569 if (waiting_for_completion_.latest_packet_time >= 0) {
570 UpdateJitterEstimate(waiting_for_completion_, true);
571 }
mikhal@webrtc.orgcb20a5b2013-05-15 17:10:44 +0000572 if (frame->GetState() == kStateComplete) {
573 UpdateJitterEstimate(*frame, false);
574 } else {
575 // Wait for this one to get complete.
576 waiting_for_completion_.frame_size = frame->Length();
577 waiting_for_completion_.latest_packet_time =
578 frame->LatestPacketTimeMs();
579 waiting_for_completion_.timestamp = frame->TimeStamp();
580 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000581 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000582
583 // The state must be changed to decoding before cleaning up zero sized
584 // frames to avoid empty frames being cleaned up and then given to the
mikhal@webrtc.orgcb20a5b2013-05-15 17:10:44 +0000585 // decoder. Propagates the missing_frame bit.
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000586 frame->PrepareForDecode(continuous);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000587
mikhal@webrtc.orgcb20a5b2013-05-15 17:10:44 +0000588 // We have a frame - update the last decoded state and nack list.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000589 last_decoded_state_.SetState(frame);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000590 DropPacketsFromNackList(last_decoded_state_.sequence_num());
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000591
592 if ((*frame).IsSessionComplete())
593 UpdateAveragePacketsPerFrame(frame->NumPackets());
594
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000595 return frame;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000596}
597
598// Release frame when done with decoding. Should never be used to release
599// frames from within the jitter buffer.
600void VCMJitterBuffer::ReleaseFrame(VCMEncodedFrame* frame) {
601 CriticalSectionScoped cs(crit_sect_);
602 VCMFrameBuffer* frame_buffer = static_cast<VCMFrameBuffer*>(frame);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000603 if (frame_buffer) {
604 free_frames_.push_back(frame_buffer);
605 }
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000606}
607
niklase@google.com470e71d2011-07-07 08:21:25 +0000608// Gets frame to use for this timestamp. If no match, get empty frame.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000609VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet,
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000610 VCMFrameBuffer** frame,
611 FrameList** frame_list) {
612 *frame = incomplete_frames_.PopFrame(packet.timestamp);
613 if (*frame != NULL) {
614 *frame_list = &incomplete_frames_;
615 return kNoError;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000616 }
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000617 *frame = decodable_frames_.PopFrame(packet.timestamp);
618 if (*frame != NULL) {
619 *frame_list = &decodable_frames_;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000620 return kNoError;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000621 }
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000622
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000623 *frame_list = NULL;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000624 // No match, return empty frame.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000625 *frame = GetEmptyFrame();
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000626 if (*frame == NULL) {
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000627 // No free frame! Try to reclaim some...
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000628 LOG(LS_WARNING) << "Unable to get empty frame; Recycling.";
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000629 bool found_key_frame = RecycleFramesUntilKeyFrame();
630 *frame = GetEmptyFrame();
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000631 assert(*frame);
632 if (!found_key_frame) {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000633 free_frames_.push_back(*frame);
634 return kFlushIndicator;
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000635 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000636 }
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000637 (*frame)->Reset();
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000638 return kNoError;
niklase@google.com470e71d2011-07-07 08:21:25 +0000639}
640
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000641int64_t VCMJitterBuffer::LastPacketTime(const VCMEncodedFrame* frame,
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000642 bool* retransmitted) const {
643 assert(retransmitted);
644 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000645 const VCMFrameBuffer* frame_buffer =
646 static_cast<const VCMFrameBuffer*>(frame);
647 *retransmitted = (frame_buffer->GetNackCount() > 0);
648 return frame_buffer->LatestPacketTimeMs();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000649}
650
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000651VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
652 bool* retransmitted) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000653 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000654
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000655 ++num_packets_;
asapersson@webrtc.org83b52002014-11-28 10:17:13 +0000656 if (num_packets_ == 1) {
657 time_first_packet_ms_ = clock_->TimeInMilliseconds();
658 }
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000659 // Does this packet belong to an old frame?
660 if (last_decoded_state_.IsOldPacket(&packet)) {
661 // Account only for media packets.
662 if (packet.sizeBytes > 0) {
663 num_discarded_packets_++;
664 num_consecutive_old_packets_++;
pbos@webrtc.org55707692014-12-19 15:45:03 +0000665 if (stats_callback_ != NULL)
666 stats_callback_->OnDiscardedPacketsUpdated(num_discarded_packets_);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000667 }
668 // Update last decoded sequence number if the packet arrived late and
669 // belongs to a frame with a timestamp equal to the last decoded
670 // timestamp.
671 last_decoded_state_.UpdateOldPacket(&packet);
672 DropPacketsFromNackList(last_decoded_state_.sequence_num());
673
Noah Richardse4cb4e92015-05-22 14:03:00 -0700674 // Also see if this old packet made more incomplete frames continuous.
675 FindAndInsertContinuousFramesWithState(last_decoded_state_);
676
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000677 if (num_consecutive_old_packets_ > kMaxConsecutiveOldPackets) {
678 LOG(LS_WARNING)
679 << num_consecutive_old_packets_
680 << " consecutive old packets received. Flushing the jitter buffer.";
681 Flush();
682 return kFlushIndicator;
683 }
684 return kOldPacket;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000685 }
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000686
asapersson9a4cd872015-10-23 00:27:14 -0700687 num_consecutive_old_packets_ = 0;
688
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000689 VCMFrameBuffer* frame;
690 FrameList* frame_list;
691 const VCMFrameBufferEnum error = GetFrame(packet, &frame, &frame_list);
692 if (error != kNoError)
693 return error;
694
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000695 int64_t now_ms = clock_->TimeInMilliseconds();
696 // We are keeping track of the first and latest seq numbers, and
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000697 // the number of wraps to be able to calculate how many packets we expect.
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000698 if (first_packet_since_reset_) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000699 // Now it's time to start estimating jitter
700 // reset the delay estimate.
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000701 inter_frame_delay_.Reset(now_ms);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000702 }
703
704 // Empty packets may bias the jitter estimate (lacking size component),
705 // therefore don't let empty packet trigger the following updates:
pbos22993e12015-10-19 02:39:06 -0700706 if (packet.frameType != kEmptyFrame) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000707 if (waiting_for_completion_.timestamp == packet.timestamp) {
708 // This can get bad if we have a lot of duplicate packets,
709 // we will then count some packet multiple times.
710 waiting_for_completion_.frame_size += packet.sizeBytes;
711 waiting_for_completion_.latest_packet_time = now_ms;
712 } else if (waiting_for_completion_.latest_packet_time >= 0 &&
713 waiting_for_completion_.latest_packet_time + 2000 <= now_ms) {
714 // A packet should never be more than two seconds late
715 UpdateJitterEstimate(waiting_for_completion_, true);
716 waiting_for_completion_.latest_packet_time = -1;
717 waiting_for_completion_.frame_size = 0;
718 waiting_for_completion_.timestamp = 0;
719 }
720 }
721
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000722 VCMFrameBufferStateEnum previous_state = frame->GetState();
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000723 // Insert packet.
agalusza@google.comd818dcb2013-07-29 21:48:11 +0000724 FrameData frame_data;
725 frame_data.rtt_ms = rtt_ms_;
726 frame_data.rolling_average_packets_per_frame = average_packets_per_frame_;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000727 VCMFrameBufferEnum buffer_state =
728 frame->InsertPacket(packet, now_ms, decode_error_mode_, frame_data);
729
730 if (previous_state != kStateComplete) {
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000731 TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", frame->TimeStamp(),
732 "timestamp", frame->TimeStamp());
733 }
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000734
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000735 if (buffer_state > 0) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000736 incoming_bit_count_ += packet.sizeBytes << 3;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000737 if (first_packet_since_reset_) {
738 latest_received_sequence_number_ = packet.seqNum;
739 first_packet_since_reset_ = false;
740 } else {
741 if (IsPacketRetransmitted(packet)) {
742 frame->IncrementNackCount();
743 }
pbos@webrtc.orgebb467f2014-05-19 15:28:02 +0000744 if (!UpdateNackList(packet.seqNum) &&
745 packet.frameType != kVideoFrameKey) {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000746 buffer_state = kFlushIndicator;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000747 }
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000748
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000749 latest_received_sequence_number_ = LatestSequenceNumber(
750 latest_received_sequence_number_, packet.seqNum);
751 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000752 }
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000753
754 // Is the frame already in the decodable list?
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000755 bool continuous = IsContinuous(*frame);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000756 switch (buffer_state) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000757 case kGeneralError:
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000758 case kTimeStampError:
759 case kSizeError: {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000760 free_frames_.push_back(frame);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000761 break;
762 }
763 case kCompleteSession: {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000764 if (previous_state != kStateDecodable &&
765 previous_state != kStateComplete) {
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000766 CountFrame(*frame);
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000767 if (continuous) {
768 // Signal that we have a complete session.
769 frame_event_->Set();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000770 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000771 }
kjellander@webrtc.org7d2b6a92015-01-28 18:37:58 +0000772 FALLTHROUGH();
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000773 }
774 // Note: There is no break here - continuing to kDecodableSession.
775 case kDecodableSession: {
776 *retransmitted = (frame->GetNackCount() > 0);
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000777 if (continuous) {
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000778 decodable_frames_.InsertFrame(frame);
779 FindAndInsertContinuousFrames(*frame);
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000780 } else {
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000781 incomplete_frames_.InsertFrame(frame);
782 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000783 break;
784 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000785 case kIncomplete: {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000786 if (frame->GetState() == kStateEmpty &&
787 last_decoded_state_.UpdateEmptyFrame(frame)) {
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +0000788 free_frames_.push_back(frame);
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000789 return kNoError;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000790 } else {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000791 incomplete_frames_.InsertFrame(frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000792 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000793 break;
794 }
795 case kNoError:
mikhal@webrtc.orgf31a47a2013-08-26 17:10:11 +0000796 case kOutOfBoundsPacket:
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000797 case kDuplicatePacket: {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000798 // Put back the frame where it came from.
799 if (frame_list != NULL) {
800 frame_list->InsertFrame(frame);
801 } else {
802 free_frames_.push_back(frame);
803 }
asapersson@webrtc.org96dc6852014-11-03 14:40:38 +0000804 ++num_duplicated_packets_;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000805 break;
806 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000807 case kFlushIndicator:
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000808 free_frames_.push_back(frame);
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000809 return kFlushIndicator;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000810 default: assert(false);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000811 }
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000812 return buffer_state;
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000813}
814
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000815bool VCMJitterBuffer::IsContinuousInState(const VCMFrameBuffer& frame,
816 const VCMDecodingState& decoding_state) const {
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +0000817 if (decode_error_mode_ == kWithErrors)
818 return true;
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000819 // Is this frame (complete or decodable) and continuous?
820 // kStateDecodable will never be set when decode_error_mode_ is false
821 // as SessionInfo determines this state based on the error mode (and frame
822 // completeness).
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000823 return (frame.GetState() == kStateComplete ||
824 frame.GetState() == kStateDecodable) &&
825 decoding_state.ContinuousFrame(&frame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000826}
827
828bool VCMJitterBuffer::IsContinuous(const VCMFrameBuffer& frame) const {
829 if (IsContinuousInState(frame, last_decoded_state_)) {
830 return true;
831 }
832 VCMDecodingState decoding_state;
833 decoding_state.CopyFrom(last_decoded_state_);
834 for (FrameList::const_iterator it = decodable_frames_.begin();
835 it != decodable_frames_.end(); ++it) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000836 VCMFrameBuffer* decodable_frame = it->second;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000837 if (IsNewerTimestamp(decodable_frame->TimeStamp(), frame.TimeStamp())) {
838 break;
839 }
840 decoding_state.SetState(decodable_frame);
841 if (IsContinuousInState(frame, decoding_state)) {
842 return true;
843 }
844 }
845 return false;
846}
847
848void VCMJitterBuffer::FindAndInsertContinuousFrames(
849 const VCMFrameBuffer& new_frame) {
850 VCMDecodingState decoding_state;
851 decoding_state.CopyFrom(last_decoded_state_);
852 decoding_state.SetState(&new_frame);
Noah Richardse4cb4e92015-05-22 14:03:00 -0700853 FindAndInsertContinuousFramesWithState(decoding_state);
854}
855
856void VCMJitterBuffer::FindAndInsertContinuousFramesWithState(
857 const VCMDecodingState& original_decoded_state) {
858 // Copy original_decoded_state so we can move the state forward with each
859 // decodable frame we find.
860 VCMDecodingState decoding_state;
861 decoding_state.CopyFrom(original_decoded_state);
862
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000863 // When temporal layers are available, we search for a complete or decodable
864 // frame until we hit one of the following:
865 // 1. Continuous base or sync layer.
866 // 2. The end of the list was reached.
867 for (FrameList::iterator it = incomplete_frames_.begin();
868 it != incomplete_frames_.end();) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000869 VCMFrameBuffer* frame = it->second;
Noah Richardse4cb4e92015-05-22 14:03:00 -0700870 if (IsNewerTimestamp(original_decoded_state.time_stamp(),
871 frame->TimeStamp())) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000872 ++it;
873 continue;
874 }
875 if (IsContinuousInState(*frame, decoding_state)) {
876 decodable_frames_.InsertFrame(frame);
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000877 incomplete_frames_.erase(it++);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000878 decoding_state.SetState(frame);
879 } else if (frame->TemporalId() <= 0) {
880 break;
881 } else {
882 ++it;
883 }
884 }
885}
886
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000887uint32_t VCMJitterBuffer::EstimatedJitterMs() {
888 CriticalSectionScoped cs(crit_sect_);
mikhal@webrtc.org119c67d2013-01-31 17:18:02 +0000889 // Compute RTT multiplier for estimation.
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000890 // low_rtt_nackThresholdMs_ == -1 means no FEC.
891 double rtt_mult = 1.0f;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000892 if (low_rtt_nack_threshold_ms_ >= 0 &&
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000893 rtt_ms_ >= low_rtt_nack_threshold_ms_) {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000894 // For RTTs above low_rtt_nack_threshold_ms_ we don't apply extra delay
895 // when waiting for retransmissions.
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000896 rtt_mult = 0.0f;
897 }
mikhal@webrtc.org119c67d2013-01-31 17:18:02 +0000898 return jitter_estimate_.GetJitterEstimate(rtt_mult);
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000899}
900
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000901void VCMJitterBuffer::UpdateRtt(int64_t rtt_ms) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000902 CriticalSectionScoped cs(crit_sect_);
903 rtt_ms_ = rtt_ms;
904 jitter_estimate_.UpdateRtt(rtt_ms);
905}
906
907void VCMJitterBuffer::SetNackMode(VCMNackMode mode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000908 int64_t low_rtt_nack_threshold_ms,
909 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000910 CriticalSectionScoped cs(crit_sect_);
911 nack_mode_ = mode;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000912 if (mode == kNoNack) {
913 missing_sequence_numbers_.clear();
914 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000915 assert(low_rtt_nack_threshold_ms >= -1 && high_rtt_nack_threshold_ms >= -1);
916 assert(high_rtt_nack_threshold_ms == -1 ||
917 low_rtt_nack_threshold_ms <= high_rtt_nack_threshold_ms);
918 assert(low_rtt_nack_threshold_ms > -1 || high_rtt_nack_threshold_ms == -1);
919 low_rtt_nack_threshold_ms_ = low_rtt_nack_threshold_ms;
920 high_rtt_nack_threshold_ms_ = high_rtt_nack_threshold_ms;
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +0000921 // Don't set a high start rtt if high_rtt_nack_threshold_ms_ is used, to not
Wan-Teh Changf2912872015-06-05 13:16:45 -0700922 // disable NACK in |kNack| mode.
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +0000923 if (rtt_ms_ == kDefaultRtt && high_rtt_nack_threshold_ms_ != -1) {
924 rtt_ms_ = 0;
925 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000926 if (!WaitForRetransmissions()) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000927 jitter_estimate_.ResetNackCount();
928 }
929}
930
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000931void VCMJitterBuffer::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000932 int max_packet_age_to_nack,
933 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000934 CriticalSectionScoped cs(crit_sect_);
935 assert(max_packet_age_to_nack >= 0);
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000936 assert(max_incomplete_time_ms_ >= 0);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000937 max_nack_list_size_ = max_nack_list_size;
938 max_packet_age_to_nack_ = max_packet_age_to_nack;
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000939 max_incomplete_time_ms_ = max_incomplete_time_ms;
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000940}
941
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000942VCMNackMode VCMJitterBuffer::nack_mode() const {
943 CriticalSectionScoped cs(crit_sect_);
944 return nack_mode_;
945}
946
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000947int VCMJitterBuffer::NonContinuousOrIncompleteDuration() {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000948 if (incomplete_frames_.empty()) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000949 return 0;
950 }
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000951 uint32_t start_timestamp = incomplete_frames_.Front()->TimeStamp();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000952 if (!decodable_frames_.empty()) {
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000953 start_timestamp = decodable_frames_.Back()->TimeStamp();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000954 }
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +0000955 return incomplete_frames_.Back()->TimeStamp() - start_timestamp;
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000956}
957
958uint16_t VCMJitterBuffer::EstimatedLowSequenceNumber(
959 const VCMFrameBuffer& frame) const {
960 assert(frame.GetLowSeqNum() >= 0);
961 if (frame.HaveFirstPacket())
962 return frame.GetLowSeqNum();
hclam@chromium.orgfe307e12013-05-16 21:19:59 +0000963
964 // This estimate is not accurate if more than one packet with lower sequence
965 // number is lost.
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000966 return frame.GetLowSeqNum() - 1;
967}
968
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700969std::vector<uint16_t> VCMJitterBuffer::GetNackList(bool* request_key_frame) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000970 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000971 *request_key_frame = false;
972 if (nack_mode_ == kNoNack) {
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700973 return std::vector<uint16_t>();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000974 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000975 if (last_decoded_state_.in_initial_state()) {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000976 VCMFrameBuffer* next_frame = NextFrame();
agalusza@google.comd177c102013-08-08 01:12:33 +0000977 const bool first_frame_is_key = next_frame &&
978 next_frame->FrameType() == kVideoFrameKey &&
979 next_frame->HaveFirstPacket();
stefan@webrtc.org885cd132013-04-16 09:38:26 +0000980 if (!first_frame_is_key) {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +0000981 bool have_non_empty_frame = decodable_frames_.end() != find_if(
982 decodable_frames_.begin(), decodable_frames_.end(),
983 HasNonEmptyState);
984 if (!have_non_empty_frame) {
985 have_non_empty_frame = incomplete_frames_.end() != find_if(
986 incomplete_frames_.begin(), incomplete_frames_.end(),
987 HasNonEmptyState);
988 }
stefan@webrtc.org885cd132013-04-16 09:38:26 +0000989 bool found_key_frame = RecycleFramesUntilKeyFrame();
990 if (!found_key_frame) {
991 *request_key_frame = have_non_empty_frame;
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700992 return std::vector<uint16_t>();
stefan@webrtc.org885cd132013-04-16 09:38:26 +0000993 }
994 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000995 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000996 if (TooLargeNackList()) {
997 *request_key_frame = !HandleTooLargeNackList();
stefan@webrtc.org912981f2012-10-12 07:04:52 +0000998 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000999 if (max_incomplete_time_ms_ > 0) {
1000 int non_continuous_incomplete_duration =
1001 NonContinuousOrIncompleteDuration();
1002 if (non_continuous_incomplete_duration > 90 * max_incomplete_time_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001003 LOG_F(LS_WARNING) << "Too long non-decodable duration: "
1004 << non_continuous_incomplete_duration << " > "
1005 << 90 * max_incomplete_time_ms_;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001006 FrameList::reverse_iterator rit = find_if(incomplete_frames_.rbegin(),
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001007 incomplete_frames_.rend(), IsKeyFrame);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001008 if (rit == incomplete_frames_.rend()) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001009 // Request a key frame if we don't have one already.
1010 *request_key_frame = true;
Wan-Teh Changb1825a42015-06-03 15:03:35 -07001011 return std::vector<uint16_t>();
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001012 } else {
1013 // Skip to the last key frame. If it's incomplete we will start
1014 // NACKing it.
hclam@chromium.orgfe307e12013-05-16 21:19:59 +00001015 // Note that the estimated low sequence number is correct for VP8
1016 // streams because only the first packet of a key frame is marked.
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001017 last_decoded_state_.Reset();
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001018 DropPacketsFromNackList(EstimatedLowSequenceNumber(*rit->second));
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001019 }
1020 }
1021 }
Wan-Teh Changb1825a42015-06-03 15:03:35 -07001022 std::vector<uint16_t> nack_list(missing_sequence_numbers_.begin(),
1023 missing_sequence_numbers_.end());
1024 return nack_list;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001025}
1026
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +00001027void VCMJitterBuffer::SetDecodeErrorMode(VCMDecodeErrorMode error_mode) {
1028 CriticalSectionScoped cs(crit_sect_);
mikhal@webrtc.org3c5a9242013-09-03 20:45:36 +00001029 decode_error_mode_ = error_mode;
agalusza@google.comd177c102013-08-08 01:12:33 +00001030}
1031
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001032VCMFrameBuffer* VCMJitterBuffer::NextFrame() const {
1033 if (!decodable_frames_.empty())
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001034 return decodable_frames_.Front();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001035 if (!incomplete_frames_.empty())
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001036 return incomplete_frames_.Front();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001037 return NULL;
1038}
1039
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001040bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
1041 if (nack_mode_ == kNoNack) {
1042 return true;
1043 }
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001044 // Make sure we don't add packets which are already too old to be decoded.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001045 if (!last_decoded_state_.in_initial_state()) {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001046 latest_received_sequence_number_ = LatestSequenceNumber(
1047 latest_received_sequence_number_,
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +00001048 last_decoded_state_.sequence_num());
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001049 }
stefan@webrtc.org7bc465b2013-04-11 17:48:02 +00001050 if (IsNewerSequenceNumber(sequence_number,
1051 latest_received_sequence_number_)) {
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001052 // Push any missing sequence numbers to the NACK list.
1053 for (uint16_t i = latest_received_sequence_number_ + 1;
stefan@webrtc.orga5dee332013-05-07 11:11:17 +00001054 IsNewerSequenceNumber(sequence_number, i); ++i) {
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001055 missing_sequence_numbers_.insert(missing_sequence_numbers_.end(), i);
sprang@webrtc.org0200f702015-02-16 12:06:00 +00001056 TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "AddNack",
1057 "seqnum", i);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001058 }
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001059 if (TooLargeNackList() && !HandleTooLargeNackList()) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001060 LOG(LS_WARNING) << "Requesting key frame due to too large NACK list.";
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001061 return false;
1062 }
1063 if (MissingTooOldPacket(sequence_number) &&
1064 !HandleTooOldPackets(sequence_number)) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001065 LOG(LS_WARNING) << "Requesting key frame due to missing too old packets";
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001066 return false;
1067 }
1068 } else {
1069 missing_sequence_numbers_.erase(sequence_number);
sprang@webrtc.org0200f702015-02-16 12:06:00 +00001070 TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RemoveNack",
1071 "seqnum", sequence_number);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001072 }
1073 return true;
1074}
1075
1076bool VCMJitterBuffer::TooLargeNackList() const {
1077 return missing_sequence_numbers_.size() > max_nack_list_size_;
1078}
1079
1080bool VCMJitterBuffer::HandleTooLargeNackList() {
1081 // Recycle frames until the NACK list is small enough. It is likely cheaper to
1082 // request a key frame than to retransmit this many missing packets.
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001083 LOG_F(LS_WARNING) << "NACK list has grown too large: "
1084 << missing_sequence_numbers_.size() << " > "
1085 << max_nack_list_size_;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001086 bool key_frame_found = false;
stefan@webrtc.org4d2f5de2013-04-09 18:24:41 +00001087 while (TooLargeNackList()) {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001088 key_frame_found = RecycleFramesUntilKeyFrame();
1089 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001090 return key_frame_found;
1091}
1092
1093bool VCMJitterBuffer::MissingTooOldPacket(
1094 uint16_t latest_sequence_number) const {
1095 if (missing_sequence_numbers_.empty()) {
1096 return false;
1097 }
1098 const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
1099 *missing_sequence_numbers_.begin();
1100 // Recycle frames if the NACK list contains too old sequence numbers as
1101 // the packets may have already been dropped by the sender.
1102 return age_of_oldest_missing_packet > max_packet_age_to_nack_;
1103}
1104
1105bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) {
1106 bool key_frame_found = false;
1107 const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
1108 *missing_sequence_numbers_.begin();
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001109 LOG_F(LS_WARNING) << "NACK list contains too old sequence numbers: "
1110 << age_of_oldest_missing_packet << " > "
1111 << max_packet_age_to_nack_;
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001112 while (MissingTooOldPacket(latest_sequence_number)) {
1113 key_frame_found = RecycleFramesUntilKeyFrame();
1114 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001115 return key_frame_found;
1116}
1117
1118void VCMJitterBuffer::DropPacketsFromNackList(
1119 uint16_t last_decoded_sequence_number) {
1120 // Erase all sequence numbers from the NACK list which we won't need any
1121 // longer.
1122 missing_sequence_numbers_.erase(missing_sequence_numbers_.begin(),
1123 missing_sequence_numbers_.upper_bound(
1124 last_decoded_sequence_number));
1125}
1126
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001127int64_t VCMJitterBuffer::LastDecodedTimestamp() const {
1128 CriticalSectionScoped cs(crit_sect_);
1129 return last_decoded_state_.time_stamp();
1130}
1131
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001132void VCMJitterBuffer::RenderBufferSize(uint32_t* timestamp_start,
1133 uint32_t* timestamp_end) {
1134 CriticalSectionScoped cs(crit_sect_);
1135 CleanUpOldOrEmptyFrames();
1136 *timestamp_start = 0;
1137 *timestamp_end = 0;
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001138 if (decodable_frames_.empty()) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +00001139 return;
1140 }
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001141 *timestamp_start = decodable_frames_.Front()->TimeStamp();
1142 *timestamp_end = decodable_frames_.Back()->TimeStamp();
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +00001143}
1144
pbos@webrtc.org55707692014-12-19 15:45:03 +00001145void VCMJitterBuffer::RegisterStatsCallback(
1146 VCMReceiveStatisticsCallback* callback) {
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +00001147 CriticalSectionScoped cs(crit_sect_);
pbos@webrtc.org55707692014-12-19 15:45:03 +00001148 stats_callback_ = callback;
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +00001149}
1150
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001151VCMFrameBuffer* VCMJitterBuffer::GetEmptyFrame() {
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001152 if (free_frames_.empty()) {
1153 if (!TryToIncreaseJitterBufferSize()) {
1154 return NULL;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001155 }
1156 }
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001157 VCMFrameBuffer* frame = free_frames_.front();
1158 free_frames_.pop_front();
1159 return frame;
1160}
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001161
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001162bool VCMJitterBuffer::TryToIncreaseJitterBufferSize() {
1163 if (max_number_of_frames_ >= kMaxNumberOfFrames)
1164 return false;
pbos@webrtc.org4f16c872014-11-24 09:06:48 +00001165 free_frames_.push_back(new VCMFrameBuffer());
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001166 ++max_number_of_frames_;
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001167 TRACE_COUNTER1("webrtc", "JBMaxFrames", max_number_of_frames_);
1168 return true;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001169}
1170
1171// Recycle oldest frames up to a key frame, used if jitter buffer is completely
1172// full.
1173bool VCMJitterBuffer::RecycleFramesUntilKeyFrame() {
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001174 // First release incomplete frames, and only release decodable frames if there
1175 // are no incomplete ones.
1176 FrameList::iterator key_frame_it;
1177 bool key_frame_found = false;
1178 int dropped_frames = 0;
1179 dropped_frames += incomplete_frames_.RecycleFramesUntilKeyFrame(
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001180 &key_frame_it, &free_frames_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001181 key_frame_found = key_frame_it != incomplete_frames_.end();
1182 if (dropped_frames == 0) {
1183 dropped_frames += decodable_frames_.RecycleFramesUntilKeyFrame(
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001184 &key_frame_it, &free_frames_);
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001185 key_frame_found = key_frame_it != decodable_frames_.end();
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001186 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001187 TRACE_EVENT_INSTANT0("webrtc", "JB::RecycleFramesUntilKeyFrame");
1188 if (key_frame_found) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +00001189 LOG(LS_INFO) << "Found key frame while dropping frames.";
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001190 // Reset last decoded state to make sure the next frame decoded is a key
1191 // frame, and start NACKing from here.
1192 last_decoded_state_.Reset();
stefan@webrtc.org50fb4af2013-06-17 07:33:58 +00001193 DropPacketsFromNackList(EstimatedLowSequenceNumber(*key_frame_it->second));
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001194 } else if (decodable_frames_.empty()) {
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001195 // All frames dropped. Reset the decoding state and clear missing sequence
1196 // numbers as we're starting fresh.
1197 last_decoded_state_.Reset();
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001198 missing_sequence_numbers_.clear();
edjee@google.com79b02892013-04-04 19:43:34 +00001199 }
stefan@webrtc.orgc8b29a22013-06-17 07:13:16 +00001200 return key_frame_found;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001201}
1202
1203// Must be called under the critical section |crit_sect_|.
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001204void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +00001205 incoming_frame_count_++;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001206
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001207 if (frame.FrameType() == kVideoFrameKey) {
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +00001208 TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
1209 frame.TimeStamp(), "KeyComplete");
hclam@chromium.org806dc3b2013-04-09 19:54:10 +00001210 } else {
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +00001211 TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
1212 frame.TimeStamp(), "DeltaComplete");
hclam@chromium.org806dc3b2013-04-09 19:54:10 +00001213 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001214
1215 // Update receive statistics. We count all layers, thus when you use layers
1216 // adding all key and delta frames might differ from frame count.
stefan@webrtc.org4cf1a8a2013-06-27 15:20:14 +00001217 if (frame.IsSessionComplete()) {
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +00001218 if (frame.FrameType() == kVideoFrameKey) {
1219 ++receive_statistics_.key_frames;
1220 } else {
1221 ++receive_statistics_.delta_frames;
1222 }
pbos@webrtc.org55707692014-12-19 15:45:03 +00001223 if (stats_callback_ != NULL)
1224 stats_callback_->OnFrameCountsUpdated(receive_statistics_);
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001225 }
niklase@google.com470e71d2011-07-07 08:21:25 +00001226}
1227
agalusza@google.comd818dcb2013-07-29 21:48:11 +00001228void VCMJitterBuffer::UpdateAveragePacketsPerFrame(int current_number_packets) {
1229 if (frame_counter_ > kFastConvergeThreshold) {
1230 average_packets_per_frame_ = average_packets_per_frame_
1231 * (1 - kNormalConvergeMultiplier)
1232 + current_number_packets * kNormalConvergeMultiplier;
1233 } else if (frame_counter_ > 0) {
1234 average_packets_per_frame_ = average_packets_per_frame_
1235 * (1 - kFastConvergeMultiplier)
1236 + current_number_packets * kFastConvergeMultiplier;
1237 frame_counter_++;
1238 } else {
1239 average_packets_per_frame_ = current_number_packets;
1240 frame_counter_++;
1241 }
1242}
1243
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001244// Must be called under the critical section |crit_sect_|.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001245void VCMJitterBuffer::CleanUpOldOrEmptyFrames() {
pbos@webrtc.org4f16c872014-11-24 09:06:48 +00001246 decodable_frames_.CleanUpOldOrEmptyFrames(&last_decoded_state_,
1247 &free_frames_);
1248 incomplete_frames_.CleanUpOldOrEmptyFrames(&last_decoded_state_,
1249 &free_frames_);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001250 if (!last_decoded_state_.in_initial_state()) {
1251 DropPacketsFromNackList(last_decoded_state_.sequence_num());
1252 }
mikhal@webrtc.org832caca2011-12-13 21:15:05 +00001253}
1254
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001255// Must be called from within |crit_sect_|.
1256bool VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001257 return missing_sequence_numbers_.find(packet.seqNum) !=
1258 missing_sequence_numbers_.end();
niklase@google.com470e71d2011-07-07 08:21:25 +00001259}
1260
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001261// Must be called under the critical section |crit_sect_|. Should never be
1262// called with retransmitted frames, they must be filtered out before this
1263// function is called.
1264void VCMJitterBuffer::UpdateJitterEstimate(const VCMJitterSample& sample,
1265 bool incomplete_frame) {
1266 if (sample.latest_packet_time == -1) {
1267 return;
1268 }
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001269 UpdateJitterEstimate(sample.latest_packet_time, sample.timestamp,
1270 sample.frame_size, incomplete_frame);
1271}
1272
1273// Must be called under the critical section crit_sect_. Should never be
1274// called with retransmitted frames, they must be filtered out before this
1275// function is called.
1276void VCMJitterBuffer::UpdateJitterEstimate(const VCMFrameBuffer& frame,
1277 bool incomplete_frame) {
1278 if (frame.LatestPacketTimeMs() == -1) {
1279 return;
1280 }
1281 // No retransmitted frames should be a part of the jitter
1282 // estimate.
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001283 UpdateJitterEstimate(frame.LatestPacketTimeMs(), frame.TimeStamp(),
1284 frame.Length(), incomplete_frame);
1285}
1286
1287// Must be called under the critical section |crit_sect_|. Should never be
1288// called with retransmitted frames, they must be filtered out before this
1289// function is called.
1290void VCMJitterBuffer::UpdateJitterEstimate(
1291 int64_t latest_packet_time_ms,
1292 uint32_t timestamp,
1293 unsigned int frame_size,
1294 bool incomplete_frame) {
1295 if (latest_packet_time_ms == -1) {
1296 return;
1297 }
1298 int64_t frame_delay;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001299 bool not_reordered = inter_frame_delay_.CalculateDelay(timestamp,
1300 &frame_delay,
1301 latest_packet_time_ms);
1302 // Filter out frames which have been reordered in time by the network
1303 if (not_reordered) {
1304 // Update the jitter estimate with the new samples
1305 jitter_estimate_.UpdateEstimate(frame_delay, frame_size, incomplete_frame);
1306 }
1307}
1308
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001309bool VCMJitterBuffer::WaitForRetransmissions() {
1310 if (nack_mode_ == kNoNack) {
1311 // NACK disabled -> don't wait for retransmissions.
1312 return false;
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001313 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +00001314 // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
1315 // that case we don't wait for retransmissions.
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001316 if (high_rtt_nack_threshold_ms_ >= 0 &&
pkasting@chromium.org16825b12015-01-12 21:51:21 +00001317 rtt_ms_ >= high_rtt_nack_threshold_ms_) {
stefan@webrtc.org912981f2012-10-12 07:04:52 +00001318 return false;
1319 }
1320 return true;
1321}
stefan@webrtc.org932ab182011-11-29 11:33:31 +00001322} // namespace webrtc