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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
stefan@webrtc.org91c63082012-01-31 10:49:08 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "modules/video_coding/receiver.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000012
13#include <assert.h>
14
pbos@webrtc.org3f655aa2014-03-18 11:10:11 +000015#include <cstdlib>
kwiberg0eb15ed2015-12-17 03:04:15 -080016#include <utility>
philipel9d3ab612015-12-21 04:12:39 -080017#include <vector>
pbos@webrtc.org3f655aa2014-03-18 11:10:11 +000018
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020019#include "modules/video_coding/encoded_frame.h"
20#include "modules/video_coding/internal_defines.h"
21#include "modules/video_coding/media_opt_util.h"
22#include "rtc_base/logging.h"
23#include "rtc_base/trace_event.h"
24#include "system_wrappers/include/clock.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000025
niklase@google.com470e71d2011-07-07 08:21:25 +000026namespace webrtc {
27
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000028enum { kMaxReceiverDelayMs = 10000 };
29
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000030VCMReceiver::VCMReceiver(VCMTiming* timing,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000031 Clock* clock,
Wan-Teh Chang92d94892015-05-28 13:36:06 -070032 EventFactory* event_factory)
philipel83f831a2016-03-12 03:30:23 -080033 : VCMReceiver::VCMReceiver(timing,
34 clock,
35 event_factory,
36 nullptr, // NackSender
37 nullptr) // KeyframeRequestSender
38{}
39
40VCMReceiver::VCMReceiver(VCMTiming* timing,
41 Clock* clock,
42 EventFactory* event_factory,
43 NackSender* nack_sender,
44 KeyFrameRequestSender* keyframe_request_sender)
Peter Boström0b250722016-04-22 18:23:15 +020045 : VCMReceiver(
46 timing,
47 clock,
48 std::unique_ptr<EventWrapper>(event_factory
49 ? event_factory->CreateEvent()
50 : EventWrapper::Create()),
51 std::unique_ptr<EventWrapper>(event_factory
52 ? event_factory->CreateEvent()
53 : EventWrapper::Create()),
54 nack_sender,
55 keyframe_request_sender) {}
Qiang Chend4cec152015-06-19 09:17:00 -070056
57VCMReceiver::VCMReceiver(VCMTiming* timing,
58 Clock* clock,
kwiberg3f55dea2016-02-29 05:51:59 -080059 std::unique_ptr<EventWrapper> receiver_event,
60 std::unique_ptr<EventWrapper> jitter_buffer_event)
philipel83f831a2016-03-12 03:30:23 -080061 : VCMReceiver::VCMReceiver(timing,
62 clock,
63 std::move(receiver_event),
64 std::move(jitter_buffer_event),
65 nullptr, // NackSender
66 nullptr) // KeyframeRequestSender
67{}
68
69VCMReceiver::VCMReceiver(VCMTiming* timing,
70 Clock* clock,
71 std::unique_ptr<EventWrapper> receiver_event,
72 std::unique_ptr<EventWrapper> jitter_buffer_event,
73 NackSender* nack_sender,
74 KeyFrameRequestSender* keyframe_request_sender)
kthelgasond701dfd2017-03-27 07:24:57 -070075 : clock_(clock),
philipel83f831a2016-03-12 03:30:23 -080076 jitter_buffer_(clock_,
77 std::move(jitter_buffer_event),
78 nack_sender,
79 keyframe_request_sender),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000080 timing_(timing),
kwiberg0eb15ed2015-12-17 03:04:15 -080081 render_wait_event_(std::move(receiver_event)),
Peter Boström5464a6e2015-04-21 16:35:50 +020082 max_video_delay_ms_(kMaxVideoDelayMs) {
83 Reset();
84}
niklase@google.com470e71d2011-07-07 08:21:25 +000085
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000086VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000087 render_wait_event_->Set();
niklase@google.com470e71d2011-07-07 08:21:25 +000088}
89
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000090void VCMReceiver::Reset() {
kthelgasond701dfd2017-03-27 07:24:57 -070091 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000092 if (!jitter_buffer_.Running()) {
93 jitter_buffer_.Start();
94 } else {
95 jitter_buffer_.Flush();
96 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000097}
98
pkasting@chromium.org16825b12015-01-12 21:51:21 +000099void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000100 jitter_buffer_.UpdateRtt(rtt);
101}
102
Johan Ahlers95348f72016-06-28 11:11:28 +0200103int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000104 // Insert the packet into the jitter buffer. The packet can either be empty or
105 // contain media at this point.
106 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -0800107 const VCMFrameBufferEnum ret =
108 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000109 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000110 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000111 } else if (ret == kFlushIndicator) {
112 return VCM_FLUSH_INDICATOR;
113 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000114 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000115 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000116 if (ret == kCompleteSession && !retransmitted) {
117 // We don't want to include timestamps which have suffered from
118 // retransmission here, since we compensate with extra retransmission
119 // delay within the jitter estimate.
120 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
121 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000122 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000123}
124
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +0000125void VCMReceiver::TriggerDecoderShutdown() {
126 jitter_buffer_.Stop();
127 render_wait_event_->Set();
128}
129
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000130VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800131 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000132 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000133 uint32_t frame_timestamp = 0;
isheriff6b4b5f32016-06-08 00:24:21 -0700134 int min_playout_delay_ms = -1;
135 int max_playout_delay_ms = -1;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200136 int64_t render_time_ms = 0;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000137 // Exhaust wait time to get a complete frame for decoding.
isheriff6b4b5f32016-06-08 00:24:21 -0700138 VCMEncodedFrame* found_frame =
139 jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000140
isheriff6b4b5f32016-06-08 00:24:21 -0700141 if (found_frame) {
142 frame_timestamp = found_frame->TimeStamp();
143 min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
144 max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
145 } else {
146 if (!jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp))
147 return nullptr;
148 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000149
isheriff6b4b5f32016-06-08 00:24:21 -0700150 if (min_playout_delay_ms >= 0)
151 timing_->set_min_playout_delay(min_playout_delay_ms);
152
153 if (max_playout_delay_ms >= 0)
154 timing_->set_max_playout_delay(max_playout_delay_ms);
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000155
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000156 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000157 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000158 const int64_t now_ms = clock_->TimeInMilliseconds();
159 timing_->UpdateCurrentDelay(frame_timestamp);
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200160 render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000161 // Check render timing.
162 bool timing_error = false;
163 // Assume that render timing errors are due to changes in the video stream.
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200164 if (render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000165 timing_error = true;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200166 } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
167 int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
Mirko Bonadei675513b2017-11-09 11:09:25 +0100168 RTC_LOG(LS_WARNING)
169 << "A frame about to be decoded is out of the configured "
170 << "delay bounds (" << frame_delay << " > " << max_video_delay_ms_
171 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000172 timing_error = true;
173 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
174 max_video_delay_ms_) {
Mirko Bonadei675513b2017-11-09 11:09:25 +0100175 RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
176 << max_video_delay_ms_
177 << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000178 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000179 }
180
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000181 if (timing_error) {
182 // Timing error => reset timing and flush the jitter buffer.
183 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000184 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000185 return NULL;
186 }
187
perkj796cfaf2015-12-10 09:27:38 -0800188 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000189 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800190 const int32_t available_wait_time =
191 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000192 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800193 uint16_t new_max_wait_time =
194 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200195 uint32_t wait_time_ms =
196 timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000197 if (new_max_wait_time < wait_time_ms) {
198 // We're not allowed to wait until the frame is supposed to be rendered,
199 // waiting as long as we're allowed to avoid busy looping, and then return
200 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700201 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000202 return NULL;
203 }
204 // Wait until it's time to render.
205 render_wait_event_->Wait(wait_time_ms);
206 }
207
208 // Extract the frame from the jitter buffer and set the render time.
209 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000210 if (frame == NULL) {
211 return NULL;
212 }
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200213 frame->SetRenderTime(render_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800214 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200215 "render_time", frame->RenderTimeMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000216 if (!frame->Complete()) {
217 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000218 bool retransmitted = false;
219 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000220 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000221 if (last_packet_time_ms >= 0 && !retransmitted) {
222 // We don't want to include timestamps which have suffered from
223 // retransmission here, since we compensate with extra retransmission
224 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000225 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000226 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000227 }
228 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000229}
230
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000231void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
232 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000233}
234
philipel9d3ab612015-12-21 04:12:39 -0800235void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000236 assert(bitrate);
237 assert(framerate);
238 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000239}
240
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000241void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000242 int64_t low_rtt_nack_threshold_ms,
243 int64_t high_rtt_nack_threshold_ms) {
kthelgasond701dfd2017-03-27 07:24:57 -0700244 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000245 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000246 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
247 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000248}
249
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000250void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000251 int max_packet_age_to_nack,
252 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800253 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000254 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000255}
256
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000257VCMNackMode VCMReceiver::NackMode() const {
kthelgasond701dfd2017-03-27 07:24:57 -0700258 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000259 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000260}
261
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700262std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
263 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000264}
265
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000266void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
267 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000268}
269
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000270VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000271 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000272}
273
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000274int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
kthelgasond701dfd2017-03-27 07:24:57 -0700275 rtc::CritScope cs(&crit_sect_);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000276 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
277 return -1;
278 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000279 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000280 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000281 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000282 return 0;
283}
284
pbos@webrtc.org55707692014-12-19 15:45:03 +0000285void VCMReceiver::RegisterStatsCallback(
286 VCMReceiveStatisticsCallback* callback) {
287 jitter_buffer_.RegisterStatsCallback(callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000288}
289
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000290} // namespace webrtc