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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>
Yves Gerey3e707812018-11-28 16:47:49 +010014#include <cstdint>
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
Yves Gerey3e707812018-11-28 16:47:49 +010019#include "absl/memory/memory.h"
20#include "api/video/encoded_image.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020021#include "modules/video_coding/encoded_frame.h"
22#include "modules/video_coding/internal_defines.h"
Yves Gerey3e707812018-11-28 16:47:49 +010023#include "modules/video_coding/jitter_buffer_common.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020024#include "rtc_base/logging.h"
Ilya Nikolaevskiy8c4fe162018-02-27 15:49:47 +010025#include "rtc_base/numerics/safe_conversions.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020026#include "rtc_base/trace_event.h"
27#include "system_wrappers/include/clock.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000028
niklase@google.com470e71d2011-07-07 08:21:25 +000029namespace webrtc {
30
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000031enum { kMaxReceiverDelayMs = 10000 };
32
Niels Möller689983f2018-11-07 16:36:22 +010033VCMReceiver::VCMReceiver(VCMTiming* timing, Clock* clock)
philipel83f831a2016-03-12 03:30:23 -080034 : VCMReceiver::VCMReceiver(timing,
35 clock,
Niels Möller689983f2018-11-07 16:36:22 +010036 absl::WrapUnique(EventWrapper::Create()),
37 absl::WrapUnique(EventWrapper::Create()),
philipel83f831a2016-03-12 03:30:23 -080038 nullptr, // NackSender
39 nullptr) // KeyframeRequestSender
40{}
41
42VCMReceiver::VCMReceiver(VCMTiming* timing,
43 Clock* clock,
philipel83f831a2016-03-12 03:30:23 -080044 NackSender* nack_sender,
45 KeyFrameRequestSender* keyframe_request_sender)
Niels Möller689983f2018-11-07 16:36:22 +010046 : VCMReceiver(timing,
47 clock,
48 absl::WrapUnique(EventWrapper::Create()),
49 absl::WrapUnique(EventWrapper::Create()),
50 nack_sender,
51 keyframe_request_sender) {}
Qiang Chend4cec152015-06-19 09:17:00 -070052
53VCMReceiver::VCMReceiver(VCMTiming* timing,
54 Clock* clock,
kwiberg3f55dea2016-02-29 05:51:59 -080055 std::unique_ptr<EventWrapper> receiver_event,
56 std::unique_ptr<EventWrapper> jitter_buffer_event)
philipel83f831a2016-03-12 03:30:23 -080057 : VCMReceiver::VCMReceiver(timing,
58 clock,
59 std::move(receiver_event),
60 std::move(jitter_buffer_event),
61 nullptr, // NackSender
62 nullptr) // KeyframeRequestSender
63{}
64
65VCMReceiver::VCMReceiver(VCMTiming* timing,
66 Clock* clock,
67 std::unique_ptr<EventWrapper> receiver_event,
68 std::unique_ptr<EventWrapper> jitter_buffer_event,
69 NackSender* nack_sender,
70 KeyFrameRequestSender* keyframe_request_sender)
kthelgasond701dfd2017-03-27 07:24:57 -070071 : clock_(clock),
philipel83f831a2016-03-12 03:30:23 -080072 jitter_buffer_(clock_,
73 std::move(jitter_buffer_event),
74 nack_sender,
75 keyframe_request_sender),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000076 timing_(timing),
kwiberg0eb15ed2015-12-17 03:04:15 -080077 render_wait_event_(std::move(receiver_event)),
Peter Boström5464a6e2015-04-21 16:35:50 +020078 max_video_delay_ms_(kMaxVideoDelayMs) {
79 Reset();
80}
niklase@google.com470e71d2011-07-07 08:21:25 +000081
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000082VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000083 render_wait_event_->Set();
niklase@google.com470e71d2011-07-07 08:21:25 +000084}
85
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000086void VCMReceiver::Reset() {
kthelgasond701dfd2017-03-27 07:24:57 -070087 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000088 if (!jitter_buffer_.Running()) {
89 jitter_buffer_.Start();
90 } else {
91 jitter_buffer_.Flush();
92 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000093}
94
pkasting@chromium.org16825b12015-01-12 21:51:21 +000095void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000096 jitter_buffer_.UpdateRtt(rtt);
97}
98
Johan Ahlers95348f72016-06-28 11:11:28 +020099int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000100 // Insert the packet into the jitter buffer. The packet can either be empty or
101 // contain media at this point.
102 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -0800103 const VCMFrameBufferEnum ret =
104 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000105 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000106 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000107 } else if (ret == kFlushIndicator) {
108 return VCM_FLUSH_INDICATOR;
109 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000110 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000111 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000112 if (ret == kCompleteSession && !retransmitted) {
113 // We don't want to include timestamps which have suffered from
114 // retransmission here, since we compensate with extra retransmission
115 // delay within the jitter estimate.
116 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
117 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000118 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000119}
120
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +0000121void VCMReceiver::TriggerDecoderShutdown() {
122 jitter_buffer_.Stop();
123 render_wait_event_->Set();
124}
125
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000126VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800127 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000128 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000129 uint32_t frame_timestamp = 0;
isheriff6b4b5f32016-06-08 00:24:21 -0700130 int min_playout_delay_ms = -1;
131 int max_playout_delay_ms = -1;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200132 int64_t render_time_ms = 0;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000133 // Exhaust wait time to get a complete frame for decoding.
isheriff6b4b5f32016-06-08 00:24:21 -0700134 VCMEncodedFrame* found_frame =
135 jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000136
isheriff6b4b5f32016-06-08 00:24:21 -0700137 if (found_frame) {
Niels Möller23775882018-08-16 10:24:12 +0200138 frame_timestamp = found_frame->Timestamp();
isheriff6b4b5f32016-06-08 00:24:21 -0700139 min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
140 max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
141 } else {
Niels Möller375b3462019-01-10 15:35:56 +0100142 return nullptr;
isheriff6b4b5f32016-06-08 00:24:21 -0700143 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000144
isheriff6b4b5f32016-06-08 00:24:21 -0700145 if (min_playout_delay_ms >= 0)
146 timing_->set_min_playout_delay(min_playout_delay_ms);
147
148 if (max_playout_delay_ms >= 0)
149 timing_->set_max_playout_delay(max_playout_delay_ms);
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000150
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000151 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000152 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000153 const int64_t now_ms = clock_->TimeInMilliseconds();
154 timing_->UpdateCurrentDelay(frame_timestamp);
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200155 render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000156 // Check render timing.
157 bool timing_error = false;
158 // Assume that render timing errors are due to changes in the video stream.
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200159 if (render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000160 timing_error = true;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200161 } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
162 int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
Mirko Bonadei675513b2017-11-09 11:09:25 +0100163 RTC_LOG(LS_WARNING)
164 << "A frame about to be decoded is out of the configured "
165 << "delay bounds (" << frame_delay << " > " << max_video_delay_ms_
166 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000167 timing_error = true;
168 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
169 max_video_delay_ms_) {
Mirko Bonadei675513b2017-11-09 11:09:25 +0100170 RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
171 << max_video_delay_ms_
172 << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000173 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000174 }
175
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000176 if (timing_error) {
177 // Timing error => reset timing and flush the jitter buffer.
178 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000179 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000180 return NULL;
181 }
182
perkj796cfaf2015-12-10 09:27:38 -0800183 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000184 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800185 const int32_t available_wait_time =
186 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000187 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800188 uint16_t new_max_wait_time =
189 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
Ilya Nikolaevskiy8c4fe162018-02-27 15:49:47 +0100190 uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
191 timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds()));
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000192 if (new_max_wait_time < wait_time_ms) {
193 // We're not allowed to wait until the frame is supposed to be rendered,
194 // waiting as long as we're allowed to avoid busy looping, and then return
195 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700196 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000197 return NULL;
198 }
199 // Wait until it's time to render.
200 render_wait_event_->Wait(wait_time_ms);
201 }
202
203 // Extract the frame from the jitter buffer and set the render time.
204 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000205 if (frame == NULL) {
206 return NULL;
207 }
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200208 frame->SetRenderTime(render_time_ms);
Niels Möller23775882018-08-16 10:24:12 +0200209 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->Timestamp(), "SetRenderTS",
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200210 "render_time", frame->RenderTimeMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000211 if (!frame->Complete()) {
212 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000213 bool retransmitted = false;
214 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000215 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000216 if (last_packet_time_ms >= 0 && !retransmitted) {
217 // We don't want to include timestamps which have suffered from
218 // retransmission here, since we compensate with extra retransmission
219 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000220 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000221 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000222 }
223 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000224}
225
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000226void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
227 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000228}
229
philipel9d3ab612015-12-21 04:12:39 -0800230void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000231 assert(bitrate);
232 assert(framerate);
233 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000234}
235
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000236void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000237 int64_t low_rtt_nack_threshold_ms,
238 int64_t high_rtt_nack_threshold_ms) {
kthelgasond701dfd2017-03-27 07:24:57 -0700239 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000240 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000241 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
242 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000243}
244
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000245void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000246 int max_packet_age_to_nack,
247 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800248 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000249 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000250}
251
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000252VCMNackMode VCMReceiver::NackMode() const {
kthelgasond701dfd2017-03-27 07:24:57 -0700253 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000254 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000255}
256
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700257std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
258 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000259}
260
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000261} // namespace webrtc