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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()),
Niels Möllerdb64d992019-03-29 14:30:53 +010037 absl::WrapUnique(EventWrapper::Create())) {}
Qiang Chend4cec152015-06-19 09:17:00 -070038
39VCMReceiver::VCMReceiver(VCMTiming* timing,
40 Clock* clock,
kwiberg3f55dea2016-02-29 05:51:59 -080041 std::unique_ptr<EventWrapper> receiver_event,
42 std::unique_ptr<EventWrapper> jitter_buffer_event)
kthelgasond701dfd2017-03-27 07:24:57 -070043 : clock_(clock),
Niels Möllerdb64d992019-03-29 14:30:53 +010044 jitter_buffer_(clock_, std::move(jitter_buffer_event)),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000045 timing_(timing),
kwiberg0eb15ed2015-12-17 03:04:15 -080046 render_wait_event_(std::move(receiver_event)),
Peter Boström5464a6e2015-04-21 16:35:50 +020047 max_video_delay_ms_(kMaxVideoDelayMs) {
48 Reset();
49}
niklase@google.com470e71d2011-07-07 08:21:25 +000050
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000051VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000052 render_wait_event_->Set();
niklase@google.com470e71d2011-07-07 08:21:25 +000053}
54
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000055void VCMReceiver::Reset() {
kthelgasond701dfd2017-03-27 07:24:57 -070056 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000057 if (!jitter_buffer_.Running()) {
58 jitter_buffer_.Start();
59 } else {
60 jitter_buffer_.Flush();
61 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000062}
63
Johan Ahlers95348f72016-06-28 11:11:28 +020064int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000065 // Insert the packet into the jitter buffer. The packet can either be empty or
66 // contain media at this point.
67 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -080068 const VCMFrameBufferEnum ret =
69 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000070 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000071 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000072 } else if (ret == kFlushIndicator) {
73 return VCM_FLUSH_INDICATOR;
74 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000075 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000076 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000077 if (ret == kCompleteSession && !retransmitted) {
78 // We don't want to include timestamps which have suffered from
79 // retransmission here, since we compensate with extra retransmission
80 // delay within the jitter estimate.
81 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
82 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000083 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000084}
85
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +000086void VCMReceiver::TriggerDecoderShutdown() {
87 jitter_buffer_.Stop();
88 render_wait_event_->Set();
89}
90
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000091VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
perkj796cfaf2015-12-10 09:27:38 -080092 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000093 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000094 uint32_t frame_timestamp = 0;
isheriff6b4b5f32016-06-08 00:24:21 -070095 int min_playout_delay_ms = -1;
96 int max_playout_delay_ms = -1;
Johan Ahlers31b2ec42016-06-28 13:32:49 +020097 int64_t render_time_ms = 0;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000098 // Exhaust wait time to get a complete frame for decoding.
isheriff6b4b5f32016-06-08 00:24:21 -070099 VCMEncodedFrame* found_frame =
100 jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000101
isheriff6b4b5f32016-06-08 00:24:21 -0700102 if (found_frame) {
Niels Möller23775882018-08-16 10:24:12 +0200103 frame_timestamp = found_frame->Timestamp();
isheriff6b4b5f32016-06-08 00:24:21 -0700104 min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
105 max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
106 } else {
Niels Möller375b3462019-01-10 15:35:56 +0100107 return nullptr;
isheriff6b4b5f32016-06-08 00:24:21 -0700108 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000109
isheriff6b4b5f32016-06-08 00:24:21 -0700110 if (min_playout_delay_ms >= 0)
111 timing_->set_min_playout_delay(min_playout_delay_ms);
112
113 if (max_playout_delay_ms >= 0)
114 timing_->set_max_playout_delay(max_playout_delay_ms);
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000115
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000116 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000117 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000118 const int64_t now_ms = clock_->TimeInMilliseconds();
119 timing_->UpdateCurrentDelay(frame_timestamp);
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200120 render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000121 // Check render timing.
122 bool timing_error = false;
123 // Assume that render timing errors are due to changes in the video stream.
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200124 if (render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000125 timing_error = true;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200126 } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
127 int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
Mirko Bonadei675513b2017-11-09 11:09:25 +0100128 RTC_LOG(LS_WARNING)
129 << "A frame about to be decoded is out of the configured "
130 << "delay bounds (" << frame_delay << " > " << max_video_delay_ms_
131 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000132 timing_error = true;
133 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
134 max_video_delay_ms_) {
Mirko Bonadei675513b2017-11-09 11:09:25 +0100135 RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
136 << max_video_delay_ms_
137 << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000138 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000139 }
140
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000141 if (timing_error) {
142 // Timing error => reset timing and flush the jitter buffer.
143 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000144 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000145 return NULL;
146 }
147
perkj796cfaf2015-12-10 09:27:38 -0800148 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000149 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800150 const int32_t available_wait_time =
151 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000152 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800153 uint16_t new_max_wait_time =
154 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
Ilya Nikolaevskiy8c4fe162018-02-27 15:49:47 +0100155 uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
156 timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds()));
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000157 if (new_max_wait_time < wait_time_ms) {
158 // We're not allowed to wait until the frame is supposed to be rendered,
159 // waiting as long as we're allowed to avoid busy looping, and then return
160 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700161 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000162 return NULL;
163 }
164 // Wait until it's time to render.
165 render_wait_event_->Wait(wait_time_ms);
166 }
167
168 // Extract the frame from the jitter buffer and set the render time.
169 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000170 if (frame == NULL) {
171 return NULL;
172 }
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200173 frame->SetRenderTime(render_time_ms);
Niels Möller23775882018-08-16 10:24:12 +0200174 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->Timestamp(), "SetRenderTS",
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200175 "render_time", frame->RenderTimeMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000176 if (!frame->Complete()) {
177 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000178 bool retransmitted = false;
179 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000180 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000181 if (last_packet_time_ms >= 0 && !retransmitted) {
182 // We don't want to include timestamps which have suffered from
183 // retransmission here, since we compensate with extra retransmission
184 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000185 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000186 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000187 }
188 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000189}
190
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000191void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
192 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000193}
194
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000195void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000196 int max_packet_age_to_nack,
197 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800198 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000199 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000200}
201
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700202std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
203 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000204}
205
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000206} // namespace webrtc