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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>
Jonas Olssona4d87372019-07-05 19:08:33 +020014
Yves Gerey3e707812018-11-28 16:47:49 +010015#include <cstdint>
pbos@webrtc.org3f655aa2014-03-18 11:10:11 +000016#include <cstdlib>
kwiberg0eb15ed2015-12-17 03:04:15 -080017#include <utility>
philipel9d3ab612015-12-21 04:12:39 -080018#include <vector>
pbos@webrtc.org3f655aa2014-03-18 11:10:11 +000019
Yves Gerey3e707812018-11-28 16:47:49 +010020#include "absl/memory/memory.h"
21#include "api/video/encoded_image.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020022#include "modules/video_coding/encoded_frame.h"
23#include "modules/video_coding/internal_defines.h"
Yves Gerey3e707812018-11-28 16:47:49 +010024#include "modules/video_coding/jitter_buffer_common.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020025#include "rtc_base/logging.h"
Ilya Nikolaevskiy8c4fe162018-02-27 15:49:47 +010026#include "rtc_base/numerics/safe_conversions.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "rtc_base/trace_event.h"
28#include "system_wrappers/include/clock.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000029
niklase@google.com470e71d2011-07-07 08:21:25 +000030namespace webrtc {
31
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000032enum { kMaxReceiverDelayMs = 10000 };
33
Niels Möller689983f2018-11-07 16:36:22 +010034VCMReceiver::VCMReceiver(VCMTiming* timing, Clock* clock)
philipel83f831a2016-03-12 03:30:23 -080035 : VCMReceiver::VCMReceiver(timing,
36 clock,
Niels Möller689983f2018-11-07 16:36:22 +010037 absl::WrapUnique(EventWrapper::Create()),
Niels Möllerdb64d992019-03-29 14:30:53 +010038 absl::WrapUnique(EventWrapper::Create())) {}
Qiang Chend4cec152015-06-19 09:17:00 -070039
40VCMReceiver::VCMReceiver(VCMTiming* timing,
41 Clock* clock,
kwiberg3f55dea2016-02-29 05:51:59 -080042 std::unique_ptr<EventWrapper> receiver_event,
43 std::unique_ptr<EventWrapper> jitter_buffer_event)
kthelgasond701dfd2017-03-27 07:24:57 -070044 : clock_(clock),
Niels Möllerdb64d992019-03-29 14:30:53 +010045 jitter_buffer_(clock_, std::move(jitter_buffer_event)),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000046 timing_(timing),
kwiberg0eb15ed2015-12-17 03:04:15 -080047 render_wait_event_(std::move(receiver_event)),
Peter Boström5464a6e2015-04-21 16:35:50 +020048 max_video_delay_ms_(kMaxVideoDelayMs) {
49 Reset();
50}
niklase@google.com470e71d2011-07-07 08:21:25 +000051
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000052VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000053 render_wait_event_->Set();
niklase@google.com470e71d2011-07-07 08:21:25 +000054}
55
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000056void VCMReceiver::Reset() {
kthelgasond701dfd2017-03-27 07:24:57 -070057 rtc::CritScope cs(&crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000058 if (!jitter_buffer_.Running()) {
59 jitter_buffer_.Start();
60 } else {
61 jitter_buffer_.Flush();
62 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000063}
64
Johan Ahlers95348f72016-06-28 11:11:28 +020065int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000066 // Insert the packet into the jitter buffer. The packet can either be empty or
67 // contain media at this point.
68 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -080069 const VCMFrameBufferEnum ret =
70 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000071 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000072 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000073 } else if (ret == kFlushIndicator) {
74 return VCM_FLUSH_INDICATOR;
75 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000076 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000077 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000078 if (ret == kCompleteSession && !retransmitted) {
79 // We don't want to include timestamps which have suffered from
80 // retransmission here, since we compensate with extra retransmission
81 // delay within the jitter estimate.
82 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
83 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000084 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000085}
86
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +000087void VCMReceiver::TriggerDecoderShutdown() {
88 jitter_buffer_.Stop();
89 render_wait_event_->Set();
90}
91
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000092VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
perkj796cfaf2015-12-10 09:27:38 -080093 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000094 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000095 uint32_t frame_timestamp = 0;
isheriff6b4b5f32016-06-08 00:24:21 -070096 int min_playout_delay_ms = -1;
97 int max_playout_delay_ms = -1;
Johan Ahlers31b2ec42016-06-28 13:32:49 +020098 int64_t render_time_ms = 0;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000099 // Exhaust wait time to get a complete frame for decoding.
isheriff6b4b5f32016-06-08 00:24:21 -0700100 VCMEncodedFrame* found_frame =
101 jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000102
isheriff6b4b5f32016-06-08 00:24:21 -0700103 if (found_frame) {
Niels Möller23775882018-08-16 10:24:12 +0200104 frame_timestamp = found_frame->Timestamp();
isheriff6b4b5f32016-06-08 00:24:21 -0700105 min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
106 max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
107 } else {
Niels Möller375b3462019-01-10 15:35:56 +0100108 return nullptr;
isheriff6b4b5f32016-06-08 00:24:21 -0700109 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000110
isheriff6b4b5f32016-06-08 00:24:21 -0700111 if (min_playout_delay_ms >= 0)
112 timing_->set_min_playout_delay(min_playout_delay_ms);
113
114 if (max_playout_delay_ms >= 0)
115 timing_->set_max_playout_delay(max_playout_delay_ms);
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000116
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000117 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000118 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000119 const int64_t now_ms = clock_->TimeInMilliseconds();
120 timing_->UpdateCurrentDelay(frame_timestamp);
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200121 render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000122 // Check render timing.
123 bool timing_error = false;
124 // Assume that render timing errors are due to changes in the video stream.
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200125 if (render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000126 timing_error = true;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200127 } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
128 int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
Mirko Bonadei675513b2017-11-09 11:09:25 +0100129 RTC_LOG(LS_WARNING)
130 << "A frame about to be decoded is out of the configured "
131 << "delay bounds (" << frame_delay << " > " << max_video_delay_ms_
132 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000133 timing_error = true;
134 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
135 max_video_delay_ms_) {
Mirko Bonadei675513b2017-11-09 11:09:25 +0100136 RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
137 << max_video_delay_ms_
138 << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000139 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000140 }
141
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000142 if (timing_error) {
143 // Timing error => reset timing and flush the jitter buffer.
144 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000145 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000146 return NULL;
147 }
148
perkj796cfaf2015-12-10 09:27:38 -0800149 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000150 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800151 const int32_t available_wait_time =
152 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000153 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800154 uint16_t new_max_wait_time =
155 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
Ilya Nikolaevskiy8c4fe162018-02-27 15:49:47 +0100156 uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
157 timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds()));
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000158 if (new_max_wait_time < wait_time_ms) {
159 // We're not allowed to wait until the frame is supposed to be rendered,
160 // waiting as long as we're allowed to avoid busy looping, and then return
161 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700162 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000163 return NULL;
164 }
165 // Wait until it's time to render.
166 render_wait_event_->Wait(wait_time_ms);
167 }
168
169 // Extract the frame from the jitter buffer and set the render time.
170 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000171 if (frame == NULL) {
172 return NULL;
173 }
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200174 frame->SetRenderTime(render_time_ms);
Niels Möller23775882018-08-16 10:24:12 +0200175 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->Timestamp(), "SetRenderTS",
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200176 "render_time", frame->RenderTimeMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000177 if (!frame->Complete()) {
178 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000179 bool retransmitted = false;
180 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000181 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000182 if (last_packet_time_ms >= 0 && !retransmitted) {
183 // We don't want to include timestamps which have suffered from
184 // retransmission here, since we compensate with extra retransmission
185 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000186 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000187 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000188 }
189 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000190}
191
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000192void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
193 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000194}
195
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000196void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000197 int max_packet_age_to_nack,
198 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800199 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000200 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000201}
202
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700203std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
204 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000205}
206
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000207} // namespace webrtc