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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
Henrik Kjellander2557b862015-11-18 22:00:21 +010011#include "webrtc/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
pbos854e84c2015-11-16 16:39:06 -080019#include "webrtc/base/logging.h"
tommie4f96502015-10-20 23:00:48 -070020#include "webrtc/base/trace_event.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010021#include "webrtc/modules/video_coding/encoded_frame.h"
22#include "webrtc/modules/video_coding/internal_defines.h"
23#include "webrtc/modules/video_coding/media_opt_util.h"
Henrik Kjellander98f53512015-10-28 18:17:40 +010024#include "webrtc/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)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000075 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000076 clock_(clock),
philipel83f831a2016-03-12 03:30:23 -080077 jitter_buffer_(clock_,
78 std::move(jitter_buffer_event),
79 nack_sender,
80 keyframe_request_sender),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000081 timing_(timing),
kwiberg0eb15ed2015-12-17 03:04:15 -080082 render_wait_event_(std::move(receiver_event)),
Peter Boström5464a6e2015-04-21 16:35:50 +020083 max_video_delay_ms_(kMaxVideoDelayMs) {
84 Reset();
85}
niklase@google.com470e71d2011-07-07 08:21:25 +000086
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000087VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000088 render_wait_event_->Set();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000089 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000090}
91
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000092void VCMReceiver::Reset() {
93 CriticalSectionScoped cs(crit_sect_);
94 if (!jitter_buffer_.Running()) {
95 jitter_buffer_.Start();
96 } else {
97 jitter_buffer_.Flush();
98 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000099}
100
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000101void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000102 jitter_buffer_.UpdateRtt(rtt);
103}
104
Johan Ahlers95348f72016-06-28 11:11:28 +0200105int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000106 // Insert the packet into the jitter buffer. The packet can either be empty or
107 // contain media at this point.
108 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -0800109 const VCMFrameBufferEnum ret =
110 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000111 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000112 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000113 } else if (ret == kFlushIndicator) {
114 return VCM_FLUSH_INDICATOR;
115 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000116 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000117 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000118 if (ret == kCompleteSession && !retransmitted) {
119 // We don't want to include timestamps which have suffered from
120 // retransmission here, since we compensate with extra retransmission
121 // delay within the jitter estimate.
122 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
123 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000124 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000125}
126
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +0000127void VCMReceiver::TriggerDecoderShutdown() {
128 jitter_buffer_.Stop();
129 render_wait_event_->Set();
130}
131
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000132VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800133 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000134 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000135 uint32_t frame_timestamp = 0;
isheriff6b4b5f32016-06-08 00:24:21 -0700136 int min_playout_delay_ms = -1;
137 int max_playout_delay_ms = -1;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200138 int64_t render_time_ms = 0;
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000139 // Exhaust wait time to get a complete frame for decoding.
isheriff6b4b5f32016-06-08 00:24:21 -0700140 VCMEncodedFrame* found_frame =
141 jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000142
isheriff6b4b5f32016-06-08 00:24:21 -0700143 if (found_frame) {
144 frame_timestamp = found_frame->TimeStamp();
145 min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
146 max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
147 } else {
148 if (!jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp))
149 return nullptr;
150 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000151
isheriff6b4b5f32016-06-08 00:24:21 -0700152 if (min_playout_delay_ms >= 0)
153 timing_->set_min_playout_delay(min_playout_delay_ms);
154
155 if (max_playout_delay_ms >= 0)
156 timing_->set_max_playout_delay(max_playout_delay_ms);
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000157
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000158 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000159 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000160 const int64_t now_ms = clock_->TimeInMilliseconds();
161 timing_->UpdateCurrentDelay(frame_timestamp);
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200162 render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000163 // Check render timing.
164 bool timing_error = false;
165 // Assume that render timing errors are due to changes in the video stream.
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200166 if (render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000167 timing_error = true;
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200168 } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
169 int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000170 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
171 << "delay bounds (" << frame_delay << " > "
172 << max_video_delay_ms_
173 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000174 timing_error = true;
175 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
176 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000177 LOG(LS_WARNING) << "The video target delay has grown larger than "
178 << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000179 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000180 }
181
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000182 if (timing_error) {
183 // Timing error => reset timing and flush the jitter buffer.
184 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000185 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000186 return NULL;
187 }
188
perkj796cfaf2015-12-10 09:27:38 -0800189 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000190 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800191 const int32_t available_wait_time =
192 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000193 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800194 uint16_t new_max_wait_time =
195 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200196 uint32_t wait_time_ms =
197 timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000198 if (new_max_wait_time < wait_time_ms) {
199 // We're not allowed to wait until the frame is supposed to be rendered,
200 // waiting as long as we're allowed to avoid busy looping, and then return
201 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700202 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000203 return NULL;
204 }
205 // Wait until it's time to render.
206 render_wait_event_->Wait(wait_time_ms);
207 }
208
209 // Extract the frame from the jitter buffer and set the render time.
210 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000211 if (frame == NULL) {
212 return NULL;
213 }
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200214 frame->SetRenderTime(render_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800215 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
Johan Ahlers31b2ec42016-06-28 13:32:49 +0200216 "render_time", frame->RenderTimeMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000217 if (!frame->Complete()) {
218 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000219 bool retransmitted = false;
220 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000221 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000222 if (last_packet_time_ms >= 0 && !retransmitted) {
223 // We don't want to include timestamps which have suffered from
224 // retransmission here, since we compensate with extra retransmission
225 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000226 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000227 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000228 }
229 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000230}
231
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000232void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
233 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000234}
235
philipel9d3ab612015-12-21 04:12:39 -0800236void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000237 assert(bitrate);
238 assert(framerate);
239 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000240}
241
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000242uint32_t VCMReceiver::DiscardedPackets() const {
243 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000244}
245
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000246void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000247 int64_t low_rtt_nack_threshold_ms,
248 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000249 CriticalSectionScoped cs(crit_sect_);
250 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000251 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
252 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000253}
254
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000255void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000256 int max_packet_age_to_nack,
257 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800258 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000259 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000260}
261
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000262VCMNackMode VCMReceiver::NackMode() const {
263 CriticalSectionScoped cs(crit_sect_);
264 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000265}
266
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700267std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
268 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000269}
270
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000271void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
272 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000273}
274
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000275VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000276 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000277}
278
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000279int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
280 CriticalSectionScoped cs(crit_sect_);
281 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
282 return -1;
283 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000284 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000285 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000286 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000287 return 0;
288}
289
pbos@webrtc.org55707692014-12-19 15:45:03 +0000290void VCMReceiver::RegisterStatsCallback(
291 VCMReceiveStatisticsCallback* callback) {
292 jitter_buffer_.RegisterStatsCallback(callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000293}
294
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000295} // namespace webrtc