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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)
Qiang Chend4cec152015-06-19 09:17:00 -070045 : VCMReceiver(timing,
46 clock,
kwiberg3f55dea2016-02-29 05:51:59 -080047 std::unique_ptr<EventWrapper>(event_factory->CreateEvent()),
philipel83f831a2016-03-12 03:30:23 -080048 std::unique_ptr<EventWrapper>(event_factory->CreateEvent()),
49 nack_sender,
50 keyframe_request_sender) {}
Qiang Chend4cec152015-06-19 09:17:00 -070051
52VCMReceiver::VCMReceiver(VCMTiming* timing,
53 Clock* clock,
kwiberg3f55dea2016-02-29 05:51:59 -080054 std::unique_ptr<EventWrapper> receiver_event,
55 std::unique_ptr<EventWrapper> jitter_buffer_event)
philipel83f831a2016-03-12 03:30:23 -080056 : VCMReceiver::VCMReceiver(timing,
57 clock,
58 std::move(receiver_event),
59 std::move(jitter_buffer_event),
60 nullptr, // NackSender
61 nullptr) // KeyframeRequestSender
62{}
63
64VCMReceiver::VCMReceiver(VCMTiming* timing,
65 Clock* clock,
66 std::unique_ptr<EventWrapper> receiver_event,
67 std::unique_ptr<EventWrapper> jitter_buffer_event,
68 NackSender* nack_sender,
69 KeyFrameRequestSender* keyframe_request_sender)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000070 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000071 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();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000084 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000085}
86
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000087void VCMReceiver::Reset() {
88 CriticalSectionScoped cs(crit_sect_);
89 if (!jitter_buffer_.Running()) {
90 jitter_buffer_.Start();
91 } else {
92 jitter_buffer_.Flush();
93 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000094}
95
pkasting@chromium.org16825b12015-01-12 21:51:21 +000096void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000097 jitter_buffer_.UpdateRtt(rtt);
98}
99
philipel83f831a2016-03-12 03:30:23 -0800100int64_t VCMReceiver::TimeUntilNextProcess() {
101 return jitter_buffer_.TimeUntilNextProcess();
102}
103
104void VCMReceiver::Process() {
105 jitter_buffer_.Process();
106}
107
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000108int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
109 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000110 uint16_t frame_height) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000111 // Insert the packet into the jitter buffer. The packet can either be empty or
112 // contain media at this point.
113 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -0800114 const VCMFrameBufferEnum ret =
115 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000116 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000117 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000118 } else if (ret == kFlushIndicator) {
119 return VCM_FLUSH_INDICATOR;
120 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000121 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000122 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000123 if (ret == kCompleteSession && !retransmitted) {
124 // We don't want to include timestamps which have suffered from
125 // retransmission here, since we compensate with extra retransmission
126 // delay within the jitter estimate.
127 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
128 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000129 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000130}
131
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +0000132void VCMReceiver::TriggerDecoderShutdown() {
133 jitter_buffer_.Stop();
134 render_wait_event_->Set();
135}
136
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000137VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
philipel9d3ab612015-12-21 04:12:39 -0800138 int64_t* next_render_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800139 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000140 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000141 uint32_t frame_timestamp = 0;
142 // Exhaust wait time to get a complete frame for decoding.
philipel9d3ab612015-12-21 04:12:39 -0800143 bool found_frame =
144 jitter_buffer_.NextCompleteTimestamp(max_wait_time_ms, &frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000145
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000146 if (!found_frame)
147 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000148
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000149 if (!found_frame)
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000150 return NULL;
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000151
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000152 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000153 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000154 const int64_t now_ms = clock_->TimeInMilliseconds();
155 timing_->UpdateCurrentDelay(frame_timestamp);
philipel9d3ab612015-12-21 04:12:39 -0800156 *next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000157 // Check render timing.
158 bool timing_error = false;
159 // Assume that render timing errors are due to changes in the video stream.
philipel9d3ab612015-12-21 04:12:39 -0800160 if (*next_render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000161 timing_error = true;
philipel9d3ab612015-12-21 04:12:39 -0800162 } else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
163 int frame_delay = static_cast<int>(std::abs(*next_render_time_ms - now_ms));
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000164 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
165 << "delay bounds (" << frame_delay << " > "
166 << max_video_delay_ms_
167 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000168 timing_error = true;
169 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
170 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000171 LOG(LS_WARNING) << "The video target delay has grown larger than "
172 << max_video_delay_ms_ << " 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));
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000190 uint32_t wait_time_ms = timing_->MaxWaitingTime(
philipel9d3ab612015-12-21 04:12:39 -0800191 *next_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 }
philipel9d3ab612015-12-21 04:12:39 -0800208 frame->SetRenderTime(*next_render_time_ms);
209 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
210 "render_time", *next_render_time_ms);
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.org1ea4b502013-01-07 08:49:41 +0000236uint32_t VCMReceiver::DiscardedPackets() const {
237 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000238}
239
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000240void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000241 int64_t low_rtt_nack_threshold_ms,
242 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000243 CriticalSectionScoped cs(crit_sect_);
244 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000245 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
246 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000247}
248
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000249void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000250 int max_packet_age_to_nack,
251 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800252 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000253 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000254}
255
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000256VCMNackMode VCMReceiver::NackMode() const {
257 CriticalSectionScoped cs(crit_sect_);
258 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000259}
260
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700261std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
262 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000263}
264
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000265void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
266 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000267}
268
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000269VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000270 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000271}
272
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000273int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
274 CriticalSectionScoped cs(crit_sect_);
275 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
276 return -1;
277 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000278 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000279 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000280 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000281 return 0;
282}
283
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000284int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000285 uint32_t timestamp_start = 0u;
286 uint32_t timestamp_end = 0u;
287 // Render timestamps are computed just prior to decoding. Therefore this is
288 // only an estimate based on frames' timestamps and current timing state.
289 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
290 if (timestamp_start == timestamp_end) {
291 return 0;
292 }
293 // Update timing.
294 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000295 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000296 // Get render timestamps.
297 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
298 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
299 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000300}
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000301
pbos@webrtc.org55707692014-12-19 15:45:03 +0000302void VCMReceiver::RegisterStatsCallback(
303 VCMReceiveStatisticsCallback* callback) {
304 jitter_buffer_.RegisterStatsCallback(callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000305}
306
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000307} // namespace webrtc