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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>
pbos@webrtc.org3f655aa2014-03-18 11:10:11 +000017
pbos854e84c2015-11-16 16:39:06 -080018#include "webrtc/base/logging.h"
tommie4f96502015-10-20 23:00:48 -070019#include "webrtc/base/trace_event.h"
Henrik Kjellander2557b862015-11-18 22:00:21 +010020#include "webrtc/modules/video_coding/encoded_frame.h"
21#include "webrtc/modules/video_coding/internal_defines.h"
22#include "webrtc/modules/video_coding/media_opt_util.h"
Henrik Kjellander98f53512015-10-28 18:17:40 +010023#include "webrtc/system_wrappers/include/clock.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000024
niklase@google.com470e71d2011-07-07 08:21:25 +000025namespace webrtc {
26
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000027enum { kMaxReceiverDelayMs = 10000 };
28
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000029VCMReceiver::VCMReceiver(VCMTiming* timing,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000030 Clock* clock,
Wan-Teh Chang92d94892015-05-28 13:36:06 -070031 EventFactory* event_factory)
Qiang Chend4cec152015-06-19 09:17:00 -070032 : VCMReceiver(timing,
33 clock,
34 rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent()),
35 rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent())) {
36}
37
38VCMReceiver::VCMReceiver(VCMTiming* timing,
39 Clock* clock,
40 rtc::scoped_ptr<EventWrapper> receiver_event,
41 rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000042 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000043 clock_(clock),
kwiberg0eb15ed2015-12-17 03:04:15 -080044 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();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000053 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000054}
55
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000056void VCMReceiver::Reset() {
57 CriticalSectionScoped cs(crit_sect_);
58 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
pkasting@chromium.org16825b12015-01-12 21:51:21 +000065void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000066 jitter_buffer_.UpdateRtt(rtt);
67}
68
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000069int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
70 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000071 uint16_t frame_height) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000072 // Insert the packet into the jitter buffer. The packet can either be empty or
73 // contain media at this point.
74 bool retransmitted = false;
75 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
76 &retransmitted);
77 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000078 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000079 } else if (ret == kFlushIndicator) {
80 return VCM_FLUSH_INDICATOR;
81 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000082 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000083 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000084 if (ret == kCompleteSession && !retransmitted) {
85 // We don't want to include timestamps which have suffered from
86 // retransmission here, since we compensate with extra retransmission
87 // delay within the jitter estimate.
88 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
89 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000090 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000091}
92
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +000093void VCMReceiver::TriggerDecoderShutdown() {
94 jitter_buffer_.Stop();
95 render_wait_event_->Set();
96}
97
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000098VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
99 int64_t& next_render_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800100 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000101 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000102 uint32_t frame_timestamp = 0;
103 // Exhaust wait time to get a complete frame for decoding.
104 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
105 max_wait_time_ms, &frame_timestamp);
106
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000107 if (!found_frame)
108 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000109
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000110 if (!found_frame)
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000111 return NULL;
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000112
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000113 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000114 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000115 const int64_t now_ms = clock_->TimeInMilliseconds();
116 timing_->UpdateCurrentDelay(frame_timestamp);
117 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
118 // Check render timing.
119 bool timing_error = false;
120 // Assume that render timing errors are due to changes in the video stream.
121 if (next_render_time_ms < 0) {
122 timing_error = true;
pbos@webrtc.orgb5f30292014-03-13 08:53:39 +0000123 } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000124 int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
125 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
126 << "delay bounds (" << frame_delay << " > "
127 << max_video_delay_ms_
128 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000129 timing_error = true;
130 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
131 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000132 LOG(LS_WARNING) << "The video target delay has grown larger than "
133 << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000134 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000135 }
136
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000137 if (timing_error) {
138 // Timing error => reset timing and flush the jitter buffer.
139 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000140 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000141 return NULL;
142 }
143
perkj796cfaf2015-12-10 09:27:38 -0800144 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000145 // Decode frame as close as possible to the render timestamp.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000146 const int32_t available_wait_time = max_wait_time_ms -
147 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
148 uint16_t new_max_wait_time = static_cast<uint16_t>(
149 VCM_MAX(available_wait_time, 0));
150 uint32_t wait_time_ms = timing_->MaxWaitingTime(
151 next_render_time_ms, clock_->TimeInMilliseconds());
152 if (new_max_wait_time < wait_time_ms) {
153 // We're not allowed to wait until the frame is supposed to be rendered,
154 // waiting as long as we're allowed to avoid busy looping, and then return
155 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700156 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000157 return NULL;
158 }
159 // Wait until it's time to render.
160 render_wait_event_->Wait(wait_time_ms);
161 }
162
163 // Extract the frame from the jitter buffer and set the render time.
164 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000165 if (frame == NULL) {
166 return NULL;
167 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000168 frame->SetRenderTime(next_render_time_ms);
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000169 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
170 "SetRenderTS", "render_time", next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000171 if (!frame->Complete()) {
172 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000173 bool retransmitted = false;
174 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000175 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000176 if (last_packet_time_ms >= 0 && !retransmitted) {
177 // We don't want to include timestamps which have suffered from
178 // retransmission here, since we compensate with extra retransmission
179 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000180 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000181 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000182 }
183 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000184}
185
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000186void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
187 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000188}
189
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000190void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
191 uint32_t* framerate) {
192 assert(bitrate);
193 assert(framerate);
194 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000195}
196
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000197uint32_t VCMReceiver::DiscardedPackets() const {
198 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000199}
200
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000201void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000202 int64_t low_rtt_nack_threshold_ms,
203 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000204 CriticalSectionScoped cs(crit_sect_);
205 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000206 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
207 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000208}
209
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000210void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000211 int max_packet_age_to_nack,
212 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000213 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000214 max_packet_age_to_nack,
215 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000216}
217
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000218VCMNackMode VCMReceiver::NackMode() const {
219 CriticalSectionScoped cs(crit_sect_);
220 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000221}
222
Wan-Teh Changb1825a42015-06-03 15:03:35 -0700223std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
224 return jitter_buffer_.GetNackList(request_key_frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000225}
226
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000227void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
228 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000229}
230
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000231VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000232 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000233}
234
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000235int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
236 CriticalSectionScoped cs(crit_sect_);
237 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
238 return -1;
239 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000240 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000241 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000242 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000243 return 0;
244}
245
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000246int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000247 uint32_t timestamp_start = 0u;
248 uint32_t timestamp_end = 0u;
249 // Render timestamps are computed just prior to decoding. Therefore this is
250 // only an estimate based on frames' timestamps and current timing state.
251 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
252 if (timestamp_start == timestamp_end) {
253 return 0;
254 }
255 // Update timing.
256 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000257 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000258 // Get render timestamps.
259 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
260 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
261 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000262}
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000263
pbos@webrtc.org55707692014-12-19 15:45:03 +0000264void VCMReceiver::RegisterStatsCallback(
265 VCMReceiveStatisticsCallback* callback) {
266 jitter_buffer_.RegisterStatsCallback(callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000267}
268
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000269} // namespace webrtc