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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)
Qiang Chend4cec152015-06-19 09:17:00 -070033 : VCMReceiver(timing,
34 clock,
kwiberg3f55dea2016-02-29 05:51:59 -080035 std::unique_ptr<EventWrapper>(event_factory->CreateEvent()),
36 std::unique_ptr<EventWrapper>(event_factory->CreateEvent())) {
Qiang Chend4cec152015-06-19 09:17:00 -070037}
38
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)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000043 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000044 clock_(clock),
kwiberg0eb15ed2015-12-17 03:04:15 -080045 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();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000054 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000055}
56
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000057void VCMReceiver::Reset() {
58 CriticalSectionScoped cs(crit_sect_);
59 if (!jitter_buffer_.Running()) {
60 jitter_buffer_.Start();
61 } else {
62 jitter_buffer_.Flush();
63 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000064}
65
pkasting@chromium.org16825b12015-01-12 21:51:21 +000066void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000067 jitter_buffer_.UpdateRtt(rtt);
68}
69
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000070int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
71 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000072 uint16_t frame_height) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000073 // Insert the packet into the jitter buffer. The packet can either be empty or
74 // contain media at this point.
75 bool retransmitted = false;
philipel9d3ab612015-12-21 04:12:39 -080076 const VCMFrameBufferEnum ret =
77 jitter_buffer_.InsertPacket(packet, &retransmitted);
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000078 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000079 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000080 } else if (ret == kFlushIndicator) {
81 return VCM_FLUSH_INDICATOR;
82 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000083 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000084 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000085 if (ret == kCompleteSession && !retransmitted) {
86 // We don't want to include timestamps which have suffered from
87 // retransmission here, since we compensate with extra retransmission
88 // delay within the jitter estimate.
89 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
90 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000091 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000092}
93
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +000094void VCMReceiver::TriggerDecoderShutdown() {
95 jitter_buffer_.Stop();
96 render_wait_event_->Set();
97}
98
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000099VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
philipel9d3ab612015-12-21 04:12:39 -0800100 int64_t* next_render_time_ms,
perkj796cfaf2015-12-10 09:27:38 -0800101 bool prefer_late_decoding) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000102 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000103 uint32_t frame_timestamp = 0;
104 // Exhaust wait time to get a complete frame for decoding.
philipel9d3ab612015-12-21 04:12:39 -0800105 bool found_frame =
106 jitter_buffer_.NextCompleteTimestamp(max_wait_time_ms, &frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000107
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000108 if (!found_frame)
109 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000110
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000111 if (!found_frame)
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000112 return NULL;
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000113
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000114 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000115 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000116 const int64_t now_ms = clock_->TimeInMilliseconds();
117 timing_->UpdateCurrentDelay(frame_timestamp);
philipel9d3ab612015-12-21 04:12:39 -0800118 *next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000119 // Check render timing.
120 bool timing_error = false;
121 // Assume that render timing errors are due to changes in the video stream.
philipel9d3ab612015-12-21 04:12:39 -0800122 if (*next_render_time_ms < 0) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000123 timing_error = true;
philipel9d3ab612015-12-21 04:12:39 -0800124 } else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
125 int frame_delay = static_cast<int>(std::abs(*next_render_time_ms - now_ms));
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000126 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
127 << "delay bounds (" << frame_delay << " > "
128 << max_video_delay_ms_
129 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000130 timing_error = true;
131 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
132 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000133 LOG(LS_WARNING) << "The video target delay has grown larger than "
134 << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000135 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000136 }
137
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000138 if (timing_error) {
139 // Timing error => reset timing and flush the jitter buffer.
140 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000141 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000142 return NULL;
143 }
144
perkj796cfaf2015-12-10 09:27:38 -0800145 if (prefer_late_decoding) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000146 // Decode frame as close as possible to the render timestamp.
philipel9d3ab612015-12-21 04:12:39 -0800147 const int32_t available_wait_time =
148 max_wait_time_ms -
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000149 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
philipel9d3ab612015-12-21 04:12:39 -0800150 uint16_t new_max_wait_time =
151 static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000152 uint32_t wait_time_ms = timing_->MaxWaitingTime(
philipel9d3ab612015-12-21 04:12:39 -0800153 *next_render_time_ms, clock_->TimeInMilliseconds());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000154 if (new_max_wait_time < wait_time_ms) {
155 // We're not allowed to wait until the frame is supposed to be rendered,
156 // waiting as long as we're allowed to avoid busy looping, and then return
157 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700158 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000159 return NULL;
160 }
161 // Wait until it's time to render.
162 render_wait_event_->Wait(wait_time_ms);
163 }
164
165 // Extract the frame from the jitter buffer and set the render time.
166 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000167 if (frame == NULL) {
168 return NULL;
169 }
philipel9d3ab612015-12-21 04:12:39 -0800170 frame->SetRenderTime(*next_render_time_ms);
171 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
172 "render_time", *next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000173 if (!frame->Complete()) {
174 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000175 bool retransmitted = false;
176 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000177 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000178 if (last_packet_time_ms >= 0 && !retransmitted) {
179 // We don't want to include timestamps which have suffered from
180 // retransmission here, since we compensate with extra retransmission
181 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000182 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000183 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000184 }
185 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000186}
187
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000188void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
189 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000190}
191
philipel9d3ab612015-12-21 04:12:39 -0800192void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000193 assert(bitrate);
194 assert(framerate);
195 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000196}
197
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000198uint32_t VCMReceiver::DiscardedPackets() const {
199 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000200}
201
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000202void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000203 int64_t low_rtt_nack_threshold_ms,
204 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000205 CriticalSectionScoped cs(crit_sect_);
206 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000207 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
208 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000209}
210
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000211void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000212 int max_packet_age_to_nack,
213 int max_incomplete_time_ms) {
philipel9d3ab612015-12-21 04:12:39 -0800214 jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000215 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