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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
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000011#include "webrtc/modules/video_coding/main/source/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>
16
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000017#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
18#include "webrtc/modules/video_coding/main/source/internal_defines.h"
19#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000020#include "webrtc/system_wrappers/interface/clock.h"
stefan@webrtc.org34c5da62014-04-11 14:08:35 +000021#include "webrtc/system_wrappers/interface/logging.h"
hclam@chromium.org806dc3b2013-04-09 19:54:10 +000022#include "webrtc/system_wrappers/interface/trace_event.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000023
niklase@google.com470e71d2011-07-07 08:21:25 +000024namespace webrtc {
25
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000026enum { kMaxReceiverDelayMs = 10000 };
27
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000028VCMReceiver::VCMReceiver(VCMTiming* timing,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000029 Clock* clock,
Wan-Teh Chang92d94892015-05-28 13:36:06 -070030 EventFactory* event_factory)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000031 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000032 clock_(clock),
stefan@webrtc.org34c5da62014-04-11 14:08:35 +000033 jitter_buffer_(clock_, event_factory),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000034 timing_(timing),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000035 render_wait_event_(event_factory->CreateEvent()),
Peter Boström5464a6e2015-04-21 16:35:50 +020036 max_video_delay_ms_(kMaxVideoDelayMs) {
37 Reset();
38}
niklase@google.com470e71d2011-07-07 08:21:25 +000039
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000040VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000041 render_wait_event_->Set();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000042 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000043}
44
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000045void VCMReceiver::Reset() {
46 CriticalSectionScoped cs(crit_sect_);
47 if (!jitter_buffer_.Running()) {
48 jitter_buffer_.Start();
49 } else {
50 jitter_buffer_.Flush();
51 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000052}
53
pkasting@chromium.org16825b12015-01-12 21:51:21 +000054void VCMReceiver::UpdateRtt(int64_t rtt) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000055 jitter_buffer_.UpdateRtt(rtt);
56}
57
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000058int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
59 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000060 uint16_t frame_height) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000061 // Insert the packet into the jitter buffer. The packet can either be empty or
62 // contain media at this point.
63 bool retransmitted = false;
64 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
65 &retransmitted);
66 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000067 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000068 } else if (ret == kFlushIndicator) {
69 return VCM_FLUSH_INDICATOR;
70 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000071 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000072 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000073 if (ret == kCompleteSession && !retransmitted) {
74 // We don't want to include timestamps which have suffered from
75 // retransmission here, since we compensate with extra retransmission
76 // delay within the jitter estimate.
77 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
78 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000079 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000080}
81
pbos@webrtc.org4dd40d62015-02-17 13:22:43 +000082void VCMReceiver::TriggerDecoderShutdown() {
83 jitter_buffer_.Stop();
84 render_wait_event_->Set();
85}
86
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000087VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
88 int64_t& next_render_time_ms,
89 bool render_timing) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000090 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000091 uint32_t frame_timestamp = 0;
92 // Exhaust wait time to get a complete frame for decoding.
93 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
94 max_wait_time_ms, &frame_timestamp);
95
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000096 if (!found_frame)
97 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000098
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000099 if (!found_frame)
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000100 return NULL;
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000101
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000102 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000103 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000104 const int64_t now_ms = clock_->TimeInMilliseconds();
105 timing_->UpdateCurrentDelay(frame_timestamp);
106 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
107 // Check render timing.
108 bool timing_error = false;
109 // Assume that render timing errors are due to changes in the video stream.
110 if (next_render_time_ms < 0) {
111 timing_error = true;
pbos@webrtc.orgb5f30292014-03-13 08:53:39 +0000112 } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000113 int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
114 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
115 << "delay bounds (" << frame_delay << " > "
116 << max_video_delay_ms_
117 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000118 timing_error = true;
119 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
120 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000121 LOG(LS_WARNING) << "The video target delay has grown larger than "
122 << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000123 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000124 }
125
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000126 if (timing_error) {
127 // Timing error => reset timing and flush the jitter buffer.
128 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000129 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000130 return NULL;
131 }
132
133 if (!render_timing) {
134 // Decode frame as close as possible to the render timestamp.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000135 const int32_t available_wait_time = max_wait_time_ms -
136 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
137 uint16_t new_max_wait_time = static_cast<uint16_t>(
138 VCM_MAX(available_wait_time, 0));
139 uint32_t wait_time_ms = timing_->MaxWaitingTime(
140 next_render_time_ms, clock_->TimeInMilliseconds());
141 if (new_max_wait_time < wait_time_ms) {
142 // We're not allowed to wait until the frame is supposed to be rendered,
143 // waiting as long as we're allowed to avoid busy looping, and then return
144 // NULL. Next call to this function might return the frame.
Niklas Enbomb4c5eaa2015-06-03 09:34:25 -0700145 render_wait_event_->Wait(new_max_wait_time);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000146 return NULL;
147 }
148 // Wait until it's time to render.
149 render_wait_event_->Wait(wait_time_ms);
150 }
151
152 // Extract the frame from the jitter buffer and set the render time.
153 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000154 if (frame == NULL) {
155 return NULL;
156 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000157 frame->SetRenderTime(next_render_time_ms);
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000158 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
159 "SetRenderTS", "render_time", next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000160 if (!frame->Complete()) {
161 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000162 bool retransmitted = false;
163 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000164 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000165 if (last_packet_time_ms >= 0 && !retransmitted) {
166 // We don't want to include timestamps which have suffered from
167 // retransmission here, since we compensate with extra retransmission
168 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000169 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000170 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000171 }
172 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000173}
174
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000175void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
176 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000177}
178
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000179void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
180 uint32_t* framerate) {
181 assert(bitrate);
182 assert(framerate);
183 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000184}
185
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000186uint32_t VCMReceiver::DiscardedPackets() const {
187 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000188}
189
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000190void VCMReceiver::SetNackMode(VCMNackMode nackMode,
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000191 int64_t low_rtt_nack_threshold_ms,
192 int64_t high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000193 CriticalSectionScoped cs(crit_sect_);
194 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000195 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
196 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000197}
198
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000199void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000200 int max_packet_age_to_nack,
201 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000202 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000203 max_packet_age_to_nack,
204 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000205}
206
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000207VCMNackMode VCMReceiver::NackMode() const {
208 CriticalSectionScoped cs(crit_sect_);
209 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000210}
211
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000212VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000213 uint16_t size,
214 uint16_t* nack_list_length) {
215 bool request_key_frame = false;
216 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
217 nack_list_length, &request_key_frame);
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000218 assert(*nack_list_length <= size);
Wan-Teh Chang92d94892015-05-28 13:36:06 -0700219 if (*nack_list_length > size) {
220 *nack_list_length = size;
221 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000222 if (internal_nack_list != NULL && *nack_list_length > 0) {
223 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000224 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000225 if (request_key_frame) {
226 return kNackKeyFrameRequest;
227 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000228 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000229}
230
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000231void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
232 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000233}
234
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000235VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000236 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000237}
238
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000239int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
240 CriticalSectionScoped cs(crit_sect_);
241 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
242 return -1;
243 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000244 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000245 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000246 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000247 return 0;
248}
249
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000250int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000251 uint32_t timestamp_start = 0u;
252 uint32_t timestamp_end = 0u;
253 // Render timestamps are computed just prior to decoding. Therefore this is
254 // only an estimate based on frames' timestamps and current timing state.
255 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
256 if (timestamp_start == timestamp_end) {
257 return 0;
258 }
259 // Update timing.
260 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000261 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000262 // Get render timestamps.
263 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
264 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
265 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000266}
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000267
pbos@webrtc.org55707692014-12-19 15:45:03 +0000268void VCMReceiver::RegisterStatsCallback(
269 VCMReceiveStatisticsCallback* callback) {
270 jitter_buffer_.RegisterStatsCallback(callback);
pbos@webrtc.orgce4e9a32014-12-18 13:50:16 +0000271}
272
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000273} // namespace webrtc