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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
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000015#include "webrtc/modules/video_coding/main/interface/video_coding.h"
16#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
17#include "webrtc/modules/video_coding/main/source/internal_defines.h"
18#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000019#include "webrtc/system_wrappers/interface/clock.h"
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000020#include "webrtc/system_wrappers/interface/trace.h"
hclam@chromium.org806dc3b2013-04-09 19:54:10 +000021#include "webrtc/system_wrappers/interface/trace_event.h"
stefan@webrtc.org91c63082012-01-31 10:49:08 +000022
niklase@google.com470e71d2011-07-07 08:21:25 +000023namespace webrtc {
24
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000025enum { kMaxReceiverDelayMs = 10000 };
26
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000027VCMReceiver::VCMReceiver(VCMTiming* timing,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000028 Clock* clock,
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000029 EventFactory* event_factory,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000030 int32_t vcm_id,
31 int32_t receiver_id,
niklase@google.com470e71d2011-07-07 08:21:25 +000032 bool master)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000033 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
34 vcm_id_(vcm_id),
35 clock_(clock),
36 receiver_id_(receiver_id),
37 master_(master),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000038 jitter_buffer_(clock_, event_factory, vcm_id, receiver_id, master),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000039 timing_(timing),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000040 render_wait_event_(event_factory->CreateEvent()),
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000041 state_(kPassive),
42 max_video_delay_ms_(kMaxVideoDelayMs) {}
niklase@google.com470e71d2011-07-07 08:21:25 +000043
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000044VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000045 render_wait_event_->Set();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000046 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000047}
48
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000049void VCMReceiver::Reset() {
50 CriticalSectionScoped cs(crit_sect_);
51 if (!jitter_buffer_.Running()) {
52 jitter_buffer_.Start();
53 } else {
54 jitter_buffer_.Flush();
55 }
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000056 render_wait_event_->Reset();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000057 if (master_) {
58 state_ = kReceiving;
59 } else {
60 state_ = kPassive;
61 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000062}
63
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000064int32_t VCMReceiver::Initialize() {
65 CriticalSectionScoped cs(crit_sect_);
66 Reset();
67 if (!master_) {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000068 SetNackMode(kNoNack, -1, -1);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000069 }
70 return VCM_OK;
71}
72
73void VCMReceiver::UpdateRtt(uint32_t rtt) {
74 jitter_buffer_.UpdateRtt(rtt);
75}
76
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000077int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
78 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000079 uint16_t frame_height) {
hclam@chromium.org8c49c1e2013-05-22 21:18:59 +000080 WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCoding,
81 VCMId(vcm_id_, receiver_id_),
82 "Inserting key frame packet seqnum=%u, timestamp=%u",
83 packet.seqNum, packet.timestamp);
84
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000085 // Insert the packet into the jitter buffer. The packet can either be empty or
86 // contain media at this point.
87 bool retransmitted = false;
88 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
89 &retransmitted);
90 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000091 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000092 } else if (ret == kFlushIndicator) {
93 return VCM_FLUSH_INDICATOR;
94 } else if (ret < 0) {
95 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
96 VCMId(vcm_id_, receiver_id_),
97 "Error inserting packet seqnum=%u, timestamp=%u",
98 packet.seqNum, packet.timestamp);
99 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000100 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000101 if (ret == kCompleteSession && !retransmitted) {
102 // We don't want to include timestamps which have suffered from
103 // retransmission here, since we compensate with extra retransmission
104 // delay within the jitter estimate.
105 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
106 }
107 if (master_) {
108 // Only trace the primary receiver to make it possible to parse and plot
109 // the trace file.
110 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
111 VCMId(vcm_id_, receiver_id_),
112 "Packet seqnum=%u timestamp=%u inserted at %u",
113 packet.seqNum, packet.timestamp,
114 MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000115 }
116 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000117}
118
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000119VCMEncodedFrame* VCMReceiver::FrameForDecoding(
120 uint16_t max_wait_time_ms,
121 int64_t& next_render_time_ms,
122 bool render_timing,
123 VCMReceiver* dual_receiver) {
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000124 TRACE_EVENT0("webrtc", "Recv::FrameForDecoding");
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000125 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000126 uint32_t frame_timestamp = 0;
127 // Exhaust wait time to get a complete frame for decoding.
128 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
129 max_wait_time_ms, &frame_timestamp);
130
131 if (!found_frame) {
132 // Get an incomplete frame when enabled.
133 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
134 dual_receiver->State() == kPassive &&
135 dual_receiver->NackMode() == kNack);
136 if (dual_receiver_enabled_and_passive &&
137 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
138 // Jitter buffer state might get corrupt with this frame.
139 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
140 }
141 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(
142 &frame_timestamp);
143 }
144
145 if (!found_frame) {
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000146 return NULL;
147 }
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000148
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000149 // We have a frame - Set timing and render timestamp.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000150 timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000151 const int64_t now_ms = clock_->TimeInMilliseconds();
152 timing_->UpdateCurrentDelay(frame_timestamp);
153 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
154 // Check render timing.
155 bool timing_error = false;
156 // Assume that render timing errors are due to changes in the video stream.
157 if (next_render_time_ms < 0) {
158 timing_error = true;
159 } else if (next_render_time_ms < now_ms - max_video_delay_ms_) {
160 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
161 VCMId(vcm_id_, receiver_id_),
162 "This frame should have been rendered more than %u ms ago."
163 "Flushing jitter buffer and resetting timing.",
164 max_video_delay_ms_);
165 timing_error = true;
166 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
167 max_video_delay_ms_) {
168 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
169 VCMId(vcm_id_, receiver_id_),
170 "More than %u ms target delay. Flushing jitter buffer and"
171 "resetting timing.", max_video_delay_ms_);
172 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000173 }
174
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000175 if (timing_error) {
176 // Timing error => reset timing and flush the jitter buffer.
177 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000178 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000179 return NULL;
180 }
181
182 if (!render_timing) {
183 // Decode frame as close as possible to the render timestamp.
184 TRACE_EVENT0("webrtc", "FrameForRendering");
185 const int32_t available_wait_time = max_wait_time_ms -
186 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
187 uint16_t new_max_wait_time = static_cast<uint16_t>(
188 VCM_MAX(available_wait_time, 0));
189 uint32_t wait_time_ms = timing_->MaxWaitingTime(
190 next_render_time_ms, clock_->TimeInMilliseconds());
191 if (new_max_wait_time < wait_time_ms) {
192 // We're not allowed to wait until the frame is supposed to be rendered,
193 // waiting as long as we're allowed to avoid busy looping, and then return
194 // NULL. Next call to this function might return the frame.
195 render_wait_event_->Wait(max_wait_time_ms);
196 return NULL;
197 }
198 // Wait until it's time to render.
199 render_wait_event_->Wait(wait_time_ms);
200 }
201
202 // Extract the frame from the jitter buffer and set the render time.
203 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000204 if (frame == NULL) {
205 return NULL;
206 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000207 frame->SetRenderTime(next_render_time_ms);
208 if (dual_receiver != NULL) {
209 dual_receiver->UpdateState(*frame);
210 }
211 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
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000230void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
231 uint32_t* framerate) {
232 assert(bitrate);
233 assert(framerate);
234 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000235}
236
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000237void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
238 assert(frame_count);
239 jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames,
240 &frame_count->numKeyFrames);
niklase@google.com470e71d2011-07-07 08:21:25 +0000241}
242
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000243uint32_t VCMReceiver::DiscardedPackets() const {
244 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000245}
246
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000247void VCMReceiver::SetNackMode(VCMNackMode nackMode,
248 int low_rtt_nack_threshold_ms,
249 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000250 CriticalSectionScoped cs(crit_sect_);
251 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000252 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
253 high_rtt_nack_threshold_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000254 if (!master_) {
255 state_ = kPassive; // The dual decoder defaults to passive.
256 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000257}
258
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000259void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000260 int max_packet_age_to_nack,
261 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000262 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000263 max_packet_age_to_nack,
264 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000265}
266
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000267VCMNackMode VCMReceiver::NackMode() const {
268 CriticalSectionScoped cs(crit_sect_);
269 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000270}
271
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000272VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000273 uint16_t size,
274 uint16_t* nack_list_length) {
275 bool request_key_frame = false;
276 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
277 nack_list_length, &request_key_frame);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000278 if (*nack_list_length > size) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000279 *nack_list_length = 0;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000280 return kNackNeedMoreMemory;
281 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000282 if (internal_nack_list != NULL && *nack_list_length > 0) {
283 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000284 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000285 if (request_key_frame) {
286 return kNackKeyFrameRequest;
287 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000288 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000289}
290
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000291// Decide whether we should change decoder state. This should be done if the
292// dual decoder has caught up with the decoder decoding with packet losses.
293bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
294 VCMReceiver& dual_receiver) const {
295 if (dual_frame == NULL) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000296 return false;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000297 }
298 if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
299 dual_receiver.UpdateState(kWaitForPrimaryDecode);
300 return true;
301 }
302 return false;
niklase@google.com470e71d2011-07-07 08:21:25 +0000303}
304
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000305void VCMReceiver::CopyJitterBufferStateFromReceiver(
306 const VCMReceiver& receiver) {
307 jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000308}
309
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000310VCMReceiverState VCMReceiver::State() const {
311 CriticalSectionScoped cs(crit_sect_);
312 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000313}
314
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000315void VCMReceiver::SetDecodeWithErrors(bool enable){
316 CriticalSectionScoped cs(crit_sect_);
317 jitter_buffer_.DecodeWithErrors(enable);
318}
319
320bool VCMReceiver::DecodeWithErrors() const {
321 CriticalSectionScoped cs(crit_sect_);
322 return jitter_buffer_.decode_with_errors();
323}
324
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000325int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
326 CriticalSectionScoped cs(crit_sect_);
327 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
328 return -1;
329 }
mikhal@webrtc.org6faba6e2013-04-30 15:39:34 +0000330 jitter_buffer_.SetMaxJitterEstimate(desired_delay_ms > 0);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000331 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000332 // Initializing timing to the desired delay.
mikhal@webrtc.org6faba6e2013-04-30 15:39:34 +0000333 timing_->SetMinimumTotalDelay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000334 return 0;
335}
336
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000337int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000338 uint32_t timestamp_start = 0u;
339 uint32_t timestamp_end = 0u;
340 // Render timestamps are computed just prior to decoding. Therefore this is
341 // only an estimate based on frames' timestamps and current timing state.
342 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
343 if (timestamp_start == timestamp_end) {
344 return 0;
345 }
346 // Update timing.
347 const int64_t now_ms = clock_->TimeInMilliseconds();
348 timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
349 // Get render timestamps.
350 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
351 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
352 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000353}
354
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000355void VCMReceiver::UpdateState(VCMReceiverState new_state) {
356 CriticalSectionScoped cs(crit_sect_);
357 assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
358 state_ = new_state;
niklase@google.com470e71d2011-07-07 08:21:25 +0000359}
360
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000361void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
362 if (jitter_buffer_.nack_mode() == kNoNack) {
363 // Dual decoder mode has not been enabled.
364 return;
365 }
366 // Update the dual receiver state.
367 if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
368 UpdateState(kPassive);
369 }
370 if (State() == kWaitForPrimaryDecode &&
371 frame.Complete() && !frame.MissingFrame()) {
372 UpdateState(kPassive);
373 }
374 if (frame.MissingFrame() || !frame.Complete()) {
375 // State was corrupted, enable dual receiver.
376 UpdateState(kReceiving);
377 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000378}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000379} // namespace webrtc