blob: 4cbbebd3127c4bf65434df740a18cf824b12bd0c [file] [log] [blame]
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) {
stefan@webrtc.orga7dc37d2013-05-23 07:21:05 +000080 if (packet.frameType == kVideoFrameKey) {
81 WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCoding,
82 VCMId(vcm_id_, receiver_id_),
83 "Inserting key frame packet seqnum=%u, timestamp=%u",
84 packet.seqNum, packet.timestamp);
85 }
hclam@chromium.org8c49c1e2013-05-22 21:18:59 +000086
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000087 // Insert the packet into the jitter buffer. The packet can either be empty or
88 // contain media at this point.
89 bool retransmitted = false;
90 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
91 &retransmitted);
92 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000093 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000094 } else if (ret == kFlushIndicator) {
95 return VCM_FLUSH_INDICATOR;
96 } else if (ret < 0) {
97 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
98 VCMId(vcm_id_, receiver_id_),
99 "Error inserting packet seqnum=%u, timestamp=%u",
100 packet.seqNum, packet.timestamp);
101 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000102 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000103 if (ret == kCompleteSession && !retransmitted) {
104 // We don't want to include timestamps which have suffered from
105 // retransmission here, since we compensate with extra retransmission
106 // delay within the jitter estimate.
107 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
108 }
109 if (master_) {
110 // Only trace the primary receiver to make it possible to parse and plot
111 // the trace file.
112 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
113 VCMId(vcm_id_, receiver_id_),
114 "Packet seqnum=%u timestamp=%u inserted at %u",
115 packet.seqNum, packet.timestamp,
116 MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000117 }
118 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000119}
120
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000121VCMEncodedFrame* VCMReceiver::FrameForDecoding(
122 uint16_t max_wait_time_ms,
123 int64_t& next_render_time_ms,
124 bool render_timing,
125 VCMReceiver* dual_receiver) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000126 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000127 uint32_t frame_timestamp = 0;
128 // Exhaust wait time to get a complete frame for decoding.
129 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
130 max_wait_time_ms, &frame_timestamp);
131
132 if (!found_frame) {
133 // Get an incomplete frame when enabled.
134 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
135 dual_receiver->State() == kPassive &&
136 dual_receiver->NackMode() == kNack);
137 if (dual_receiver_enabled_and_passive &&
138 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
139 // Jitter buffer state might get corrupt with this frame.
140 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
141 }
142 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(
143 &frame_timestamp);
144 }
145
146 if (!found_frame) {
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000147 return NULL;
148 }
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000149
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000150 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000151 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000152 const int64_t now_ms = clock_->TimeInMilliseconds();
153 timing_->UpdateCurrentDelay(frame_timestamp);
154 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
155 // Check render timing.
156 bool timing_error = false;
157 // Assume that render timing errors are due to changes in the video stream.
158 if (next_render_time_ms < 0) {
159 timing_error = true;
160 } else if (next_render_time_ms < now_ms - max_video_delay_ms_) {
161 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
162 VCMId(vcm_id_, receiver_id_),
163 "This frame should have been rendered more than %u ms ago."
164 "Flushing jitter buffer and resetting timing.",
165 max_video_delay_ms_);
166 timing_error = true;
167 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
168 max_video_delay_ms_) {
169 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
170 VCMId(vcm_id_, receiver_id_),
171 "More than %u ms target delay. Flushing jitter buffer and"
172 "resetting timing.", max_video_delay_ms_);
173 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
183 if (!render_timing) {
184 // Decode frame as close as possible to the render timestamp.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000185 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);
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000208 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
209 "SetRenderTS", "render_time", next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000210 if (dual_receiver != NULL) {
211 dual_receiver->UpdateState(*frame);
212 }
213 if (!frame->Complete()) {
214 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000215 bool retransmitted = false;
216 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000217 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000218 if (last_packet_time_ms >= 0 && !retransmitted) {
219 // We don't want to include timestamps which have suffered from
220 // retransmission here, since we compensate with extra retransmission
221 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000222 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000223 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000224 }
225 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000226}
227
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000228void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
229 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000230}
231
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000232void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
233 uint32_t* framerate) {
234 assert(bitrate);
235 assert(framerate);
236 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000237}
238
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000239void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
240 assert(frame_count);
241 jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames,
242 &frame_count->numKeyFrames);
niklase@google.com470e71d2011-07-07 08:21:25 +0000243}
244
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000245uint32_t VCMReceiver::DiscardedPackets() const {
246 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000247}
248
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000249void VCMReceiver::SetNackMode(VCMNackMode nackMode,
250 int low_rtt_nack_threshold_ms,
251 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000252 CriticalSectionScoped cs(crit_sect_);
253 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000254 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
255 high_rtt_nack_threshold_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000256 if (!master_) {
257 state_ = kPassive; // The dual decoder defaults to passive.
258 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000259}
260
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000261void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000262 int max_packet_age_to_nack,
263 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000264 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000265 max_packet_age_to_nack,
266 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000267}
268
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000269VCMNackMode VCMReceiver::NackMode() const {
270 CriticalSectionScoped cs(crit_sect_);
271 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000272}
273
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000274VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000275 uint16_t size,
276 uint16_t* nack_list_length) {
277 bool request_key_frame = false;
278 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
279 nack_list_length, &request_key_frame);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000280 if (*nack_list_length > size) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000281 *nack_list_length = 0;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000282 return kNackNeedMoreMemory;
283 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000284 if (internal_nack_list != NULL && *nack_list_length > 0) {
285 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000286 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000287 if (request_key_frame) {
288 return kNackKeyFrameRequest;
289 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000290 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000291}
292
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000293// Decide whether we should change decoder state. This should be done if the
294// dual decoder has caught up with the decoder decoding with packet losses.
295bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
296 VCMReceiver& dual_receiver) const {
297 if (dual_frame == NULL) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000298 return false;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000299 }
300 if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
301 dual_receiver.UpdateState(kWaitForPrimaryDecode);
302 return true;
303 }
304 return false;
niklase@google.com470e71d2011-07-07 08:21:25 +0000305}
306
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000307void VCMReceiver::CopyJitterBufferStateFromReceiver(
308 const VCMReceiver& receiver) {
309 jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000310}
311
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000312VCMReceiverState VCMReceiver::State() const {
313 CriticalSectionScoped cs(crit_sect_);
314 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000315}
316
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000317void VCMReceiver::SetDecodeWithErrors(bool enable){
318 CriticalSectionScoped cs(crit_sect_);
319 jitter_buffer_.DecodeWithErrors(enable);
320}
321
322bool VCMReceiver::DecodeWithErrors() const {
323 CriticalSectionScoped cs(crit_sect_);
324 return jitter_buffer_.decode_with_errors();
325}
326
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000327int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
328 CriticalSectionScoped cs(crit_sect_);
329 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
330 return -1;
331 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000332 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000333 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000334 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000335 return 0;
336}
337
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000338int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000339 uint32_t timestamp_start = 0u;
340 uint32_t timestamp_end = 0u;
341 // Render timestamps are computed just prior to decoding. Therefore this is
342 // only an estimate based on frames' timestamps and current timing state.
343 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
344 if (timestamp_start == timestamp_end) {
345 return 0;
346 }
347 // Update timing.
348 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000349 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000350 // Get render timestamps.
351 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
352 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
353 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000354}
355
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000356void VCMReceiver::UpdateState(VCMReceiverState new_state) {
357 CriticalSectionScoped cs(crit_sect_);
358 assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
359 state_ = new_state;
niklase@google.com470e71d2011-07-07 08:21:25 +0000360}
361
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000362void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
363 if (jitter_buffer_.nack_mode() == kNoNack) {
364 // Dual decoder mode has not been enabled.
365 return;
366 }
367 // Update the dual receiver state.
368 if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
369 UpdateState(kPassive);
370 }
371 if (State() == kWaitForPrimaryDecode &&
372 frame.Complete() && !frame.MissingFrame()) {
373 UpdateState(kPassive);
374 }
375 if (frame.MissingFrame() || !frame.Complete()) {
376 // State was corrupted, enable dual receiver.
377 UpdateState(kReceiving);
378 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000379}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000380} // namespace webrtc