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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) {
80 // Find an empty frame.
81 VCMEncodedFrame* buffer = NULL;
82 const int32_t error = jitter_buffer_.GetFrame(packet, buffer);
83 if (error == VCM_OLD_PACKET_ERROR) {
niklase@google.com470e71d2011-07-07 08:21:25 +000084 return VCM_OK;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000085 } else if (error != VCM_OK) {
86 return error;
87 }
88 assert(buffer);
89 {
90 CriticalSectionScoped cs(crit_sect_);
91
92 if (frame_width && frame_height) {
93 buffer->SetEncodedSize(static_cast<uint32_t>(frame_width),
94 static_cast<uint32_t>(frame_height));
95 }
96
97 if (master_) {
98 // Only trace the primary receiver to make it possible to parse and plot
99 // the trace file.
100 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
101 VCMId(vcm_id_, receiver_id_),
102 "Packet seq_no %u of frame %u at %u",
103 packet.seqNum, packet.timestamp,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000104 MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000105 }
106
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000107 const int64_t now_ms = clock_->TimeInMilliseconds();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000108
109 int64_t render_time_ms = timing_->RenderTimeMs(packet.timestamp, now_ms);
110
111 if (render_time_ms < 0) {
112 // Render time error. Assume that this is due to some change in the
113 // incoming video stream and reset the JB and the timing.
114 jitter_buffer_.Flush();
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000115 timing_->Reset(clock_->TimeInMilliseconds());
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000116 return VCM_FLUSH_INDICATOR;
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000117 } else if (render_time_ms < now_ms - max_video_delay_ms_) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000118 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
119 VCMId(vcm_id_, receiver_id_),
120 "This frame should have been rendered more than %u ms ago."
121 "Flushing jitter buffer and resetting timing.",
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000122 max_video_delay_ms_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000123 jitter_buffer_.Flush();
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000124 timing_->Reset(clock_->TimeInMilliseconds());
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000125 return VCM_FLUSH_INDICATOR;
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000126 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
127 max_video_delay_ms_) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000128 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
129 VCMId(vcm_id_, receiver_id_),
130 "More than %u ms target delay. Flushing jitter buffer and"
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000131 "resetting timing.", max_video_delay_ms_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000132 jitter_buffer_.Flush();
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000133 timing_->Reset(clock_->TimeInMilliseconds());
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000134 return VCM_FLUSH_INDICATOR;
135 }
136
137 // First packet received belonging to this frame.
138 if (buffer->Length() == 0) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000139 const int64_t now_ms = clock_->TimeInMilliseconds();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000140 if (master_) {
141 // Only trace the primary receiver to make it possible to parse and plot
142 // the trace file.
143 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
144 VCMId(vcm_id_, receiver_id_),
145 "First packet of frame %u at %u", packet.timestamp,
146 MaskWord64ToUWord32(now_ms));
147 }
148 render_time_ms = timing_->RenderTimeMs(packet.timestamp, now_ms);
149 if (render_time_ms >= 0) {
150 buffer->SetRenderTime(render_time_ms);
151 } else {
152 buffer->SetRenderTime(now_ms);
153 }
154 }
155
156 // Insert packet into the jitter buffer both media and empty packets.
157 const VCMFrameBufferEnum
158 ret = jitter_buffer_.InsertPacket(buffer, packet);
159 if (ret == kFlushIndicator) {
160 return VCM_FLUSH_INDICATOR;
161 } else if (ret < 0) {
162 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
163 VCMId(vcm_id_, receiver_id_),
164 "Error inserting packet seq_no=%u, time_stamp=%u",
165 packet.seqNum, packet.timestamp);
166 return VCM_JITTER_BUFFER_ERROR;
167 }
168 }
169 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000170}
171
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000172VCMEncodedFrame* VCMReceiver::FrameForDecoding(
173 uint16_t max_wait_time_ms,
174 int64_t& next_render_time_ms,
175 bool render_timing,
176 VCMReceiver* dual_receiver) {
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000177 TRACE_EVENT0("webrtc", "Recv::FrameForDecoding");
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000178 // No need to enter the critical section here since the jitter buffer
179 // is thread-safe.
180 FrameType incoming_frame_type = kVideoFrameDelta;
181 next_render_time_ms = -1;
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000182 const int64_t start_time_ms = clock_->TimeInMilliseconds();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000183 int64_t ret = jitter_buffer_.NextTimestamp(max_wait_time_ms,
184 &incoming_frame_type,
185 &next_render_time_ms);
186 if (ret < 0) {
187 // No timestamp in jitter buffer at the moment.
188 return NULL;
189 }
190 const uint32_t time_stamp = static_cast<uint32_t>(ret);
191
192 // Update the timing.
193 timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
194 timing_->UpdateCurrentDelay(time_stamp);
195
196 const int32_t temp_wait_time = max_wait_time_ms -
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000197 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000198 uint16_t new_max_wait_time = static_cast<uint16_t>(VCM_MAX(temp_wait_time,
199 0));
200
201 VCMEncodedFrame* frame = NULL;
202
203 if (render_timing) {
204 frame = FrameForDecoding(new_max_wait_time, next_render_time_ms,
205 dual_receiver);
206 } else {
207 frame = FrameForRendering(new_max_wait_time, next_render_time_ms,
208 dual_receiver);
209 }
210
211 if (frame != NULL) {
212 bool retransmitted = false;
213 const int64_t last_packet_time_ms =
214 jitter_buffer_.LastPacketTime(frame, &retransmitted);
215 if (last_packet_time_ms >= 0 && !retransmitted) {
216 // We don't want to include timestamps which have suffered from
217 // retransmission here, since we compensate with extra retransmission
218 // delay within the jitter estimate.
219 timing_->IncomingTimestamp(time_stamp, last_packet_time_ms);
220 }
221 if (dual_receiver != NULL) {
222 dual_receiver->UpdateState(*frame);
223 }
224 }
225 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000226}
227
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000228VCMEncodedFrame* VCMReceiver::FrameForDecoding(
229 uint16_t max_wait_time_ms,
230 int64_t next_render_time_ms,
231 VCMReceiver* dual_receiver) {
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000232 TRACE_EVENT1("webrtc", "FrameForDecoding",
233 "max_wait", max_wait_time_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000234 // How long can we wait until we must decode the next frame.
235 uint32_t wait_time_ms = timing_->MaxWaitingTime(
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000236 next_render_time_ms, clock_->TimeInMilliseconds());
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000237
238 // Try to get a complete frame from the jitter buffer.
239 VCMEncodedFrame* frame = jitter_buffer_.GetCompleteFrameForDecoding(0);
240
241 if (frame == NULL && max_wait_time_ms == 0 && wait_time_ms > 0) {
242 // If we're not allowed to wait for frames to get complete we must
243 // calculate if it's time to decode, and if it's not we will just return
244 // for now.
245 return NULL;
246 }
247
248 if (frame == NULL && VCM_MIN(wait_time_ms, max_wait_time_ms) == 0) {
249 // No time to wait for a complete frame, check if we have an incomplete.
250 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
251 dual_receiver->State() == kPassive &&
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000252 dual_receiver->NackMode() == kNack);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000253 if (dual_receiver_enabled_and_passive &&
254 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
255 // Jitter buffer state might get corrupt with this frame.
256 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
257 frame = jitter_buffer_.GetFrameForDecoding();
258 assert(frame);
259 } else {
260 frame = jitter_buffer_.GetFrameForDecoding();
niklase@google.com470e71d2011-07-07 08:21:25 +0000261 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000262 }
263 if (frame == NULL) {
264 // Wait for a complete frame.
265 frame = jitter_buffer_.GetCompleteFrameForDecoding(max_wait_time_ms);
266 }
267 if (frame == NULL) {
268 // Get an incomplete frame.
269 if (timing_->MaxWaitingTime(next_render_time_ms,
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000270 clock_->TimeInMilliseconds()) > 0) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000271 // Still time to wait for a complete frame.
272 return NULL;
niklase@google.com470e71d2011-07-07 08:21:25 +0000273 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000274
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000275 // No time left to wait, we must decode this frame now.
276 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
277 dual_receiver->State() == kPassive &&
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000278 dual_receiver->NackMode() == kNack);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000279 if (dual_receiver_enabled_and_passive &&
280 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
281 // Jitter buffer state might get corrupt with this frame.
282 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
niklase@google.com470e71d2011-07-07 08:21:25 +0000283 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000284
285 frame = jitter_buffer_.GetFrameForDecoding();
286 }
287 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000288}
289
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000290VCMEncodedFrame* VCMReceiver::FrameForRendering(uint16_t max_wait_time_ms,
291 int64_t next_render_time_ms,
292 VCMReceiver* dual_receiver) {
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000293 TRACE_EVENT0("webrtc", "FrameForRendering");
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000294 // How long MUST we wait until we must decode the next frame. This is
295 // different for the case where we have a renderer which can render at a
296 // specified time. Here we must wait as long as possible before giving the
297 // frame to the decoder, which will render the frame as soon as it has been
298 // decoded.
299 uint32_t wait_time_ms = timing_->MaxWaitingTime(
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000300 next_render_time_ms, clock_->TimeInMilliseconds());
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000301 if (max_wait_time_ms < wait_time_ms) {
302 // If we're not allowed to wait until the frame is supposed to be rendered
303 // we will have to return NULL for now.
304 return NULL;
305 }
306 // Wait until it's time to render.
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +0000307 render_wait_event_->Wait(wait_time_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000308
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000309 // Get a complete frame if possible.
310 VCMEncodedFrame* frame = jitter_buffer_.GetCompleteFrameForDecoding(0);
niklase@google.com470e71d2011-07-07 08:21:25 +0000311
mikhal@webrtc.orgc2a3aa72013-04-12 19:53:30 +0000312 if (frame == NULL) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000313 // Get an incomplete frame.
314 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
315 dual_receiver->State() == kPassive &&
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000316 dual_receiver->NackMode() == kNack);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000317 if (dual_receiver_enabled_and_passive &&
318 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
319 // Jitter buffer state might get corrupt with this frame.
320 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
niklase@google.com470e71d2011-07-07 08:21:25 +0000321 }
322
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000323 frame = jitter_buffer_.GetFrameForDecoding();
324 }
325 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000326}
327
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000328void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
329 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000330}
331
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000332void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
333 uint32_t* framerate) {
334 assert(bitrate);
335 assert(framerate);
336 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
337 *bitrate /= 1000; // Should be in kbps.
niklase@google.com470e71d2011-07-07 08:21:25 +0000338}
339
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000340void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
341 assert(frame_count);
342 jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames,
343 &frame_count->numKeyFrames);
niklase@google.com470e71d2011-07-07 08:21:25 +0000344}
345
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000346uint32_t VCMReceiver::DiscardedPackets() const {
347 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000348}
349
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000350void VCMReceiver::SetNackMode(VCMNackMode nackMode,
351 int low_rtt_nack_threshold_ms,
352 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000353 CriticalSectionScoped cs(crit_sect_);
354 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000355 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
356 high_rtt_nack_threshold_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000357 if (!master_) {
358 state_ = kPassive; // The dual decoder defaults to passive.
359 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000360}
361
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000362void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
363 int max_packet_age_to_nack) {
364 jitter_buffer_.SetNackSettings(max_nack_list_size,
365 max_packet_age_to_nack);
366}
367
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000368VCMNackMode VCMReceiver::NackMode() const {
369 CriticalSectionScoped cs(crit_sect_);
370 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000371}
372
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000373VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000374 uint16_t size,
375 uint16_t* nack_list_length) {
376 bool request_key_frame = false;
377 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
378 nack_list_length, &request_key_frame);
379 if (request_key_frame) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000380 // This combination is used to trigger key frame requests.
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000381 return kNackKeyFrameRequest;
382 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000383 if (*nack_list_length > size) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000384 return kNackNeedMoreMemory;
385 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000386 if (internal_nack_list != NULL && *nack_list_length > 0) {
387 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000388 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000389 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000390}
391
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000392// Decide whether we should change decoder state. This should be done if the
393// dual decoder has caught up with the decoder decoding with packet losses.
394bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
395 VCMReceiver& dual_receiver) const {
396 if (dual_frame == NULL) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000397 return false;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000398 }
399 if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
400 dual_receiver.UpdateState(kWaitForPrimaryDecode);
401 return true;
402 }
403 return false;
niklase@google.com470e71d2011-07-07 08:21:25 +0000404}
405
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000406void VCMReceiver::CopyJitterBufferStateFromReceiver(
407 const VCMReceiver& receiver) {
408 jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000409}
410
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000411VCMReceiverState VCMReceiver::State() const {
412 CriticalSectionScoped cs(crit_sect_);
413 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000414}
415
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000416int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
417 CriticalSectionScoped cs(crit_sect_);
418 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
419 return -1;
420 }
421 jitter_buffer_.SetMaxJitterEstimate(desired_delay_ms);
422 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
423 timing_->SetMaxVideoDelay(max_video_delay_ms_);
424 return 0;
425}
426
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000427void VCMReceiver::UpdateState(VCMReceiverState new_state) {
428 CriticalSectionScoped cs(crit_sect_);
429 assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
430 state_ = new_state;
niklase@google.com470e71d2011-07-07 08:21:25 +0000431}
432
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000433void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
434 if (jitter_buffer_.nack_mode() == kNoNack) {
435 // Dual decoder mode has not been enabled.
436 return;
437 }
438 // Update the dual receiver state.
439 if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
440 UpdateState(kPassive);
441 }
442 if (State() == kWaitForPrimaryDecode &&
443 frame.Complete() && !frame.MissingFrame()) {
444 UpdateState(kPassive);
445 }
446 if (frame.MissingFrame() || !frame.Complete()) {
447 // State was corrupted, enable dual receiver.
448 UpdateState(kReceiving);
449 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000450}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000451} // namespace webrtc