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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) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000080 // Insert the packet into the jitter buffer. The packet can either be empty or
81 // contain media at this point.
82 bool retransmitted = false;
83 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
84 &retransmitted);
85 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000086 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000087 } else if (ret == kFlushIndicator) {
88 return VCM_FLUSH_INDICATOR;
89 } else if (ret < 0) {
90 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
91 VCMId(vcm_id_, receiver_id_),
92 "Error inserting packet seqnum=%u, timestamp=%u",
93 packet.seqNum, packet.timestamp);
94 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000095 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000096 if (ret == kCompleteSession && !retransmitted) {
97 // We don't want to include timestamps which have suffered from
98 // retransmission here, since we compensate with extra retransmission
99 // delay within the jitter estimate.
100 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
101 }
102 if (master_) {
103 // Only trace the primary receiver to make it possible to parse and plot
104 // the trace file.
105 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
106 VCMId(vcm_id_, receiver_id_),
107 "Packet seqnum=%u timestamp=%u inserted at %u",
108 packet.seqNum, packet.timestamp,
109 MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000110 }
111 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000112}
113
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000114VCMEncodedFrame* VCMReceiver::FrameForDecoding(
115 uint16_t max_wait_time_ms,
116 int64_t& next_render_time_ms,
117 bool render_timing,
118 VCMReceiver* dual_receiver) {
hclam@chromium.org806dc3b2013-04-09 19:54:10 +0000119 TRACE_EVENT0("webrtc", "Recv::FrameForDecoding");
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000120 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000121 uint32_t frame_timestamp = 0;
122 // Exhaust wait time to get a complete frame for decoding.
123 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
124 max_wait_time_ms, &frame_timestamp);
125
126 if (!found_frame) {
127 // Get an incomplete frame when enabled.
128 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
129 dual_receiver->State() == kPassive &&
130 dual_receiver->NackMode() == kNack);
131 if (dual_receiver_enabled_and_passive &&
132 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
133 // Jitter buffer state might get corrupt with this frame.
134 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
135 }
136 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(
137 &frame_timestamp);
138 }
139
140 if (!found_frame) {
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000141 return NULL;
142 }
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000143
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000144 // We have a frame - Set timing and render timestamp.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000145 timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000146 const int64_t now_ms = clock_->TimeInMilliseconds();
147 timing_->UpdateCurrentDelay(frame_timestamp);
148 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
149 // Check render timing.
150 bool timing_error = false;
151 // Assume that render timing errors are due to changes in the video stream.
152 if (next_render_time_ms < 0) {
153 timing_error = true;
154 } else if (next_render_time_ms < now_ms - max_video_delay_ms_) {
155 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
156 VCMId(vcm_id_, receiver_id_),
157 "This frame should have been rendered more than %u ms ago."
158 "Flushing jitter buffer and resetting timing.",
159 max_video_delay_ms_);
160 timing_error = true;
161 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
162 max_video_delay_ms_) {
163 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
164 VCMId(vcm_id_, receiver_id_),
165 "More than %u ms target delay. Flushing jitter buffer and"
166 "resetting timing.", max_video_delay_ms_);
167 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000168 }
169
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000170 if (timing_error) {
171 // Timing error => reset timing and flush the jitter buffer.
172 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000173 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000174 return NULL;
175 }
176
177 if (!render_timing) {
178 // Decode frame as close as possible to the render timestamp.
179 TRACE_EVENT0("webrtc", "FrameForRendering");
180 const int32_t available_wait_time = max_wait_time_ms -
181 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
182 uint16_t new_max_wait_time = static_cast<uint16_t>(
183 VCM_MAX(available_wait_time, 0));
184 uint32_t wait_time_ms = timing_->MaxWaitingTime(
185 next_render_time_ms, clock_->TimeInMilliseconds());
186 if (new_max_wait_time < wait_time_ms) {
187 // We're not allowed to wait until the frame is supposed to be rendered,
188 // waiting as long as we're allowed to avoid busy looping, and then return
189 // NULL. Next call to this function might return the frame.
190 render_wait_event_->Wait(max_wait_time_ms);
191 return NULL;
192 }
193 // Wait until it's time to render.
194 render_wait_event_->Wait(wait_time_ms);
195 }
196
197 // Extract the frame from the jitter buffer and set the render time.
198 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000199 if (frame == NULL) {
200 return NULL;
201 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000202 frame->SetRenderTime(next_render_time_ms);
203 if (dual_receiver != NULL) {
204 dual_receiver->UpdateState(*frame);
205 }
206 if (!frame->Complete()) {
207 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000208 bool retransmitted = false;
209 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000210 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000211 if (last_packet_time_ms >= 0 && !retransmitted) {
212 // We don't want to include timestamps which have suffered from
213 // retransmission here, since we compensate with extra retransmission
214 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000215 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000216 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000217 }
218 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000219}
220
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000221void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
222 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000223}
224
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000225void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
226 uint32_t* framerate) {
227 assert(bitrate);
228 assert(framerate);
229 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000230}
231
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000232void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
233 assert(frame_count);
234 jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames,
235 &frame_count->numKeyFrames);
niklase@google.com470e71d2011-07-07 08:21:25 +0000236}
237
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000238uint32_t VCMReceiver::DiscardedPackets() const {
239 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000240}
241
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000242void VCMReceiver::SetNackMode(VCMNackMode nackMode,
243 int low_rtt_nack_threshold_ms,
244 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000245 CriticalSectionScoped cs(crit_sect_);
246 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000247 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
248 high_rtt_nack_threshold_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000249 if (!master_) {
250 state_ = kPassive; // The dual decoder defaults to passive.
251 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000252}
253
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000254void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000255 int max_packet_age_to_nack,
256 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000257 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000258 max_packet_age_to_nack,
259 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000260}
261
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000262VCMNackMode VCMReceiver::NackMode() const {
263 CriticalSectionScoped cs(crit_sect_);
264 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000265}
266
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000267VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000268 uint16_t size,
269 uint16_t* nack_list_length) {
270 bool request_key_frame = false;
271 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
272 nack_list_length, &request_key_frame);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000273 if (*nack_list_length > size) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000274 *nack_list_length = 0;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000275 return kNackNeedMoreMemory;
276 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000277 if (internal_nack_list != NULL && *nack_list_length > 0) {
278 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000279 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000280 if (request_key_frame) {
281 return kNackKeyFrameRequest;
282 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000283 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000284}
285
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000286// Decide whether we should change decoder state. This should be done if the
287// dual decoder has caught up with the decoder decoding with packet losses.
288bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
289 VCMReceiver& dual_receiver) const {
290 if (dual_frame == NULL) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000291 return false;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000292 }
293 if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
294 dual_receiver.UpdateState(kWaitForPrimaryDecode);
295 return true;
296 }
297 return false;
niklase@google.com470e71d2011-07-07 08:21:25 +0000298}
299
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000300void VCMReceiver::CopyJitterBufferStateFromReceiver(
301 const VCMReceiver& receiver) {
302 jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000303}
304
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000305VCMReceiverState VCMReceiver::State() const {
306 CriticalSectionScoped cs(crit_sect_);
307 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000308}
309
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000310void VCMReceiver::SetDecodeWithErrors(bool enable){
311 CriticalSectionScoped cs(crit_sect_);
312 jitter_buffer_.DecodeWithErrors(enable);
313}
314
315bool VCMReceiver::DecodeWithErrors() const {
316 CriticalSectionScoped cs(crit_sect_);
317 return jitter_buffer_.decode_with_errors();
318}
319
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000320int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
321 CriticalSectionScoped cs(crit_sect_);
322 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
323 return -1;
324 }
mikhal@webrtc.org6faba6e2013-04-30 15:39:34 +0000325 jitter_buffer_.SetMaxJitterEstimate(desired_delay_ms > 0);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000326 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000327 // Initializing timing to the desired delay.
mikhal@webrtc.org6faba6e2013-04-30 15:39:34 +0000328 timing_->SetMinimumTotalDelay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000329 return 0;
330}
331
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000332int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000333 uint32_t timestamp_start = 0u;
334 uint32_t timestamp_end = 0u;
335 // Render timestamps are computed just prior to decoding. Therefore this is
336 // only an estimate based on frames' timestamps and current timing state.
337 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
338 if (timestamp_start == timestamp_end) {
339 return 0;
340 }
341 // Update timing.
342 const int64_t now_ms = clock_->TimeInMilliseconds();
343 timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs());
344 // Get render timestamps.
345 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
346 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
347 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000348}
349
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000350void VCMReceiver::UpdateState(VCMReceiverState new_state) {
351 CriticalSectionScoped cs(crit_sect_);
352 assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
353 state_ = new_state;
niklase@google.com470e71d2011-07-07 08:21:25 +0000354}
355
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000356void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
357 if (jitter_buffer_.nack_mode() == kNoNack) {
358 // Dual decoder mode has not been enabled.
359 return;
360 }
361 // Update the dual receiver state.
362 if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
363 UpdateState(kPassive);
364 }
365 if (State() == kWaitForPrimaryDecode &&
366 frame.Complete() && !frame.MissingFrame()) {
367 UpdateState(kPassive);
368 }
369 if (frame.MissingFrame() || !frame.Complete()) {
370 // State was corrupted, enable dual receiver.
371 UpdateState(kReceiving);
372 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000373}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000374} // namespace webrtc