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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,
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000030 EventFactory* event_factory,
niklase@google.com470e71d2011-07-07 08:21:25 +000031 bool master)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000032 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000033 clock_(clock),
stefan@webrtc.org34c5da62014-04-11 14:08:35 +000034 jitter_buffer_(clock_, event_factory),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000035 timing_(timing),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000036 render_wait_event_(event_factory->CreateEvent()),
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000037 state_(kPassive),
38 max_video_delay_ms_(kMaxVideoDelayMs) {}
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 }
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000052 render_wait_event_->Reset();
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000053 state_ = kReceiving;
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000054}
55
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000056int32_t VCMReceiver::Initialize() {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000057 Reset();
stefan@webrtc.org4f3624d2013-09-20 07:43:17 +000058 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000059 return VCM_OK;
60}
61
62void VCMReceiver::UpdateRtt(uint32_t rtt) {
63 jitter_buffer_.UpdateRtt(rtt);
64}
65
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000066int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
67 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000068 uint16_t frame_height) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000069 // Insert the packet into the jitter buffer. The packet can either be empty or
70 // contain media at this point.
71 bool retransmitted = false;
72 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
73 &retransmitted);
74 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000075 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000076 } else if (ret == kFlushIndicator) {
77 return VCM_FLUSH_INDICATOR;
78 } else if (ret < 0) {
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000079 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000080 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000081 if (ret == kCompleteSession && !retransmitted) {
82 // We don't want to include timestamps which have suffered from
83 // retransmission here, since we compensate with extra retransmission
84 // delay within the jitter estimate.
85 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
86 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000087 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +000088}
89
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000090VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
91 int64_t& next_render_time_ms,
92 bool render_timing) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +000093 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +000094 uint32_t frame_timestamp = 0;
95 // Exhaust wait time to get a complete frame for decoding.
96 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
97 max_wait_time_ms, &frame_timestamp);
98
pbos@webrtc.org4f16c872014-11-24 09:06:48 +000099 if (!found_frame)
100 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000101
pbos@webrtc.org4f16c872014-11-24 09:06:48 +0000102 if (!found_frame)
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000103 return NULL;
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000104
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000105 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000106 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000107 const int64_t now_ms = clock_->TimeInMilliseconds();
108 timing_->UpdateCurrentDelay(frame_timestamp);
109 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
110 // Check render timing.
111 bool timing_error = false;
112 // Assume that render timing errors are due to changes in the video stream.
113 if (next_render_time_ms < 0) {
114 timing_error = true;
pbos@webrtc.orgb5f30292014-03-13 08:53:39 +0000115 } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000116 int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
117 LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
118 << "delay bounds (" << frame_delay << " > "
119 << max_video_delay_ms_
120 << "). Resetting the video jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000121 timing_error = true;
122 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
123 max_video_delay_ms_) {
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000124 LOG(LS_WARNING) << "The video target delay has grown larger than "
125 << max_video_delay_ms_ << " ms. Resetting jitter buffer.";
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000126 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000127 }
128
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000129 if (timing_error) {
130 // Timing error => reset timing and flush the jitter buffer.
131 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000132 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000133 return NULL;
134 }
135
136 if (!render_timing) {
137 // Decode frame as close as possible to the render timestamp.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000138 const int32_t available_wait_time = max_wait_time_ms -
139 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
140 uint16_t new_max_wait_time = static_cast<uint16_t>(
141 VCM_MAX(available_wait_time, 0));
142 uint32_t wait_time_ms = timing_->MaxWaitingTime(
143 next_render_time_ms, clock_->TimeInMilliseconds());
144 if (new_max_wait_time < wait_time_ms) {
145 // We're not allowed to wait until the frame is supposed to be rendered,
146 // waiting as long as we're allowed to avoid busy looping, and then return
147 // NULL. Next call to this function might return the frame.
148 render_wait_event_->Wait(max_wait_time_ms);
149 return NULL;
150 }
151 // Wait until it's time to render.
152 render_wait_event_->Wait(wait_time_ms);
153 }
154
155 // Extract the frame from the jitter buffer and set the render time.
156 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000157 if (frame == NULL) {
158 return NULL;
159 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000160 frame->SetRenderTime(next_render_time_ms);
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000161 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
162 "SetRenderTS", "render_time", next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000163 if (!frame->Complete()) {
164 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000165 bool retransmitted = false;
166 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000167 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000168 if (last_packet_time_ms >= 0 && !retransmitted) {
169 // We don't want to include timestamps which have suffered from
170 // retransmission here, since we compensate with extra retransmission
171 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000172 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000173 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000174 }
175 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000176}
177
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000178void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
179 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000180}
181
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000182void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
183 uint32_t* framerate) {
184 assert(bitrate);
185 assert(framerate);
186 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000187}
188
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000189void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
190 assert(frame_count);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000191 std::map<FrameType, uint32_t> counts(jitter_buffer_.FrameStatistics());
192 frame_count->numDeltaFrames = counts[kVideoFrameDelta];
193 frame_count->numKeyFrames = counts[kVideoFrameKey];
niklase@google.com470e71d2011-07-07 08:21:25 +0000194}
195
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000196uint32_t VCMReceiver::DiscardedPackets() const {
197 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000198}
199
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000200void VCMReceiver::SetNackMode(VCMNackMode nackMode,
201 int low_rtt_nack_threshold_ms,
202 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000203 CriticalSectionScoped cs(crit_sect_);
204 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000205 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
206 high_rtt_nack_threshold_ms);
niklase@google.com470e71d2011-07-07 08:21:25 +0000207}
208
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000209void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000210 int max_packet_age_to_nack,
211 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000212 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000213 max_packet_age_to_nack,
214 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000215}
216
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000217VCMNackMode VCMReceiver::NackMode() const {
218 CriticalSectionScoped cs(crit_sect_);
219 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000220}
221
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000222VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000223 uint16_t size,
224 uint16_t* nack_list_length) {
225 bool request_key_frame = false;
226 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
227 nack_list_length, &request_key_frame);
stefan@webrtc.org34c5da62014-04-11 14:08:35 +0000228 assert(*nack_list_length <= size);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000229 if (internal_nack_list != NULL && *nack_list_length > 0) {
230 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000231 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000232 if (request_key_frame) {
233 return kNackKeyFrameRequest;
234 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000235 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000236}
237
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000238VCMReceiverState VCMReceiver::State() const {
239 CriticalSectionScoped cs(crit_sect_);
240 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000241}
242
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000243void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
244 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000245}
246
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000247VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000248 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000249}
250
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000251int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
252 CriticalSectionScoped cs(crit_sect_);
253 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
254 return -1;
255 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000256 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000257 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000258 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000259 return 0;
260}
261
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000262int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000263 uint32_t timestamp_start = 0u;
264 uint32_t timestamp_end = 0u;
265 // Render timestamps are computed just prior to decoding. Therefore this is
266 // only an estimate based on frames' timestamps and current timing state.
267 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
268 if (timestamp_start == timestamp_end) {
269 return 0;
270 }
271 // Update timing.
272 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000273 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000274 // Get render timestamps.
275 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
276 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
277 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000278}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000279} // namespace webrtc