niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1 | /* |
stefan@webrtc.org | 91c6308 | 2012-01-31 10:49:08 +0000 | [diff] [blame] | 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3 | * |
| 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.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 11 | #include "webrtc/modules/video_coding/main/source/receiver.h" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 12 | |
| 13 | #include <assert.h> |
| 14 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 15 | #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.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 19 | #include "webrtc/system_wrappers/interface/clock.h" |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 20 | #include "webrtc/system_wrappers/interface/trace.h" |
hclam@chromium.org | 806dc3b | 2013-04-09 19:54:10 +0000 | [diff] [blame] | 21 | #include "webrtc/system_wrappers/interface/trace_event.h" |
stefan@webrtc.org | 91c6308 | 2012-01-31 10:49:08 +0000 | [diff] [blame] | 22 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 23 | namespace webrtc { |
| 24 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 25 | enum { kMaxReceiverDelayMs = 10000 }; |
| 26 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 27 | VCMReceiver::VCMReceiver(VCMTiming* timing, |
stefan@webrtc.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 28 | Clock* clock, |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 29 | EventFactory* event_factory, |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 30 | int32_t vcm_id, |
| 31 | int32_t receiver_id, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 32 | bool master) |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 33 | : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), |
| 34 | vcm_id_(vcm_id), |
| 35 | clock_(clock), |
| 36 | receiver_id_(receiver_id), |
| 37 | master_(master), |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 38 | jitter_buffer_(clock_, event_factory, vcm_id, receiver_id, master), |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 39 | timing_(timing), |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 40 | render_wait_event_(event_factory->CreateEvent()), |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 41 | state_(kPassive), |
| 42 | max_video_delay_ms_(kMaxVideoDelayMs) {} |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 43 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 44 | VCMReceiver::~VCMReceiver() { |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 45 | render_wait_event_->Set(); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 46 | delete crit_sect_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 47 | } |
| 48 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 49 | void 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.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 56 | render_wait_event_->Reset(); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 57 | if (master_) { |
| 58 | state_ = kReceiving; |
| 59 | } else { |
| 60 | state_ = kPassive; |
| 61 | } |
henrik.lundin@webrtc.org | baf6db5 | 2011-11-02 18:58:39 +0000 | [diff] [blame] | 62 | } |
| 63 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 64 | int32_t VCMReceiver::Initialize() { |
| 65 | CriticalSectionScoped cs(crit_sect_); |
| 66 | Reset(); |
| 67 | if (!master_) { |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 68 | SetNackMode(kNoNack, -1, -1); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 69 | } |
| 70 | return VCM_OK; |
| 71 | } |
| 72 | |
| 73 | void VCMReceiver::UpdateRtt(uint32_t rtt) { |
| 74 | jitter_buffer_.UpdateRtt(rtt); |
| 75 | } |
| 76 | |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 77 | int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, |
| 78 | uint16_t frame_width, |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 79 | uint16_t frame_height) { |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 80 | // 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.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 86 | return VCM_OK; |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 87 | } 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.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 95 | } |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 96 | 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.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 110 | } |
| 111 | return VCM_OK; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 112 | } |
| 113 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 114 | VCMEncodedFrame* 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.org | 806dc3b | 2013-04-09 19:54:10 +0000 | [diff] [blame] | 119 | TRACE_EVENT0("webrtc", "Recv::FrameForDecoding"); |
stefan@webrtc.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 120 | const int64_t start_time_ms = clock_->TimeInMilliseconds(); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 121 | 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.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 141 | return NULL; |
| 142 | } |
mikhal@webrtc.org | d3cd565 | 2013-05-03 17:54:18 +0000 | [diff] [blame] | 143 | |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 144 | // We have a frame - Set timing and render timestamp. |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 145 | timing_->SetRequiredDelay(jitter_buffer_.EstimatedJitterMs()); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 146 | 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.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 168 | } |
| 169 | |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 170 | if (timing_error) { |
| 171 | // Timing error => reset timing and flush the jitter buffer. |
| 172 | jitter_buffer_.Flush(); |
stefan@webrtc.org | 9f557c1 | 2013-05-17 12:55:07 +0000 | [diff] [blame] | 173 | timing_->Reset(); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 174 | 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.org | 8f86cc8 | 2013-05-07 18:05:21 +0000 | [diff] [blame] | 199 | if (frame == NULL) { |
| 200 | return NULL; |
| 201 | } |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 202 | 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.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 208 | bool retransmitted = false; |
| 209 | const int64_t last_packet_time_ms = |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 210 | jitter_buffer_.LastPacketTime(frame, &retransmitted); |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 211 | 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.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 215 | timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms); |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 216 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 217 | } |
| 218 | return frame; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 219 | } |
| 220 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 221 | void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) { |
| 222 | jitter_buffer_.ReleaseFrame(frame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 223 | } |
| 224 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 225 | void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, |
| 226 | uint32_t* framerate) { |
| 227 | assert(bitrate); |
| 228 | assert(framerate); |
| 229 | jitter_buffer_.IncomingRateStatistics(framerate, bitrate); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 230 | } |
| 231 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 232 | void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const { |
| 233 | assert(frame_count); |
| 234 | jitter_buffer_.FrameStatistics(&frame_count->numDeltaFrames, |
| 235 | &frame_count->numKeyFrames); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 236 | } |
| 237 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 238 | uint32_t VCMReceiver::DiscardedPackets() const { |
| 239 | return jitter_buffer_.num_discarded_packets(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 240 | } |
| 241 | |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 242 | void VCMReceiver::SetNackMode(VCMNackMode nackMode, |
| 243 | int low_rtt_nack_threshold_ms, |
| 244 | int high_rtt_nack_threshold_ms) { |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 245 | CriticalSectionScoped cs(crit_sect_); |
| 246 | // Default to always having NACK enabled in hybrid mode. |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 247 | jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms, |
| 248 | high_rtt_nack_threshold_ms); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 249 | if (!master_) { |
| 250 | state_ = kPassive; // The dual decoder defaults to passive. |
| 251 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 252 | } |
| 253 | |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 254 | void VCMReceiver::SetNackSettings(size_t max_nack_list_size, |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 255 | int max_packet_age_to_nack, |
| 256 | int max_incomplete_time_ms) { |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 257 | jitter_buffer_.SetNackSettings(max_nack_list_size, |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 258 | max_packet_age_to_nack, |
| 259 | max_incomplete_time_ms); |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 260 | } |
| 261 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 262 | VCMNackMode VCMReceiver::NackMode() const { |
| 263 | CriticalSectionScoped cs(crit_sect_); |
| 264 | return jitter_buffer_.nack_mode(); |
stefan@webrtc.org | 791eec7 | 2011-10-11 07:53:43 +0000 | [diff] [blame] | 265 | } |
| 266 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 267 | VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list, |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 268 | 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.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 273 | if (*nack_list_length > size) { |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 274 | *nack_list_length = 0; |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 275 | return kNackNeedMoreMemory; |
| 276 | } |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 277 | if (internal_nack_list != NULL && *nack_list_length > 0) { |
| 278 | memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t)); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 279 | } |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 280 | if (request_key_frame) { |
| 281 | return kNackKeyFrameRequest; |
| 282 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 283 | return kNackOk; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 284 | } |
| 285 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 286 | // 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. |
| 288 | bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame, |
| 289 | VCMReceiver& dual_receiver) const { |
| 290 | if (dual_frame == NULL) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 291 | return false; |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 292 | } |
| 293 | if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) { |
| 294 | dual_receiver.UpdateState(kWaitForPrimaryDecode); |
| 295 | return true; |
| 296 | } |
| 297 | return false; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 298 | } |
| 299 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 300 | void VCMReceiver::CopyJitterBufferStateFromReceiver( |
| 301 | const VCMReceiver& receiver) { |
| 302 | jitter_buffer_.CopyFrom(receiver.jitter_buffer_); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 303 | } |
| 304 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 305 | VCMReceiverState VCMReceiver::State() const { |
| 306 | CriticalSectionScoped cs(crit_sect_); |
| 307 | return state_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 308 | } |
| 309 | |
mikhal@webrtc.org | dc3cd21 | 2013-04-25 20:27:04 +0000 | [diff] [blame] | 310 | void VCMReceiver::SetDecodeWithErrors(bool enable){ |
| 311 | CriticalSectionScoped cs(crit_sect_); |
| 312 | jitter_buffer_.DecodeWithErrors(enable); |
| 313 | } |
| 314 | |
| 315 | bool VCMReceiver::DecodeWithErrors() const { |
| 316 | CriticalSectionScoped cs(crit_sect_); |
| 317 | return jitter_buffer_.decode_with_errors(); |
| 318 | } |
| 319 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 320 | int 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.org | 6faba6e | 2013-04-30 15:39:34 +0000 | [diff] [blame] | 325 | jitter_buffer_.SetMaxJitterEstimate(desired_delay_ms > 0); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 326 | max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs; |
mikhal@webrtc.org | dbd6a6d | 2013-04-17 16:23:22 +0000 | [diff] [blame] | 327 | // Initializing timing to the desired delay. |
mikhal@webrtc.org | 6faba6e | 2013-04-30 15:39:34 +0000 | [diff] [blame] | 328 | timing_->SetMinimumTotalDelay(desired_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 329 | return 0; |
| 330 | } |
| 331 | |
mikhal@webrtc.org | 381da4b | 2013-04-25 21:45:29 +0000 | [diff] [blame] | 332 | int VCMReceiver::RenderBufferSizeMs() { |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 333 | 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(×tamp_start, ×tamp_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.org | 381da4b | 2013-04-25 21:45:29 +0000 | [diff] [blame] | 348 | } |
| 349 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 350 | void VCMReceiver::UpdateState(VCMReceiverState new_state) { |
| 351 | CriticalSectionScoped cs(crit_sect_); |
| 352 | assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode)); |
| 353 | state_ = new_state; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 354 | } |
| 355 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 356 | void 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.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 373 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 374 | } // namespace webrtc |