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 | |
pbos@webrtc.org | 3f655aa | 2014-03-18 11:10:11 +0000 | [diff] [blame] | 15 | #include <cstdlib> |
| 16 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 17 | #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.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 20 | #include "webrtc/system_wrappers/interface/clock.h" |
stefan@webrtc.org | 34c5da6 | 2014-04-11 14:08:35 +0000 | [diff] [blame] | 21 | #include "webrtc/system_wrappers/interface/logging.h" |
hclam@chromium.org | 806dc3b | 2013-04-09 19:54:10 +0000 | [diff] [blame] | 22 | #include "webrtc/system_wrappers/interface/trace_event.h" |
stefan@webrtc.org | 91c6308 | 2012-01-31 10:49:08 +0000 | [diff] [blame] | 23 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 24 | namespace webrtc { |
| 25 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 26 | enum { kMaxReceiverDelayMs = 10000 }; |
| 27 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 28 | VCMReceiver::VCMReceiver(VCMTiming* timing, |
stefan@webrtc.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 29 | Clock* clock, |
Wan-Teh Chang | 92d9489 | 2015-05-28 13:36:06 -0700 | [diff] [blame] | 30 | EventFactory* event_factory) |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 31 | : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 32 | clock_(clock), |
stefan@webrtc.org | 34c5da6 | 2014-04-11 14:08:35 +0000 | [diff] [blame] | 33 | jitter_buffer_(clock_, event_factory), |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 34 | timing_(timing), |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 35 | render_wait_event_(event_factory->CreateEvent()), |
Peter Boström | 5464a6e | 2015-04-21 16:35:50 +0200 | [diff] [blame] | 36 | max_video_delay_ms_(kMaxVideoDelayMs) { |
| 37 | Reset(); |
| 38 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 39 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 40 | VCMReceiver::~VCMReceiver() { |
stefan@webrtc.org | 2baf5f5 | 2013-03-13 08:46:25 +0000 | [diff] [blame] | 41 | render_wait_event_->Set(); |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 42 | delete crit_sect_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 43 | } |
| 44 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 45 | void VCMReceiver::Reset() { |
| 46 | CriticalSectionScoped cs(crit_sect_); |
| 47 | if (!jitter_buffer_.Running()) { |
| 48 | jitter_buffer_.Start(); |
| 49 | } else { |
| 50 | jitter_buffer_.Flush(); |
| 51 | } |
henrik.lundin@webrtc.org | baf6db5 | 2011-11-02 18:58:39 +0000 | [diff] [blame] | 52 | } |
| 53 | |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 54 | void VCMReceiver::UpdateRtt(int64_t rtt) { |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 55 | jitter_buffer_.UpdateRtt(rtt); |
| 56 | } |
| 57 | |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 58 | int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, |
| 59 | uint16_t frame_width, |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 60 | uint16_t frame_height) { |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 61 | // Insert the packet into the jitter buffer. The packet can either be empty or |
| 62 | // contain media at this point. |
| 63 | bool retransmitted = false; |
| 64 | const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet, |
| 65 | &retransmitted); |
| 66 | if (ret == kOldPacket) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 67 | return VCM_OK; |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 68 | } else if (ret == kFlushIndicator) { |
| 69 | return VCM_FLUSH_INDICATOR; |
| 70 | } else if (ret < 0) { |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 71 | return VCM_JITTER_BUFFER_ERROR; |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 72 | } |
stefan@webrtc.org | 3417eb4 | 2013-05-21 15:25:53 +0000 | [diff] [blame] | 73 | if (ret == kCompleteSession && !retransmitted) { |
| 74 | // We don't want to include timestamps which have suffered from |
| 75 | // retransmission here, since we compensate with extra retransmission |
| 76 | // delay within the jitter estimate. |
| 77 | timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds()); |
| 78 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 79 | return VCM_OK; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 80 | } |
| 81 | |
pbos@webrtc.org | 4dd40d6 | 2015-02-17 13:22:43 +0000 | [diff] [blame] | 82 | void VCMReceiver::TriggerDecoderShutdown() { |
| 83 | jitter_buffer_.Stop(); |
| 84 | render_wait_event_->Set(); |
| 85 | } |
| 86 | |
pbos@webrtc.org | 4f16c87 | 2014-11-24 09:06:48 +0000 | [diff] [blame] | 87 | VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms, |
| 88 | int64_t& next_render_time_ms, |
| 89 | bool render_timing) { |
stefan@webrtc.org | a678a3b | 2013-01-21 07:42:11 +0000 | [diff] [blame] | 90 | const int64_t start_time_ms = clock_->TimeInMilliseconds(); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 91 | uint32_t frame_timestamp = 0; |
| 92 | // Exhaust wait time to get a complete frame for decoding. |
| 93 | bool found_frame = jitter_buffer_.NextCompleteTimestamp( |
| 94 | max_wait_time_ms, &frame_timestamp); |
| 95 | |
pbos@webrtc.org | 4f16c87 | 2014-11-24 09:06:48 +0000 | [diff] [blame] | 96 | if (!found_frame) |
| 97 | found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 98 | |
pbos@webrtc.org | 4f16c87 | 2014-11-24 09:06:48 +0000 | [diff] [blame] | 99 | if (!found_frame) |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 100 | return NULL; |
mikhal@webrtc.org | d3cd565 | 2013-05-03 17:54:18 +0000 | [diff] [blame] | 101 | |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 102 | // We have a frame - Set timing and render timestamp. |
mikhal@webrtc.org | adc64a7 | 2013-05-30 16:20:18 +0000 | [diff] [blame] | 103 | timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs()); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 104 | const int64_t now_ms = clock_->TimeInMilliseconds(); |
| 105 | timing_->UpdateCurrentDelay(frame_timestamp); |
| 106 | next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms); |
| 107 | // Check render timing. |
| 108 | bool timing_error = false; |
| 109 | // Assume that render timing errors are due to changes in the video stream. |
| 110 | if (next_render_time_ms < 0) { |
| 111 | timing_error = true; |
pbos@webrtc.org | b5f3029 | 2014-03-13 08:53:39 +0000 | [diff] [blame] | 112 | } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) { |
stefan@webrtc.org | 34c5da6 | 2014-04-11 14:08:35 +0000 | [diff] [blame] | 113 | int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms)); |
| 114 | LOG(LS_WARNING) << "A frame about to be decoded is out of the configured " |
| 115 | << "delay bounds (" << frame_delay << " > " |
| 116 | << max_video_delay_ms_ |
| 117 | << "). Resetting the video jitter buffer."; |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 118 | timing_error = true; |
| 119 | } else if (static_cast<int>(timing_->TargetVideoDelay()) > |
| 120 | max_video_delay_ms_) { |
stefan@webrtc.org | 34c5da6 | 2014-04-11 14:08:35 +0000 | [diff] [blame] | 121 | LOG(LS_WARNING) << "The video target delay has grown larger than " |
| 122 | << max_video_delay_ms_ << " ms. Resetting jitter buffer."; |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 123 | timing_error = true; |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 124 | } |
| 125 | |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 126 | if (timing_error) { |
| 127 | // Timing error => reset timing and flush the jitter buffer. |
| 128 | jitter_buffer_.Flush(); |
stefan@webrtc.org | 9f557c1 | 2013-05-17 12:55:07 +0000 | [diff] [blame] | 129 | timing_->Reset(); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 130 | return NULL; |
| 131 | } |
| 132 | |
| 133 | if (!render_timing) { |
| 134 | // Decode frame as close as possible to the render timestamp. |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 135 | const int32_t available_wait_time = max_wait_time_ms - |
| 136 | static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms); |
| 137 | uint16_t new_max_wait_time = static_cast<uint16_t>( |
| 138 | VCM_MAX(available_wait_time, 0)); |
| 139 | uint32_t wait_time_ms = timing_->MaxWaitingTime( |
| 140 | next_render_time_ms, clock_->TimeInMilliseconds()); |
| 141 | if (new_max_wait_time < wait_time_ms) { |
| 142 | // We're not allowed to wait until the frame is supposed to be rendered, |
| 143 | // waiting as long as we're allowed to avoid busy looping, and then return |
| 144 | // NULL. Next call to this function might return the frame. |
Niklas Enbom | b4c5eaa | 2015-06-03 09:34:25 -0700 | [diff] [blame] | 145 | render_wait_event_->Wait(new_max_wait_time); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 146 | return NULL; |
| 147 | } |
| 148 | // Wait until it's time to render. |
| 149 | render_wait_event_->Wait(wait_time_ms); |
| 150 | } |
| 151 | |
| 152 | // Extract the frame from the jitter buffer and set the render time. |
| 153 | VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp); |
mikhal@webrtc.org | 8f86cc8 | 2013-05-07 18:05:21 +0000 | [diff] [blame] | 154 | if (frame == NULL) { |
| 155 | return NULL; |
| 156 | } |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 157 | frame->SetRenderTime(next_render_time_ms); |
hclam@chromium.org | 1a7b9b9 | 2013-07-08 21:31:18 +0000 | [diff] [blame] | 158 | TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), |
| 159 | "SetRenderTS", "render_time", next_render_time_ms); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 160 | if (!frame->Complete()) { |
| 161 | // Update stats for incomplete frames. |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 162 | bool retransmitted = false; |
| 163 | const int64_t last_packet_time_ms = |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 164 | jitter_buffer_.LastPacketTime(frame, &retransmitted); |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 165 | if (last_packet_time_ms >= 0 && !retransmitted) { |
| 166 | // We don't want to include timestamps which have suffered from |
| 167 | // retransmission here, since we compensate with extra retransmission |
| 168 | // delay within the jitter estimate. |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 169 | timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms); |
stefan@webrtc.org | 4ce19b1 | 2013-05-06 13:16:51 +0000 | [diff] [blame] | 170 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 171 | } |
| 172 | return frame; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 173 | } |
| 174 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 175 | void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) { |
| 176 | jitter_buffer_.ReleaseFrame(frame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 177 | } |
| 178 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 179 | void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, |
| 180 | uint32_t* framerate) { |
| 181 | assert(bitrate); |
| 182 | assert(framerate); |
| 183 | jitter_buffer_.IncomingRateStatistics(framerate, bitrate); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 184 | } |
| 185 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 186 | uint32_t VCMReceiver::DiscardedPackets() const { |
| 187 | return jitter_buffer_.num_discarded_packets(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 188 | } |
| 189 | |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 190 | void VCMReceiver::SetNackMode(VCMNackMode nackMode, |
pkasting@chromium.org | 16825b1 | 2015-01-12 21:51:21 +0000 | [diff] [blame] | 191 | int64_t low_rtt_nack_threshold_ms, |
| 192 | int64_t high_rtt_nack_threshold_ms) { |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 193 | CriticalSectionScoped cs(crit_sect_); |
| 194 | // Default to always having NACK enabled in hybrid mode. |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 195 | jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms, |
| 196 | high_rtt_nack_threshold_ms); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 197 | } |
| 198 | |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 199 | void VCMReceiver::SetNackSettings(size_t max_nack_list_size, |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 200 | int max_packet_age_to_nack, |
| 201 | int max_incomplete_time_ms) { |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 202 | jitter_buffer_.SetNackSettings(max_nack_list_size, |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 203 | max_packet_age_to_nack, |
| 204 | max_incomplete_time_ms); |
stefan@webrtc.org | becf9c8 | 2013-02-01 15:09:57 +0000 | [diff] [blame] | 205 | } |
| 206 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 207 | VCMNackMode VCMReceiver::NackMode() const { |
| 208 | CriticalSectionScoped cs(crit_sect_); |
| 209 | return jitter_buffer_.nack_mode(); |
stefan@webrtc.org | 791eec7 | 2011-10-11 07:53:43 +0000 | [diff] [blame] | 210 | } |
| 211 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 212 | VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list, |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 213 | uint16_t size, |
| 214 | uint16_t* nack_list_length) { |
| 215 | bool request_key_frame = false; |
| 216 | uint16_t* internal_nack_list = jitter_buffer_.GetNackList( |
| 217 | nack_list_length, &request_key_frame); |
stefan@webrtc.org | 34c5da6 | 2014-04-11 14:08:35 +0000 | [diff] [blame] | 218 | assert(*nack_list_length <= size); |
Wan-Teh Chang | 92d9489 | 2015-05-28 13:36:06 -0700 | [diff] [blame] | 219 | if (*nack_list_length > size) { |
| 220 | *nack_list_length = size; |
| 221 | } |
stefan@webrtc.org | a64300a | 2013-03-04 15:24:40 +0000 | [diff] [blame] | 222 | if (internal_nack_list != NULL && *nack_list_length > 0) { |
| 223 | 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] | 224 | } |
stefan@webrtc.org | ef14488 | 2013-05-07 19:16:33 +0000 | [diff] [blame] | 225 | if (request_key_frame) { |
| 226 | return kNackKeyFrameRequest; |
| 227 | } |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 228 | return kNackOk; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 229 | } |
| 230 | |
mikhal@webrtc.org | dbf6a81 | 2013-08-21 20:40:47 +0000 | [diff] [blame] | 231 | void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) { |
| 232 | jitter_buffer_.SetDecodeErrorMode(decode_error_mode); |
mikhal@webrtc.org | dc3cd21 | 2013-04-25 20:27:04 +0000 | [diff] [blame] | 233 | } |
| 234 | |
agalusza@google.com | a7e360e | 2013-08-01 03:15:08 +0000 | [diff] [blame] | 235 | VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const { |
agalusza@google.com | a7e360e | 2013-08-01 03:15:08 +0000 | [diff] [blame] | 236 | return jitter_buffer_.decode_error_mode(); |
mikhal@webrtc.org | dc3cd21 | 2013-04-25 20:27:04 +0000 | [diff] [blame] | 237 | } |
| 238 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 239 | int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) { |
| 240 | CriticalSectionScoped cs(crit_sect_); |
| 241 | if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) { |
| 242 | return -1; |
| 243 | } |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 244 | max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs; |
mikhal@webrtc.org | dbd6a6d | 2013-04-17 16:23:22 +0000 | [diff] [blame] | 245 | // Initializing timing to the desired delay. |
mikhal@webrtc.org | adc64a7 | 2013-05-30 16:20:18 +0000 | [diff] [blame] | 246 | timing_->set_min_playout_delay(desired_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 247 | return 0; |
| 248 | } |
| 249 | |
mikhal@webrtc.org | 381da4b | 2013-04-25 21:45:29 +0000 | [diff] [blame] | 250 | int VCMReceiver::RenderBufferSizeMs() { |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 251 | uint32_t timestamp_start = 0u; |
| 252 | uint32_t timestamp_end = 0u; |
| 253 | // Render timestamps are computed just prior to decoding. Therefore this is |
| 254 | // only an estimate based on frames' timestamps and current timing state. |
| 255 | jitter_buffer_.RenderBufferSize(×tamp_start, ×tamp_end); |
| 256 | if (timestamp_start == timestamp_end) { |
| 257 | return 0; |
| 258 | } |
| 259 | // Update timing. |
| 260 | const int64_t now_ms = clock_->TimeInMilliseconds(); |
mikhal@webrtc.org | adc64a7 | 2013-05-30 16:20:18 +0000 | [diff] [blame] | 261 | timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs()); |
mikhal@webrtc.org | 759b041 | 2013-05-07 16:36:00 +0000 | [diff] [blame] | 262 | // Get render timestamps. |
| 263 | uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms); |
| 264 | uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms); |
| 265 | return render_end - render_start; |
mikhal@webrtc.org | 381da4b | 2013-04-25 21:45:29 +0000 | [diff] [blame] | 266 | } |
pbos@webrtc.org | ce4e9a3 | 2014-12-18 13:50:16 +0000 | [diff] [blame] | 267 | |
pbos@webrtc.org | 5570769 | 2014-12-19 15:45:03 +0000 | [diff] [blame] | 268 | void VCMReceiver::RegisterStatsCallback( |
| 269 | VCMReceiveStatisticsCallback* callback) { |
| 270 | jitter_buffer_.RegisterStatsCallback(callback); |
pbos@webrtc.org | ce4e9a3 | 2014-12-18 13:50:16 +0000 | [diff] [blame] | 271 | } |
| 272 | |
stefan@webrtc.org | 1ea4b50 | 2013-01-07 08:49:41 +0000 | [diff] [blame] | 273 | } // namespace webrtc |