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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.org1ea4b502013-01-07 08:49:41 +000021#include "webrtc/system_wrappers/interface/trace.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,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000031 int32_t vcm_id,
32 int32_t receiver_id,
niklase@google.com470e71d2011-07-07 08:21:25 +000033 bool master)
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000034 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
35 vcm_id_(vcm_id),
36 clock_(clock),
37 receiver_id_(receiver_id),
38 master_(master),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000039 jitter_buffer_(clock_, event_factory, vcm_id, receiver_id, master),
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000040 timing_(timing),
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000041 render_wait_event_(event_factory->CreateEvent()),
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +000042 state_(kPassive),
43 max_video_delay_ms_(kMaxVideoDelayMs) {}
niklase@google.com470e71d2011-07-07 08:21:25 +000044
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000045VCMReceiver::~VCMReceiver() {
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000046 render_wait_event_->Set();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000047 delete crit_sect_;
niklase@google.com470e71d2011-07-07 08:21:25 +000048}
49
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000050void VCMReceiver::Reset() {
51 CriticalSectionScoped cs(crit_sect_);
52 if (!jitter_buffer_.Running()) {
53 jitter_buffer_.Start();
54 } else {
55 jitter_buffer_.Flush();
56 }
stefan@webrtc.org2baf5f52013-03-13 08:46:25 +000057 render_wait_event_->Reset();
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000058 if (master_) {
59 state_ = kReceiving;
60 } else {
61 state_ = kPassive;
62 }
henrik.lundin@webrtc.orgbaf6db52011-11-02 18:58:39 +000063}
64
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000065int32_t VCMReceiver::Initialize() {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000066 Reset();
stefan@webrtc.org4f3624d2013-09-20 07:43:17 +000067 CriticalSectionScoped cs(crit_sect_);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000068 if (!master_) {
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000069 SetNackMode(kNoNack, -1, -1);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000070 }
71 return VCM_OK;
72}
73
74void VCMReceiver::UpdateRtt(uint32_t rtt) {
75 jitter_buffer_.UpdateRtt(rtt);
76}
77
stefan@webrtc.orga64300a2013-03-04 15:24:40 +000078int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
79 uint16_t frame_width,
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +000080 uint16_t frame_height) {
stefan@webrtc.orga7dc37d2013-05-23 07:21:05 +000081 if (packet.frameType == kVideoFrameKey) {
82 WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCoding,
83 VCMId(vcm_id_, receiver_id_),
84 "Inserting key frame packet seqnum=%u, timestamp=%u",
85 packet.seqNum, packet.timestamp);
86 }
hclam@chromium.org8c49c1e2013-05-22 21:18:59 +000087
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000088 // Insert the packet into the jitter buffer. The packet can either be empty or
89 // contain media at this point.
90 bool retransmitted = false;
91 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
92 &retransmitted);
93 if (ret == kOldPacket) {
niklase@google.com470e71d2011-07-07 08:21:25 +000094 return VCM_OK;
stefan@webrtc.org3417eb42013-05-21 15:25:53 +000095 } else if (ret == kFlushIndicator) {
96 return VCM_FLUSH_INDICATOR;
97 } else if (ret < 0) {
98 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
99 VCMId(vcm_id_, receiver_id_),
100 "Error inserting packet seqnum=%u, timestamp=%u",
101 packet.seqNum, packet.timestamp);
102 return VCM_JITTER_BUFFER_ERROR;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000103 }
stefan@webrtc.org3417eb42013-05-21 15:25:53 +0000104 if (ret == kCompleteSession && !retransmitted) {
105 // We don't want to include timestamps which have suffered from
106 // retransmission here, since we compensate with extra retransmission
107 // delay within the jitter estimate.
108 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
109 }
110 if (master_) {
111 // Only trace the primary receiver to make it possible to parse and plot
112 // the trace file.
113 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
114 VCMId(vcm_id_, receiver_id_),
115 "Packet seqnum=%u timestamp=%u inserted at %u",
116 packet.seqNum, packet.timestamp,
117 MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000118 }
119 return VCM_OK;
niklase@google.com470e71d2011-07-07 08:21:25 +0000120}
121
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000122VCMEncodedFrame* VCMReceiver::FrameForDecoding(
123 uint16_t max_wait_time_ms,
124 int64_t& next_render_time_ms,
125 bool render_timing,
126 VCMReceiver* dual_receiver) {
stefan@webrtc.orga678a3b2013-01-21 07:42:11 +0000127 const int64_t start_time_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000128 uint32_t frame_timestamp = 0;
129 // Exhaust wait time to get a complete frame for decoding.
130 bool found_frame = jitter_buffer_.NextCompleteTimestamp(
131 max_wait_time_ms, &frame_timestamp);
132
133 if (!found_frame) {
134 // Get an incomplete frame when enabled.
135 const bool dual_receiver_enabled_and_passive = (dual_receiver != NULL &&
136 dual_receiver->State() == kPassive &&
137 dual_receiver->NackMode() == kNack);
138 if (dual_receiver_enabled_and_passive &&
139 !jitter_buffer_.CompleteSequenceWithNextFrame()) {
140 // Jitter buffer state might get corrupt with this frame.
141 dual_receiver->CopyJitterBufferStateFromReceiver(*this);
142 }
143 found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(
144 &frame_timestamp);
145 }
146
147 if (!found_frame) {
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000148 return NULL;
149 }
mikhal@webrtc.orgd3cd5652013-05-03 17:54:18 +0000150
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000151 // We have a frame - Set timing and render timestamp.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000152 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000153 const int64_t now_ms = clock_->TimeInMilliseconds();
154 timing_->UpdateCurrentDelay(frame_timestamp);
155 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
156 // Check render timing.
157 bool timing_error = false;
158 // Assume that render timing errors are due to changes in the video stream.
159 if (next_render_time_ms < 0) {
160 timing_error = true;
pbos@webrtc.orgb5f30292014-03-13 08:53:39 +0000161 } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000162 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
163 VCMId(vcm_id_, receiver_id_),
stefan@webrtc.org554d1582013-09-11 08:45:26 +0000164 "This frame is out of our delay bounds, resetting jitter "
165 "buffer: %d > %d",
pbos@webrtc.orgb5f30292014-03-13 08:53:39 +0000166 static_cast<int>(std::abs(next_render_time_ms - now_ms)),
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000167 max_video_delay_ms_);
168 timing_error = true;
169 } else if (static_cast<int>(timing_->TargetVideoDelay()) >
170 max_video_delay_ms_) {
171 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
172 VCMId(vcm_id_, receiver_id_),
173 "More than %u ms target delay. Flushing jitter buffer and"
174 "resetting timing.", max_video_delay_ms_);
175 timing_error = true;
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000176 }
177
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000178 if (timing_error) {
179 // Timing error => reset timing and flush the jitter buffer.
180 jitter_buffer_.Flush();
stefan@webrtc.org9f557c12013-05-17 12:55:07 +0000181 timing_->Reset();
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000182 return NULL;
183 }
184
185 if (!render_timing) {
186 // Decode frame as close as possible to the render timestamp.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000187 const int32_t available_wait_time = max_wait_time_ms -
188 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
189 uint16_t new_max_wait_time = static_cast<uint16_t>(
190 VCM_MAX(available_wait_time, 0));
191 uint32_t wait_time_ms = timing_->MaxWaitingTime(
192 next_render_time_ms, clock_->TimeInMilliseconds());
193 if (new_max_wait_time < wait_time_ms) {
194 // We're not allowed to wait until the frame is supposed to be rendered,
195 // waiting as long as we're allowed to avoid busy looping, and then return
196 // NULL. Next call to this function might return the frame.
197 render_wait_event_->Wait(max_wait_time_ms);
198 return NULL;
199 }
200 // Wait until it's time to render.
201 render_wait_event_->Wait(wait_time_ms);
202 }
203
204 // Extract the frame from the jitter buffer and set the render time.
205 VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
mikhal@webrtc.org8f86cc82013-05-07 18:05:21 +0000206 if (frame == NULL) {
207 return NULL;
208 }
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000209 frame->SetRenderTime(next_render_time_ms);
hclam@chromium.org1a7b9b92013-07-08 21:31:18 +0000210 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
211 "SetRenderTS", "render_time", next_render_time_ms);
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000212 if (dual_receiver != NULL) {
213 dual_receiver->UpdateState(*frame);
214 }
215 if (!frame->Complete()) {
216 // Update stats for incomplete frames.
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000217 bool retransmitted = false;
218 const int64_t last_packet_time_ms =
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000219 jitter_buffer_.LastPacketTime(frame, &retransmitted);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000220 if (last_packet_time_ms >= 0 && !retransmitted) {
221 // We don't want to include timestamps which have suffered from
222 // retransmission here, since we compensate with extra retransmission
223 // delay within the jitter estimate.
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000224 timing_->IncomingTimestamp(frame_timestamp, last_packet_time_ms);
stefan@webrtc.org4ce19b12013-05-06 13:16:51 +0000225 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000226 }
227 return frame;
niklase@google.com470e71d2011-07-07 08:21:25 +0000228}
229
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000230void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
231 jitter_buffer_.ReleaseFrame(frame);
niklase@google.com470e71d2011-07-07 08:21:25 +0000232}
233
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000234void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
235 uint32_t* framerate) {
236 assert(bitrate);
237 assert(framerate);
238 jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
niklase@google.com470e71d2011-07-07 08:21:25 +0000239}
240
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000241void VCMReceiver::ReceivedFrameCount(VCMFrameCount* frame_count) const {
242 assert(frame_count);
sprang@webrtc.org71f055f2013-12-04 15:09:27 +0000243 std::map<FrameType, uint32_t> counts(jitter_buffer_.FrameStatistics());
244 frame_count->numDeltaFrames = counts[kVideoFrameDelta];
245 frame_count->numKeyFrames = counts[kVideoFrameKey];
niklase@google.com470e71d2011-07-07 08:21:25 +0000246}
247
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000248uint32_t VCMReceiver::DiscardedPackets() const {
249 return jitter_buffer_.num_discarded_packets();
niklase@google.com470e71d2011-07-07 08:21:25 +0000250}
251
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000252void VCMReceiver::SetNackMode(VCMNackMode nackMode,
253 int low_rtt_nack_threshold_ms,
254 int high_rtt_nack_threshold_ms) {
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000255 CriticalSectionScoped cs(crit_sect_);
256 // Default to always having NACK enabled in hybrid mode.
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000257 jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
258 high_rtt_nack_threshold_ms);
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000259 if (!master_) {
260 state_ = kPassive; // The dual decoder defaults to passive.
261 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000262}
263
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000264void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000265 int max_packet_age_to_nack,
266 int max_incomplete_time_ms) {
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000267 jitter_buffer_.SetNackSettings(max_nack_list_size,
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000268 max_packet_age_to_nack,
269 max_incomplete_time_ms);
stefan@webrtc.orgbecf9c82013-02-01 15:09:57 +0000270}
271
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000272VCMNackMode VCMReceiver::NackMode() const {
273 CriticalSectionScoped cs(crit_sect_);
274 return jitter_buffer_.nack_mode();
stefan@webrtc.org791eec72011-10-11 07:53:43 +0000275}
276
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000277VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000278 uint16_t size,
279 uint16_t* nack_list_length) {
280 bool request_key_frame = false;
281 uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
282 nack_list_length, &request_key_frame);
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000283 if (*nack_list_length > size) {
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000284 *nack_list_length = 0;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000285 return kNackNeedMoreMemory;
286 }
stefan@webrtc.orga64300a2013-03-04 15:24:40 +0000287 if (internal_nack_list != NULL && *nack_list_length > 0) {
288 memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000289 }
stefan@webrtc.orgef144882013-05-07 19:16:33 +0000290 if (request_key_frame) {
291 return kNackKeyFrameRequest;
292 }
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000293 return kNackOk;
niklase@google.com470e71d2011-07-07 08:21:25 +0000294}
295
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000296// Decide whether we should change decoder state. This should be done if the
297// dual decoder has caught up with the decoder decoding with packet losses.
298bool VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dual_frame,
299 VCMReceiver& dual_receiver) const {
300 if (dual_frame == NULL) {
niklase@google.com470e71d2011-07-07 08:21:25 +0000301 return false;
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000302 }
303 if (jitter_buffer_.LastDecodedTimestamp() == dual_frame->TimeStamp()) {
304 dual_receiver.UpdateState(kWaitForPrimaryDecode);
305 return true;
306 }
307 return false;
niklase@google.com470e71d2011-07-07 08:21:25 +0000308}
309
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000310void VCMReceiver::CopyJitterBufferStateFromReceiver(
311 const VCMReceiver& receiver) {
312 jitter_buffer_.CopyFrom(receiver.jitter_buffer_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000313}
314
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000315VCMReceiverState VCMReceiver::State() const {
316 CriticalSectionScoped cs(crit_sect_);
317 return state_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000318}
319
mikhal@webrtc.orgdbf6a812013-08-21 20:40:47 +0000320void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
321 jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000322}
323
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000324VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
agalusza@google.coma7e360e2013-08-01 03:15:08 +0000325 return jitter_buffer_.decode_error_mode();
mikhal@webrtc.orgdc3cd212013-04-25 20:27:04 +0000326}
327
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000328int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
329 CriticalSectionScoped cs(crit_sect_);
330 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
331 return -1;
332 }
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000333 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
mikhal@webrtc.orgdbd6a6d2013-04-17 16:23:22 +0000334 // Initializing timing to the desired delay.
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000335 timing_->set_min_playout_delay(desired_delay_ms);
mikhal@webrtc.orgef9f76a2013-02-15 23:22:18 +0000336 return 0;
337}
338
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000339int VCMReceiver::RenderBufferSizeMs() {
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000340 uint32_t timestamp_start = 0u;
341 uint32_t timestamp_end = 0u;
342 // Render timestamps are computed just prior to decoding. Therefore this is
343 // only an estimate based on frames' timestamps and current timing state.
344 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
345 if (timestamp_start == timestamp_end) {
346 return 0;
347 }
348 // Update timing.
349 const int64_t now_ms = clock_->TimeInMilliseconds();
mikhal@webrtc.orgadc64a72013-05-30 16:20:18 +0000350 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
mikhal@webrtc.org759b0412013-05-07 16:36:00 +0000351 // Get render timestamps.
352 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
353 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
354 return render_end - render_start;
mikhal@webrtc.org381da4b2013-04-25 21:45:29 +0000355}
356
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000357void VCMReceiver::UpdateState(VCMReceiverState new_state) {
358 CriticalSectionScoped cs(crit_sect_);
359 assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));
360 state_ = new_state;
niklase@google.com470e71d2011-07-07 08:21:25 +0000361}
362
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000363void VCMReceiver::UpdateState(const VCMEncodedFrame& frame) {
364 if (jitter_buffer_.nack_mode() == kNoNack) {
365 // Dual decoder mode has not been enabled.
366 return;
367 }
368 // Update the dual receiver state.
369 if (frame.Complete() && frame.FrameType() == kVideoFrameKey) {
370 UpdateState(kPassive);
371 }
372 if (State() == kWaitForPrimaryDecode &&
373 frame.Complete() && !frame.MissingFrame()) {
374 UpdateState(kPassive);
375 }
376 if (frame.MissingFrame() || !frame.Complete()) {
377 // State was corrupted, enable dual receiver.
378 UpdateState(kReceiving);
379 }
niklase@google.com470e71d2011-07-07 08:21:25 +0000380}
stefan@webrtc.org1ea4b502013-01-07 08:49:41 +0000381} // namespace webrtc