Running FrameBuffer on task queue.

This prepares for running WebRTC in simulated time where event::Wait
based timing doesn't work.

Bug: webrtc:10365
Change-Id: Ia0f9b1cc8e3c8c27a38e45b40487050a4699d8cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129962
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27422}
This commit is contained in:
Sebastian Jansson 2019-04-02 15:08:14 +02:00 committed by Commit Bot
parent d98cbd8f91
commit 13943b7b7f
5 changed files with 519 additions and 255 deletions

View file

@ -151,6 +151,7 @@ rtc_static_library("video_coding") {
"..:module_api_public",
"../../api:fec_controller_api",
"../../api:rtp_headers",
"../../api/task_queue:global_task_queue_factory",
"../../api/units:data_rate",
"../../api/video:builtin_video_bitrate_allocator_factory",
"../../api/video:encoded_frame",
@ -170,6 +171,7 @@ rtc_static_library("video_coding") {
"../../rtc_base/experiments:jitter_upper_bound_experiment",
"../../rtc_base/experiments:rtt_mult_experiment",
"../../rtc_base/system:fallthrough",
"../../rtc_base/task_utils:repeating_task",
"../../rtc_base/third_party/base64",
"../../rtc_base/time:timestamp_extrapolator",
"../../system_wrappers",

View file

@ -17,6 +17,8 @@
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/task_queue/global_task_queue_factory.h"
#include "api/video/encoded_image.h"
#include "api/video/video_timing.h"
#include "modules/video_coding/include/video_coding_defines.h"
@ -45,14 +47,30 @@ constexpr int kMaxFramesHistory = 1 << 13;
constexpr int kMaxAllowedFrameDelayMs = 5;
constexpr int64_t kLogNonDecodedIntervalMs = 5000;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateQueue(
TaskQueueFactory* task_queue_factory) {
if (!task_queue_factory)
task_queue_factory = &GlobalTaskQueueFactory();
return task_queue_factory->CreateTaskQueue("FrameBuffer",
TaskQueueFactory::Priority::HIGH);
}
} // namespace
FrameBuffer::FrameBuffer(Clock* clock,
VCMJitterEstimator* jitter_estimator,
VCMTiming* timing,
VCMReceiveStatisticsCallback* stats_proxy)
: FrameBuffer(clock, nullptr, jitter_estimator, timing, stats_proxy) {}
FrameBuffer::FrameBuffer(Clock* clock,
TaskQueueFactory* task_queue_factory,
VCMJitterEstimator* jitter_estimator,
VCMTiming* timing,
VCMReceiveStatisticsCallback* stats_callback)
: decoded_frames_history_(kMaxFramesHistory),
clock_(clock),
use_task_queue_(task_queue_factory != nullptr),
jitter_estimator_(jitter_estimator),
timing_(timing),
inter_frame_delay_(clock_->TimeInMilliseconds()),
@ -61,14 +79,69 @@ FrameBuffer::FrameBuffer(Clock* clock,
stats_callback_(stats_callback),
last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
add_rtt_to_playout_delay_(
webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")) {}
webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")),
task_queue_(CreateQueue(task_queue_factory)) {}
FrameBuffer::~FrameBuffer() {}
void FrameBuffer::NextFrame(
int64_t max_wait_time_ms,
bool keyframe_required,
std::function<void(std::unique_ptr<EncodedFrame>, ReturnReason)> handler) {
RTC_DCHECK(use_task_queue_);
TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame");
int64_t latest_return_time_ms =
clock_->TimeInMilliseconds() + max_wait_time_ms;
task_queue_.PostTask([=] {
RTC_DCHECK_RUN_ON(&task_queue_);
rtc::CritScope lock(&crit_);
if (stopped_) {
return;
}
latest_return_time_ms_ = latest_return_time_ms;
keyframe_required_ = keyframe_required;
frame_handler_ = handler;
NextFrameOnQueue();
});
}
void FrameBuffer::NextFrameOnQueue() {
RTC_DCHECK(use_task_queue_);
RTC_DCHECK(!callback_task_.Running());
int64_t wait_ms = UpdateFramesToDecode(clock_->TimeInMilliseconds());
callback_task_ = RepeatingTaskHandle::DelayedStart(
task_queue_.Get(), TimeDelta::ms(wait_ms), [this] {
// If this task has not been cancelled, we did not get any new frames
// while waiting. Continue with frame delivery.
RTC_DCHECK_RUN_ON(&task_queue_);
rtc::CritScope lock(&crit_);
if (!frames_to_decode_.empty()) {
// We have frames, deliver!
frame_handler_(absl::WrapUnique(GetFrameToDecode()), kFrameFound);
frame_handler_ = {};
callback_task_.Stop();
return TimeDelta::Zero(); // Ignored.
} else if (clock_->TimeInMilliseconds() >= latest_return_time_ms_) {
// We have timed out, signal this and stop repeating.
frame_handler_(nullptr, kTimeout);
frame_handler_ = {};
callback_task_.Stop();
return TimeDelta::Zero(); // Ignored.
} else {
// If there's no frames to decode and there is still time left, it
// means that the frame buffer was cleared between creation and
// execution of this task. Continue waiting for the remaining time.
int64_t wait_ms = UpdateFramesToDecode(clock_->TimeInMilliseconds());
return TimeDelta::ms(wait_ms);
}
});
}
FrameBuffer::ReturnReason FrameBuffer::NextFrame(
int64_t max_wait_time_ms,
std::unique_ptr<EncodedFrame>* frame_out,
bool keyframe_required) {
RTC_DCHECK(!use_task_queue_);
TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame");
int64_t latest_return_time_ms =
clock_->TimeInMilliseconds() + max_wait_time_ms;
@ -83,183 +156,25 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
if (stopped_)
return kStopped;
wait_ms = max_wait_time_ms;
// Need to hold |crit_| in order to access frames_to_decode_. therefore we
// Need to hold |crit_| in order to access the members. therefore we
// set it here in the loop instead of outside the loop in order to not
// acquire the lock unnecessarily.
frames_to_decode_.clear();
// |last_continuous_frame_| may be empty below, but nullopt is smaller
// than everything else and loop will immediately terminate as expected.
for (auto frame_it = frames_.begin();
frame_it != frames_.end() &&
frame_it->first <= last_continuous_frame_;
++frame_it) {
if (!frame_it->second.continuous ||
frame_it->second.num_missing_decodable > 0) {
continue;
}
EncodedFrame* frame = frame_it->second.frame.get();
if (keyframe_required && !frame->is_keyframe())
continue;
auto last_decoded_frame_timestamp =
decoded_frames_history_.GetLastDecodedFrameTimestamp();
// TODO(https://bugs.webrtc.org/9974): consider removing this check
// as it may make a stream undecodable after a very long delay between
// frames.
if (last_decoded_frame_timestamp &&
AheadOf(*last_decoded_frame_timestamp, frame->Timestamp())) {
continue;
}
// Only ever return all parts of a superframe. Therefore skip this
// frame if it's not a beginning of a superframe.
if (frame->inter_layer_predicted) {
continue;
}
// Gather all remaining frames for the same superframe.
std::vector<FrameMap::iterator> current_superframe;
current_superframe.push_back(frame_it);
bool last_layer_completed =
frame_it->second.frame->is_last_spatial_layer;
FrameMap::iterator next_frame_it = frame_it;
while (true) {
++next_frame_it;
if (next_frame_it == frames_.end() ||
next_frame_it->first.picture_id != frame->id.picture_id ||
!next_frame_it->second.continuous) {
break;
}
// Check if the next frame has some undecoded references other than
// the previous frame in the same superframe.
size_t num_allowed_undecoded_refs =
(next_frame_it->second.frame->inter_layer_predicted) ? 1 : 0;
if (next_frame_it->second.num_missing_decodable >
num_allowed_undecoded_refs) {
break;
}
// All frames in the superframe should have the same timestamp.
if (frame->Timestamp() != next_frame_it->second.frame->Timestamp()) {
RTC_LOG(LS_WARNING)
<< "Frames in a single superframe have different"
" timestamps. Skipping undecodable superframe.";
break;
}
current_superframe.push_back(next_frame_it);
last_layer_completed =
next_frame_it->second.frame->is_last_spatial_layer;
}
// Check if the current superframe is complete.
// TODO(bugs.webrtc.org/10064): consider returning all available to
// decode frames even if the superframe is not complete yet.
if (!last_layer_completed) {
continue;
}
frames_to_decode_ = std::move(current_superframe);
if (frame->RenderTime() == -1) {
frame->SetRenderTime(
timing_->RenderTimeMs(frame->Timestamp(), now_ms));
}
wait_ms = timing_->MaxWaitingTime(frame->RenderTime(), now_ms);
// This will cause the frame buffer to prefer high framerate rather
// than high resolution in the case of the decoder not decoding fast
// enough and the stream has multiple spatial and temporal layers.
// For multiple temporal layers it may cause non-base layer frames to be
// skipped if they are late.
if (wait_ms < -kMaxAllowedFrameDelayMs)
continue;
break;
}
} // rtc::Critscope lock(&crit_);
wait_ms = std::min<int64_t>(wait_ms, latest_return_time_ms - now_ms);
wait_ms = std::max<int64_t>(wait_ms, 0);
keyframe_required_ = keyframe_required;
latest_return_time_ms_ = latest_return_time_ms;
wait_ms = UpdateFramesToDecode(now_ms);
}
} while (new_continuous_frame_event_.Wait(wait_ms));
{
rtc::CritScope lock(&crit_);
now_ms = clock_->TimeInMilliseconds();
// TODO(ilnik): remove |frames_out| use frames_to_decode_ directly.
std::vector<EncodedFrame*> frames_out;
if (!frames_to_decode_.empty()) {
bool superframe_delayed_by_retransmission = false;
size_t superframe_size = 0;
EncodedFrame* first_frame = frames_to_decode_[0]->second.frame.get();
int64_t render_time_ms = first_frame->RenderTime();
int64_t receive_time_ms = first_frame->ReceivedTime();
// Gracefully handle bad RTP timestamps and render time issues.
if (HasBadRenderTiming(*first_frame, now_ms)) {
jitter_estimator_->Reset();
timing_->Reset();
render_time_ms =
timing_->RenderTimeMs(first_frame->Timestamp(), now_ms);
}
for (FrameMap::iterator& frame_it : frames_to_decode_) {
RTC_DCHECK(frame_it != frames_.end());
EncodedFrame* frame = frame_it->second.frame.release();
frame->SetRenderTime(render_time_ms);
superframe_delayed_by_retransmission |=
frame->delayed_by_retransmission();
receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime());
superframe_size += frame->size();
PropagateDecodability(frame_it->second);
decoded_frames_history_.InsertDecoded(frame_it->first,
frame->Timestamp());
// Remove decoded frame and all undecoded frames before it.
frames_.erase(frames_.begin(), ++frame_it);
frames_out.push_back(frame);
}
if (!superframe_delayed_by_retransmission) {
int64_t frame_delay;
if (inter_frame_delay_.CalculateDelay(first_frame->Timestamp(),
&frame_delay, receive_time_ms)) {
jitter_estimator_->UpdateEstimate(frame_delay, superframe_size);
}
float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
if (RttMultExperiment::RttMultEnabled()) {
rtt_mult = RttMultExperiment::GetRttMultValue();
}
timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
timing_->UpdateCurrentDelay(render_time_ms, now_ms);
} else {
if (RttMultExperiment::RttMultEnabled() || add_rtt_to_playout_delay_)
jitter_estimator_->FrameNacked();
}
UpdateJitterDelay();
UpdateTimingFrameInfo();
}
if (!frames_out.empty()) {
if (frames_out.size() == 1) {
frame_out->reset(frames_out[0]);
} else {
frame_out->reset(CombineAndDeleteFrames(frames_out));
}
frame_out->reset(GetFrameToDecode());
return kFrameFound;
}
} // rtc::Critscope lock(&crit_)
}
if (latest_return_time_ms - now_ms > 0) {
if (latest_return_time_ms - clock_->TimeInMilliseconds() > 0) {
// If |next_frame_it_ == frames_.end()| and there is still time left, it
// means that the frame buffer was cleared as the thread in this function
// was waiting to acquire |crit_| in order to return. Wait for the
@ -269,6 +184,166 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
return kTimeout;
}
int64_t FrameBuffer::UpdateFramesToDecode(int64_t now_ms) {
int64_t wait_ms = latest_return_time_ms_ - now_ms;
frames_to_decode_.clear();
// |last_continuous_frame_| may be empty below, but nullopt is smaller
// than everything else and loop will immediately terminate as expected.
for (auto frame_it = frames_.begin();
frame_it != frames_.end() && frame_it->first <= last_continuous_frame_;
++frame_it) {
if (!frame_it->second.continuous ||
frame_it->second.num_missing_decodable > 0) {
continue;
}
EncodedFrame* frame = frame_it->second.frame.get();
if (keyframe_required_ && !frame->is_keyframe())
continue;
auto last_decoded_frame_timestamp =
decoded_frames_history_.GetLastDecodedFrameTimestamp();
// TODO(https://bugs.webrtc.org/9974): consider removing this check
// as it may make a stream undecodable after a very long delay between
// frames.
if (last_decoded_frame_timestamp &&
AheadOf(*last_decoded_frame_timestamp, frame->Timestamp())) {
continue;
}
// Only ever return all parts of a superframe. Therefore skip this
// frame if it's not a beginning of a superframe.
if (frame->inter_layer_predicted) {
continue;
}
// Gather all remaining frames for the same superframe.
std::vector<FrameMap::iterator> current_superframe;
current_superframe.push_back(frame_it);
bool last_layer_completed = frame_it->second.frame->is_last_spatial_layer;
FrameMap::iterator next_frame_it = frame_it;
while (true) {
++next_frame_it;
if (next_frame_it == frames_.end() ||
next_frame_it->first.picture_id != frame->id.picture_id ||
!next_frame_it->second.continuous) {
break;
}
// Check if the next frame has some undecoded references other than
// the previous frame in the same superframe.
size_t num_allowed_undecoded_refs =
(next_frame_it->second.frame->inter_layer_predicted) ? 1 : 0;
if (next_frame_it->second.num_missing_decodable >
num_allowed_undecoded_refs) {
break;
}
// All frames in the superframe should have the same timestamp.
if (frame->Timestamp() != next_frame_it->second.frame->Timestamp()) {
RTC_LOG(LS_WARNING) << "Frames in a single superframe have different"
" timestamps. Skipping undecodable superframe.";
break;
}
current_superframe.push_back(next_frame_it);
last_layer_completed = next_frame_it->second.frame->is_last_spatial_layer;
}
// Check if the current superframe is complete.
// TODO(bugs.webrtc.org/10064): consider returning all available to
// decode frames even if the superframe is not complete yet.
if (!last_layer_completed) {
continue;
}
frames_to_decode_ = std::move(current_superframe);
if (frame->RenderTime() == -1) {
frame->SetRenderTime(timing_->RenderTimeMs(frame->Timestamp(), now_ms));
}
wait_ms = timing_->MaxWaitingTime(frame->RenderTime(), now_ms);
// This will cause the frame buffer to prefer high framerate rather
// than high resolution in the case of the decoder not decoding fast
// enough and the stream has multiple spatial and temporal layers.
// For multiple temporal layers it may cause non-base layer frames to be
// skipped if they are late.
if (wait_ms < -kMaxAllowedFrameDelayMs)
continue;
break;
}
wait_ms = std::min<int64_t>(wait_ms, latest_return_time_ms_ - now_ms);
wait_ms = std::max<int64_t>(wait_ms, 0);
return wait_ms;
}
EncodedFrame* FrameBuffer::GetFrameToDecode() {
int64_t now_ms = clock_->TimeInMilliseconds();
// TODO(ilnik): remove |frames_out| use frames_to_decode_ directly.
std::vector<EncodedFrame*> frames_out;
RTC_DCHECK(!frames_to_decode_.empty());
bool superframe_delayed_by_retransmission = false;
size_t superframe_size = 0;
EncodedFrame* first_frame = frames_to_decode_[0]->second.frame.get();
int64_t render_time_ms = first_frame->RenderTime();
int64_t receive_time_ms = first_frame->ReceivedTime();
// Gracefully handle bad RTP timestamps and render time issues.
if (HasBadRenderTiming(*first_frame, now_ms)) {
jitter_estimator_->Reset();
timing_->Reset();
render_time_ms = timing_->RenderTimeMs(first_frame->Timestamp(), now_ms);
}
for (FrameMap::iterator& frame_it : frames_to_decode_) {
RTC_DCHECK(frame_it != frames_.end());
EncodedFrame* frame = frame_it->second.frame.release();
frame->SetRenderTime(render_time_ms);
superframe_delayed_by_retransmission |= frame->delayed_by_retransmission();
receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime());
superframe_size += frame->size();
PropagateDecodability(frame_it->second);
decoded_frames_history_.InsertDecoded(frame_it->first, frame->Timestamp());
// Remove decoded frame and all undecoded frames before it.
frames_.erase(frames_.begin(), ++frame_it);
frames_out.push_back(frame);
}
if (!superframe_delayed_by_retransmission) {
int64_t frame_delay;
if (inter_frame_delay_.CalculateDelay(first_frame->Timestamp(),
&frame_delay, receive_time_ms)) {
jitter_estimator_->UpdateEstimate(frame_delay, superframe_size);
}
float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
if (RttMultExperiment::RttMultEnabled()) {
rtt_mult = RttMultExperiment::GetRttMultValue();
}
timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
timing_->UpdateCurrentDelay(render_time_ms, now_ms);
} else {
if (RttMultExperiment::RttMultEnabled() || add_rtt_to_playout_delay_)
jitter_estimator_->FrameNacked();
}
UpdateJitterDelay();
UpdateTimingFrameInfo();
if (frames_out.size() == 1) {
return frames_out[0];
} else {
return CombineAndDeleteFrames(frames_out);
}
}
bool FrameBuffer::HasBadRenderTiming(const EncodedFrame& frame,
int64_t now_ms) {
// Assume that render timing errors are due to changes in the video stream.
@ -297,33 +372,63 @@ bool FrameBuffer::HasBadRenderTiming(const EncodedFrame& frame,
return false;
}
void FrameBuffer::SafePost(std::function<void()> func) {
if (!use_task_queue_) {
func();
} else {
task_queue_.PostTask(func);
}
}
void FrameBuffer::SetProtectionMode(VCMVideoProtection mode) {
TRACE_EVENT0("webrtc", "FrameBuffer::SetProtectionMode");
rtc::CritScope lock(&crit_);
protection_mode_ = mode;
SafePost([this, mode] {
rtc::CritScope lock(&crit_);
protection_mode_ = mode;
});
}
void FrameBuffer::Start() {
TRACE_EVENT0("webrtc", "FrameBuffer::Start");
rtc::CritScope lock(&crit_);
stopped_ = false;
SafePost([this] {
rtc::CritScope lock(&crit_);
stopped_ = false;
});
}
void FrameBuffer::Stop() {
TRACE_EVENT0("webrtc", "FrameBuffer::Stop");
rtc::CritScope lock(&crit_);
stopped_ = true;
new_continuous_frame_event_.Set();
if (!use_task_queue_) {
rtc::CritScope lock(&crit_);
stopped_ = true;
new_continuous_frame_event_.Set();
} else {
rtc::Event done;
task_queue_.PostTask([this, &done] {
rtc::CritScope lock(&crit_);
stopped_ = true;
if (frame_handler_) {
RTC_DCHECK(callback_task_.Running());
callback_task_.Stop();
frame_handler_ = {};
}
done.Set();
});
done.Wait(rtc::Event::kForever);
}
}
void FrameBuffer::Clear() {
rtc::CritScope lock(&crit_);
ClearFramesAndHistory();
SafePost([this] {
rtc::CritScope lock(&crit_);
ClearFramesAndHistory();
});
}
void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
rtc::CritScope lock(&crit_);
jitter_estimator_->UpdateRtt(rtt_ms);
SafePost([this, rtt_ms] {
rtc::CritScope lock(&crit_);
jitter_estimator_->UpdateRtt(rtt_ms);
});
}
bool FrameBuffer::ValidReferences(const EncodedFrame& frame) const {
@ -384,6 +489,22 @@ bool FrameBuffer::IsCompleteSuperFrame(const EncodedFrame& frame) {
return true;
}
void FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame,
std::function<void(int64_t)> picture_id_handler) {
struct InsertFrameTask {
void operator()() {
RTC_DCHECK_RUN_ON(&frame_buffer->task_queue_);
int64_t last_continuous_pid = frame_buffer->InsertFrame(std::move(frame));
picture_id_handler(last_continuous_pid);
}
FrameBuffer* frame_buffer;
std::unique_ptr<EncodedFrame> frame;
std::function<void(int64_t)> picture_id_handler;
};
task_queue_.PostTask(
InsertFrameTask{this, std::move(frame), std::move(picture_id_handler)});
}
int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
RTC_DCHECK(frame);
@ -487,9 +608,14 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
last_continuous_picture_id = last_continuous_frame_->picture_id;
// Since we now have new continuous frames there might be a better frame
// to return from NextFrame. Signal that thread so that it again can choose
// which frame to return.
new_continuous_frame_event_.Set();
// to return from NextFrame.
if (!use_task_queue_) {
new_continuous_frame_event_.Set();
} else if (callback_task_.Running()) {
RTC_CHECK(frame_handler_);
callback_task_.Stop();
NextFrameOnQueue();
}
}
return last_continuous_picture_id;

View file

@ -27,6 +27,8 @@
#include "rtc_base/event.h"
#include "rtc_base/experiments/rtt_mult_experiment.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
@ -45,7 +47,13 @@ class FrameBuffer {
FrameBuffer(Clock* clock,
VCMJitterEstimator* jitter_estimator,
VCMTiming* timing,
VCMReceiveStatisticsCallback* stats_proxy);
VCMReceiveStatisticsCallback* stats_callback);
FrameBuffer(Clock* clock,
TaskQueueFactory* task_queue_factory,
VCMJitterEstimator* jitter_estimator,
VCMTiming* timing,
VCMReceiveStatisticsCallback* stats_callback);
virtual ~FrameBuffer();
@ -54,6 +62,9 @@ class FrameBuffer {
// TODO(philipel): Return a VideoLayerFrameId and not only the picture id.
int64_t InsertFrame(std::unique_ptr<EncodedFrame> frame);
void InsertFrame(std::unique_ptr<EncodedFrame> frame,
std::function<void(int64_t)> picture_id_handler);
// Get the next frame for decoding. Will return at latest after
// |max_wait_time_ms|.
// - If a frame is available within |max_wait_time_ms| it will return
@ -64,6 +75,10 @@ class FrameBuffer {
ReturnReason NextFrame(int64_t max_wait_time_ms,
std::unique_ptr<EncodedFrame>* frame_out,
bool keyframe_required = false);
void NextFrame(
int64_t max_wait_time_ms,
bool keyframe_required,
std::function<void(std::unique_ptr<EncodedFrame>, ReturnReason)> handler);
// Tells the FrameBuffer which protection mode that is in use. Affects
// the frame timing.
@ -115,9 +130,16 @@ class FrameBuffer {
using FrameMap = std::map<VideoLayerFrameId, FrameInfo>;
void SafePost(std::function<void()> func);
// Check that the references of |frame| are valid.
bool ValidReferences(const EncodedFrame& frame) const;
void NextFrameOnQueue() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
int64_t UpdateFramesToDecode(int64_t now_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
EncodedFrame* GetFrameToDecode() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Update all directly dependent and indirectly dependent frames and mark
// them as continuous if all their references has been fulfilled.
void PropagateContinuity(FrameMap::iterator start)
@ -158,9 +180,19 @@ class FrameBuffer {
FrameMap frames_ RTC_GUARDED_BY(crit_);
DecodedFramesHistory decoded_frames_history_ RTC_GUARDED_BY(crit_);
// TODO(srte): Remove this lock when always running on task queue.
rtc::CriticalSection crit_;
Clock* const clock_;
const bool use_task_queue_;
RepeatingTaskHandle callback_task_ RTC_GUARDED_BY(crit_);
std::function<void(std::unique_ptr<EncodedFrame>, ReturnReason)>
frame_handler_ RTC_GUARDED_BY(crit_);
int64_t latest_return_time_ms_ RTC_GUARDED_BY(crit_);
bool keyframe_required_ RTC_GUARDED_BY(crit_);
rtc::Event new_continuous_frame_event_;
VCMJitterEstimator* const jitter_estimator_ RTC_GUARDED_BY(crit_);
VCMTiming* const timing_ RTC_GUARDED_BY(crit_);
VCMInterFrameDelay inter_frame_delay_ RTC_GUARDED_BY(crit_);
@ -174,6 +206,8 @@ class FrameBuffer {
const bool add_rtt_to_playout_delay_;
// Defined last so it is destroyed before other members.
rtc::TaskQueue task_queue_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
};

View file

@ -56,6 +56,10 @@
namespace webrtc {
namespace {
using video_coding::EncodedFrame;
using ReturnReason = video_coding::FrameBuffer::ReturnReason;
constexpr int kMinBaseMinimumDelayMs = 0;
constexpr int kMaxBaseMinimumDelayMs = 10000;
@ -184,6 +188,8 @@ VideoReceiveStream::VideoReceiveStream(
num_cpu_cores_(num_cpu_cores),
process_thread_(process_thread),
clock_(clock),
use_task_queue_(
!field_trial::IsDisabled("WebRTC-Video-DecodeOnTaskQueue")),
decode_thread_(&DecodeThreadFunction,
this,
"DecodingThread",
@ -212,7 +218,10 @@ VideoReceiveStream::VideoReceiveStream(
.value_or(kMaxWaitForKeyFrameMs)),
max_wait_for_frame_ms_(KeyframeIntervalSettings::ParseFromFieldTrials()
.MaxWaitForFrameMs()
.value_or(kMaxWaitForFrameMs)) {
.value_or(kMaxWaitForFrameMs)),
decode_queue_(task_queue_factory_->CreateTaskQueue(
"DecodingQueue",
TaskQueueFactory::Priority::HIGH)) {
RTC_LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
RTC_DCHECK(config_.renderer);
@ -237,7 +246,8 @@ VideoReceiveStream::VideoReceiveStream(
jitter_estimator_.reset(new VCMJitterEstimator(clock_));
frame_buffer_.reset(new video_coding::FrameBuffer(
clock_, jitter_estimator_.get(), timing_.get(), &stats_proxy_));
clock_, use_task_queue_ ? task_queue_factory_ : nullptr,
jitter_estimator_.get(), timing_.get(), &stats_proxy_));
process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE);
@ -308,7 +318,7 @@ void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
void VideoReceiveStream::Start() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
if (decode_thread_.IsRunning()) {
if (decoder_running_) {
return;
}
@ -387,7 +397,17 @@ void VideoReceiveStream::Start() {
// Start the decode thread
video_receiver_.DecoderThreadStarting();
stats_proxy_.DecoderThreadStarting();
decode_thread_.Start();
if (!use_task_queue_) {
decode_thread_.Start();
} else {
decode_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
RTC_DCHECK(decoder_stopped_);
decoder_stopped_ = false;
StartNextDecode();
});
}
decoder_running_ = true;
rtp_video_stream_receiver_.StartReceive();
}
@ -401,13 +421,24 @@ void VideoReceiveStream::Stop() {
frame_buffer_->Stop();
call_stats_->DeregisterStatsObserver(this);
if (decode_thread_.IsRunning()) {
if (decoder_running_) {
// TriggerDecoderShutdown will release any waiting decoder thread and make
// it stop immediately, instead of waiting for a timeout. Needs to be called
// before joining the decoder thread.
video_receiver_.TriggerDecoderShutdown();
if (!use_task_queue_) {
decode_thread_.Stop();
} else {
rtc::Event done;
decode_queue_.PostTask([this, &done] {
RTC_DCHECK_RUN_ON(&decode_queue_);
decoder_stopped_ = true;
done.Set();
});
done.Wait(rtc::Event::kForever);
}
decoder_running_ = false;
decode_thread_.Stop();
video_receiver_.DecoderThreadStopped();
stats_proxy_.DecoderThreadStopped();
// Deregister external decoders so they are no longer running during
@ -511,10 +542,17 @@ void VideoReceiveStream::OnCompleteFrame(
frame_maximum_playout_delay_ms_ = playout_delay.max_ms;
UpdatePlayoutDelays();
}
int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
if (last_continuous_pid != -1)
rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
if (!use_task_queue_) {
int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
if (last_continuous_pid != -1)
rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
} else {
frame_buffer_->InsertFrame(
std::move(frame), [this](int64_t last_continuous_pid) {
if (last_continuous_pid != -1)
rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
});
}
}
void VideoReceiveStream::OnData(uint64_t channel_id,
@ -562,6 +600,51 @@ void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
UpdatePlayoutDelays();
}
int64_t VideoReceiveStream::GetWaitMs() const {
return keyframe_required_ ? max_wait_for_keyframe_ms_
: max_wait_for_frame_ms_;
}
void VideoReceiveStream::StartNextDecode() {
RTC_DCHECK(use_task_queue_);
TRACE_EVENT0("webrtc", "VideoReceiveStream::StartNextDecode");
struct DecodeTask {
void operator()() {
RTC_DCHECK_RUN_ON(&stream->decode_queue_);
if (stream->decoder_stopped_)
return;
if (frame) {
stream->HandleEncodedFrame(std::move(frame));
} else {
stream->HandleFrameBufferTimeout();
}
}
VideoReceiveStream* stream;
std::unique_ptr<EncodedFrame> frame;
};
// TODO(philipel): Call NextFrame with |keyframe_required| argument set when
// downstream project has been fixed.
frame_buffer_->NextFrame(
GetWaitMs(), /*keyframe_required*/ false,
[this](std::unique_ptr<EncodedFrame> frame, ReturnReason res) {
RTC_DCHECK_EQ(frame == nullptr, res == ReturnReason::kTimeout);
RTC_DCHECK_EQ(frame != nullptr, res == ReturnReason::kFrameFound);
decode_queue_.PostTask(DecodeTask{this, std::move(frame)});
// Start the next decode after a delay or when the previous decode is
// finished (as it will be blocked by the queue).
constexpr int kMinDecodeIntervalMs = 1;
decode_queue_.PostDelayedTask(
[this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
if (!decoder_stopped_)
StartNextDecode();
},
kMinDecodeIntervalMs);
});
}
void VideoReceiveStream::DecodeThreadFunction(void* ptr) {
ScopedRegisterThreadForDebugging thread_dbg(RTC_FROM_HERE);
while (static_cast<VideoReceiveStream*>(ptr)->Decode()) {
@ -569,82 +652,87 @@ void VideoReceiveStream::DecodeThreadFunction(void* ptr) {
}
bool VideoReceiveStream::Decode() {
RTC_DCHECK(!use_task_queue_);
TRACE_EVENT0("webrtc", "VideoReceiveStream::Decode");
const int wait_ms =
keyframe_required_ ? max_wait_for_keyframe_ms_ : max_wait_for_frame_ms_;
std::unique_ptr<video_coding::EncodedFrame> frame;
// TODO(philipel): Call NextFrame with |keyframe_required| argument when
// downstream project has been fixed.
video_coding::FrameBuffer::ReturnReason res =
frame_buffer_->NextFrame(wait_ms, &frame);
if (res == video_coding::FrameBuffer::ReturnReason::kStopped) {
frame_buffer_->NextFrame(GetWaitMs(), &frame);
if (res == ReturnReason::kStopped) {
return false;
}
if (frame) {
int64_t now_ms = clock_->TimeInMilliseconds();
RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kFrameFound);
// Current OnPreDecode only cares about QP for VP8.
int qp = -1;
if (frame->CodecSpecific()->codecType == kVideoCodecVP8) {
if (!vp8::GetQp(frame->data(), frame->size(), &qp)) {
RTC_LOG(LS_WARNING) << "Failed to extract QP from VP8 video frame";
}
}
stats_proxy_.OnPreDecode(frame->CodecSpecific()->codecType, qp);
int decode_result = video_receiver_.Decode(frame.get());
if (decode_result == WEBRTC_VIDEO_CODEC_OK ||
decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
keyframe_required_ = false;
frame_decoded_ = true;
rtp_video_stream_receiver_.FrameDecoded(frame->id.picture_id);
if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
RequestKeyFrame();
} else if (!frame_decoded_ || !keyframe_required_ ||
(last_keyframe_request_ms_ + max_wait_for_keyframe_ms_ <
now_ms)) {
keyframe_required_ = true;
// TODO(philipel): Remove this keyframe request when downstream project
// has been fixed.
RequestKeyFrame();
last_keyframe_request_ms_ = now_ms;
}
RTC_DCHECK_EQ(res, ReturnReason::kFrameFound);
HandleEncodedFrame(std::move(frame));
} else {
RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kTimeout);
int64_t now_ms = clock_->TimeInMilliseconds();
absl::optional<int64_t> last_packet_ms =
rtp_video_stream_receiver_.LastReceivedPacketMs();
absl::optional<int64_t> last_keyframe_packet_ms =
rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
// To avoid spamming keyframe requests for a stream that is not active we
// check if we have received a packet within the last 5 seconds.
bool stream_is_active = last_packet_ms && now_ms - *last_packet_ms < 5000;
if (!stream_is_active)
stats_proxy_.OnStreamInactive();
// If we recently have been receiving packets belonging to a keyframe then
// we assume a keyframe is currently being received.
bool receiving_keyframe =
last_keyframe_packet_ms &&
now_ms - *last_keyframe_packet_ms < max_wait_for_keyframe_ms_;
if (stream_is_active && !receiving_keyframe &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
RTC_LOG(LS_WARNING) << "No decodable frame in " << wait_ms
<< " ms, requesting keyframe.";
RequestKeyFrame();
}
RTC_DCHECK_EQ(res, ReturnReason::kTimeout);
HandleFrameBufferTimeout();
}
return true;
}
void VideoReceiveStream::HandleEncodedFrame(
std::unique_ptr<EncodedFrame> frame) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Current OnPreDecode only cares about QP for VP8.
int qp = -1;
if (frame->CodecSpecific()->codecType == kVideoCodecVP8) {
if (!vp8::GetQp(frame->data(), frame->size(), &qp)) {
RTC_LOG(LS_WARNING) << "Failed to extract QP from VP8 video frame";
}
}
stats_proxy_.OnPreDecode(frame->CodecSpecific()->codecType, qp);
int decode_result = video_receiver_.Decode(frame.get());
if (decode_result == WEBRTC_VIDEO_CODEC_OK ||
decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
keyframe_required_ = false;
frame_decoded_ = true;
rtp_video_stream_receiver_.FrameDecoded(frame->id.picture_id);
if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
RequestKeyFrame();
} else if (!frame_decoded_ || !keyframe_required_ ||
(last_keyframe_request_ms_ + max_wait_for_keyframe_ms_ < now_ms)) {
keyframe_required_ = true;
// TODO(philipel): Remove this keyframe request when downstream project
// has been fixed.
RequestKeyFrame();
last_keyframe_request_ms_ = now_ms;
}
}
void VideoReceiveStream::HandleFrameBufferTimeout() {
int64_t now_ms = clock_->TimeInMilliseconds();
absl::optional<int64_t> last_packet_ms =
rtp_video_stream_receiver_.LastReceivedPacketMs();
absl::optional<int64_t> last_keyframe_packet_ms =
rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
// To avoid spamming keyframe requests for a stream that is not active we
// check if we have received a packet within the last 5 seconds.
bool stream_is_active = last_packet_ms && now_ms - *last_packet_ms < 5000;
if (!stream_is_active)
stats_proxy_.OnStreamInactive();
// If we recently have been receiving packets belonging to a keyframe then
// we assume a keyframe is currently being received.
bool receiving_keyframe =
last_keyframe_packet_ms &&
now_ms - *last_keyframe_packet_ms < max_wait_for_keyframe_ms_;
if (stream_is_active && !receiving_keyframe &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
RTC_LOG(LS_WARNING) << "No decodable frame in " << GetWaitMs()
<< " ms, requesting keyframe.";
RequestKeyFrame();
}
}
void VideoReceiveStream::UpdatePlayoutDelays() const {
const int minimum_delay_ms =
std::max({frame_minimum_playout_delay_ms_, base_minimum_playout_delay_ms_,

View file

@ -23,6 +23,7 @@
#include "modules/video_coding/frame_buffer2.h"
#include "modules/video_coding/video_coding_impl.h"
#include "rtc_base/sequenced_task_checker.h"
#include "rtc_base/task_queue.h"
#include "system_wrappers/include/clock.h"
#include "video/receive_statistics_proxy.h"
#include "video/rtp_streams_synchronizer.h"
@ -129,8 +130,13 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
std::vector<webrtc::RtpSource> GetSources() const override;
private:
int64_t GetWaitMs() const;
void StartNextDecode() RTC_RUN_ON(decode_queue_);
static void DecodeThreadFunction(void* ptr);
bool Decode();
void HandleEncodedFrame(std::unique_ptr<video_coding::EncodedFrame> frame);
void HandleFrameBufferTimeout();
void UpdatePlayoutDelays() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(playout_delay_lock_);
@ -146,10 +152,15 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
ProcessThread* const process_thread_;
Clock* const clock_;
const bool use_task_queue_;
rtc::PlatformThread decode_thread_;
CallStats* const call_stats_;
bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
bool decoder_stopped_ RTC_GUARDED_BY(decode_queue_) = true;
ReceiveStatisticsProxy stats_proxy_;
// Shared by media and rtx stream receivers, since the latter has no RtpRtcp
// module of its own.
@ -165,10 +176,10 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
// TODO(nisse, philipel): Creation and ownership of video encoders should be
// moved to the new VideoStreamDecoder.
std::vector<std::unique_ptr<VideoDecoder>> video_decoders_;
std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
// Members for the new jitter buffer experiment.
std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
@ -204,6 +215,9 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
// Maximum delay as decided by the RTP playout delay extension.
int frame_maximum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
// Defined last so they are destroyed before all other members.
rtc::TaskQueue decode_queue_;
};
} // namespace internal
} // namespace webrtc