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webrtc_m130/video/video_receive_stream2.cc
Tommi a98cea863d Remove the PendingTaskSafetyFlag::Pointer type add ScopedTaskSafety. 
ScopedTaskSafety simplifies usage of PendingTaskSafetyFlag,
so this CL also includes ToQueuedTask support for ScopedTaskSafety
and test updates.

This is following up on feedback in the following CL:
https://webrtc-review.googlesource.com/c/src/+/174262

Change-Id: Idd38dfc1914b24a05fdc4ad256b409dcf1795fc0
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174740
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31241}
2020-05-13 14:17:39 +00:00

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/*
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "video/video_receive_stream2.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/video/encoded_image.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "call/rtx_receive_stream.h"
#include "common_video/include/incoming_video_stream.h"
#include "media/base/h264_profile_level_id.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "modules/video_coding/include/video_error_codes.h"
#include "modules/video_coding/timing.h"
#include "modules/video_coding/utility/vp8_header_parser.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/keyframe_interval_settings.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/thread_registry.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "video/call_stats2.h"
#include "video/frame_dumping_decoder.h"
#include "video/receive_statistics_proxy.h"
namespace webrtc {
namespace internal {
constexpr int VideoReceiveStream2::kMaxWaitForKeyFrameMs;
namespace {
using video_coding::EncodedFrame;
using ReturnReason = video_coding::FrameBuffer::ReturnReason;
constexpr int kMinBaseMinimumDelayMs = 0;
constexpr int kMaxBaseMinimumDelayMs = 10000;
constexpr int kMaxWaitForFrameMs = 3000;
// Concrete instance of RecordableEncodedFrame wrapping needed content
// from video_coding::EncodedFrame.
class WebRtcRecordableEncodedFrame : public RecordableEncodedFrame {
public:
explicit WebRtcRecordableEncodedFrame(const EncodedFrame& frame)
: buffer_(frame.GetEncodedData()),
render_time_ms_(frame.RenderTime()),
codec_(frame.CodecSpecific()->codecType),
is_key_frame_(frame.FrameType() == VideoFrameType::kVideoFrameKey),
resolution_{frame.EncodedImage()._encodedWidth,
frame.EncodedImage()._encodedHeight} {
if (frame.ColorSpace()) {
color_space_ = *frame.ColorSpace();
}
}
// VideoEncodedSinkInterface::FrameBuffer
rtc::scoped_refptr<const EncodedImageBufferInterface> encoded_buffer()
const override {
return buffer_;
}
absl::optional<webrtc::ColorSpace> color_space() const override {
return color_space_;
}
VideoCodecType codec() const override { return codec_; }
bool is_key_frame() const override { return is_key_frame_; }
EncodedResolution resolution() const override { return resolution_; }
Timestamp render_time() const override {
return Timestamp::Millis(render_time_ms_);
}
private:
rtc::scoped_refptr<EncodedImageBufferInterface> buffer_;
int64_t render_time_ms_;
VideoCodecType codec_;
bool is_key_frame_;
EncodedResolution resolution_;
absl::optional<webrtc::ColorSpace> color_space_;
};
VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
VideoCodec codec;
memset(&codec, 0, sizeof(codec));
codec.plType = decoder.payload_type;
codec.codecType = PayloadStringToCodecType(decoder.video_format.name);
if (codec.codecType == kVideoCodecVP8) {
*(codec.VP8()) = VideoEncoder::GetDefaultVp8Settings();
} else if (codec.codecType == kVideoCodecVP9) {
*(codec.VP9()) = VideoEncoder::GetDefaultVp9Settings();
} else if (codec.codecType == kVideoCodecH264) {
*(codec.H264()) = VideoEncoder::GetDefaultH264Settings();
} else if (codec.codecType == kVideoCodecMultiplex) {
VideoReceiveStream::Decoder associated_decoder = decoder;
associated_decoder.video_format =
SdpVideoFormat(CodecTypeToPayloadString(kVideoCodecVP9));
VideoCodec associated_codec = CreateDecoderVideoCodec(associated_decoder);
associated_codec.codecType = kVideoCodecMultiplex;
return associated_codec;
}
codec.width = 320;
codec.height = 180;
const int kDefaultStartBitrate = 300;
codec.startBitrate = codec.minBitrate = codec.maxBitrate =
kDefaultStartBitrate;
return codec;
}
// Video decoder class to be used for unknown codecs. Doesn't support decoding
// but logs messages to LS_ERROR.
class NullVideoDecoder : public webrtc::VideoDecoder {
public:
int32_t InitDecode(const webrtc::VideoCodec* codec_settings,
int32_t number_of_cores) override {
RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t Decode(const webrtc::EncodedImage& input_image,
bool missing_frames,
int64_t render_time_ms) override {
RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t RegisterDecodeCompleteCallback(
webrtc::DecodedImageCallback* callback) override {
RTC_LOG(LS_ERROR)
<< "Can't register decode complete callback on NullVideoDecoder.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; }
const char* ImplementationName() const override { return "NullVideoDecoder"; }
};
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Maximum time between frames before resetting the FrameBuffer to avoid RTP
// timestamps wraparound to affect FrameBuffer.
constexpr int kInactiveStreamThresholdMs = 600000; // 10 minutes.
} // namespace
VideoReceiveStream2::VideoReceiveStream2(
TaskQueueFactory* task_queue_factory,
TaskQueueBase* current_queue,
RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
ProcessThread* process_thread,
CallStats* call_stats,
Clock* clock,
VCMTiming* timing)
: task_queue_factory_(task_queue_factory),
transport_adapter_(config.rtcp_send_transport),
config_(std::move(config)),
num_cpu_cores_(num_cpu_cores),
worker_thread_(current_queue),
clock_(clock),
call_stats_(call_stats),
source_tracker_(clock_),
stats_proxy_(&config_, clock_, worker_thread_),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
timing_(timing),
video_receiver_(clock_, timing_.get()),
rtp_video_stream_receiver_(clock_,
&transport_adapter_,
call_stats->AsRtcpRttStats(),
packet_router,
&config_,
rtp_receive_statistics_.get(),
&stats_proxy_,
&stats_proxy_,
process_thread,
this, // NackSender
nullptr, // Use default KeyFrameRequestSender
this, // OnCompleteFrameCallback
config_.frame_decryptor,
config_.frame_transformer),
rtp_stream_sync_(current_queue, this),
max_wait_for_keyframe_ms_(KeyframeIntervalSettings::ParseFromFieldTrials()
.MaxWaitForKeyframeMs()
.value_or(kMaxWaitForKeyFrameMs)),
max_wait_for_frame_ms_(KeyframeIntervalSettings::ParseFromFieldTrials()
.MaxWaitForFrameMs()
.value_or(kMaxWaitForFrameMs)),
decode_queue_(task_queue_factory_->CreateTaskQueue(
"DecodingQueue",
TaskQueueFactory::Priority::HIGH)) {
RTC_LOG(LS_INFO) << "VideoReceiveStream2: " << config_.ToString();
RTC_DCHECK(worker_thread_);
RTC_DCHECK(config_.renderer);
RTC_DCHECK(call_stats_);
module_process_sequence_checker_.Detach();
network_sequence_checker_.Detach();
RTC_DCHECK(!config_.decoders.empty());
std::set<int> decoder_payload_types;
for (const Decoder& decoder : config_.decoders) {
RTC_CHECK(decoder.decoder_factory);
RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
decoder_payload_types.end())
<< "Duplicate payload type (" << decoder.payload_type
<< ") for different decoders.";
decoder_payload_types.insert(decoder.payload_type);
}
timing_->set_render_delay(config_.render_delay_ms);
frame_buffer_.reset(
new video_coding::FrameBuffer(clock_, timing_.get(), &stats_proxy_));
// Register with RtpStreamReceiverController.
media_receiver_ = receiver_controller->CreateReceiver(
config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
if (config_.rtp.rtx_ssrc) {
rtx_receive_stream_ = std::make_unique<RtxReceiveStream>(
&rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
rtx_receiver_ = receiver_controller->CreateReceiver(
config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
} else {
rtp_receive_statistics_->EnableRetransmitDetection(config.rtp.remote_ssrc,
true);
}
}
VideoReceiveStream2::~VideoReceiveStream2() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_LOG(LS_INFO) << "~VideoReceiveStream2: " << config_.ToString();
Stop();
}
void VideoReceiveStream2::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtp_video_stream_receiver_.SignalNetworkState(state);
}
bool VideoReceiveStream2::DeliverRtcp(const uint8_t* packet, size_t length) {
return rtp_video_stream_receiver_.DeliverRtcp(packet, length);
}
void VideoReceiveStream2::SetSync(Syncable* audio_syncable) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtp_stream_sync_.ConfigureSync(audio_syncable);
}
void VideoReceiveStream2::Start() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
if (decoder_running_) {
return;
}
const bool protected_by_fec = config_.rtp.protected_by_flexfec ||
rtp_video_stream_receiver_.IsUlpfecEnabled();
frame_buffer_->Start();
if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
protected_by_fec) {
frame_buffer_->SetProtectionMode(kProtectionNackFEC);
}
transport_adapter_.Enable();
rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
if (config_.enable_prerenderer_smoothing) {
incoming_video_stream_.reset(new IncomingVideoStream(
task_queue_factory_, config_.render_delay_ms, this));
renderer = incoming_video_stream_.get();
} else {
renderer = this;
}
for (const Decoder& decoder : config_.decoders) {
std::unique_ptr<VideoDecoder> video_decoder =
decoder.decoder_factory->LegacyCreateVideoDecoder(decoder.video_format,
config_.stream_id);
// If we still have no valid decoder, we have to create a "Null" decoder
// that ignores all calls. The reason we can get into this state is that the
// old decoder factory interface doesn't have a way to query supported
// codecs.
if (!video_decoder) {
video_decoder = std::make_unique<NullVideoDecoder>();
}
std::string decoded_output_file =
field_trial::FindFullName("WebRTC-DecoderDataDumpDirectory");
// Because '/' can't be used inside a field trial parameter, we use ';'
// instead.
// This is only relevant to WebRTC-DecoderDataDumpDirectory
// field trial. ';' is chosen arbitrary. Even though it's a legal character
// in some file systems, we can sacrifice ability to use it in the path to
// dumped video, since it's developers-only feature for debugging.
absl::c_replace(decoded_output_file, ';', '/');
if (!decoded_output_file.empty()) {
char filename_buffer[256];
rtc::SimpleStringBuilder ssb(filename_buffer);
ssb << decoded_output_file << "/webrtc_receive_stream_"
<< this->config_.rtp.remote_ssrc << "-" << rtc::TimeMicros()
<< ".ivf";
video_decoder = CreateFrameDumpingDecoderWrapper(
std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
}
video_decoders_.push_back(std::move(video_decoder));
video_receiver_.RegisterExternalDecoder(video_decoders_.back().get(),
decoder.payload_type);
VideoCodec codec = CreateDecoderVideoCodec(decoder);
const bool raw_payload =
config_.rtp.raw_payload_types.count(codec.plType) > 0;
rtp_video_stream_receiver_.AddReceiveCodec(
codec, decoder.video_format.parameters, raw_payload);
RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
&codec, num_cpu_cores_, false));
}
RTC_DCHECK(renderer != nullptr);
video_stream_decoder_.reset(
new VideoStreamDecoder(&video_receiver_, &stats_proxy_, renderer));
// Make sure we register as a stats observer *after* we've prepared the
// |video_stream_decoder_|.
call_stats_->RegisterStatsObserver(this);
// Start decoding on task queue.
video_receiver_.DecoderThreadStarting();
stats_proxy_.DecoderThreadStarting();
decode_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
decoder_stopped_ = false;
StartNextDecode();
});
decoder_running_ = true;
rtp_video_stream_receiver_.StartReceive();
}
void VideoReceiveStream2::Stop() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtp_video_stream_receiver_.StopReceive();
stats_proxy_.OnUniqueFramesCounted(
rtp_video_stream_receiver_.GetUniqueFramesSeen());
decode_queue_.PostTask([this] { frame_buffer_->Stop(); });
call_stats_->DeregisterStatsObserver(this);
if (decoder_running_) {
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;
video_receiver_.DecoderThreadStopped();
stats_proxy_.DecoderThreadStopped();
// Deregister external decoders so they are no longer running during
// destruction. This effectively stops the VCM since the decoder thread is
// stopped, the VCM is deregistered and no asynchronous decoder threads are
// running.
for (const Decoder& decoder : config_.decoders)
video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
UpdateHistograms();
}
video_stream_decoder_.reset();
incoming_video_stream_.reset();
transport_adapter_.Disable();
}
VideoReceiveStream::Stats VideoReceiveStream2::GetStats() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
VideoReceiveStream2::Stats stats = stats_proxy_.GetStats();
stats.total_bitrate_bps = 0;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(stats.ssrc);
if (statistician) {
stats.rtp_stats = statistician->GetStats();
stats.total_bitrate_bps = statistician->BitrateReceived();
}
if (config_.rtp.rtx_ssrc) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(config_.rtp.rtx_ssrc);
if (rtx_statistician)
stats.total_bitrate_bps += rtx_statistician->BitrateReceived();
}
return stats;
}
void VideoReceiveStream2::UpdateHistograms() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
absl::optional<int> fraction_lost;
StreamDataCounters rtp_stats;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(config_.rtp.remote_ssrc);
if (statistician) {
fraction_lost = statistician->GetFractionLostInPercent();
rtp_stats = statistician->GetReceiveStreamDataCounters();
}
if (config_.rtp.rtx_ssrc) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(config_.rtp.rtx_ssrc);
if (rtx_statistician) {
StreamDataCounters rtx_stats =
rtx_statistician->GetReceiveStreamDataCounters();
stats_proxy_.UpdateHistograms(fraction_lost, rtp_stats, &rtx_stats);
return;
}
}
stats_proxy_.UpdateHistograms(fraction_lost, rtp_stats, nullptr);
}
void VideoReceiveStream2::AddSecondarySink(RtpPacketSinkInterface* sink) {
rtp_video_stream_receiver_.AddSecondarySink(sink);
}
void VideoReceiveStream2::RemoveSecondarySink(
const RtpPacketSinkInterface* sink) {
rtp_video_stream_receiver_.RemoveSecondarySink(sink);
}
bool VideoReceiveStream2::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
if (delay_ms < kMinBaseMinimumDelayMs || delay_ms > kMaxBaseMinimumDelayMs) {
return false;
}
// TODO(bugs.webrtc.org/11489): Consider posting to worker.
rtc::CritScope cs(&playout_delay_lock_);
base_minimum_playout_delay_ms_ = delay_ms;
UpdatePlayoutDelays();
return true;
}
int VideoReceiveStream2::GetBaseMinimumPlayoutDelayMs() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtc::CritScope cs(&playout_delay_lock_);
return base_minimum_playout_delay_ms_;
}
void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) {
VideoFrameMetaData frame_meta(video_frame, clock_->CurrentTime());
worker_thread_->PostTask(
ToQueuedTask(task_safety_, [frame_meta, this]() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
int64_t video_playout_ntp_ms;
int64_t sync_offset_ms;
double estimated_freq_khz;
if (rtp_stream_sync_.GetStreamSyncOffsetInMs(
frame_meta.rtp_timestamp, frame_meta.render_time_ms(),
&video_playout_ntp_ms, &sync_offset_ms, &estimated_freq_khz)) {
stats_proxy_.OnSyncOffsetUpdated(video_playout_ntp_ms, sync_offset_ms,
estimated_freq_khz);
}
stats_proxy_.OnRenderedFrame(frame_meta);
}));
source_tracker_.OnFrameDelivered(video_frame.packet_infos());
config_.renderer->OnFrame(video_frame);
}
void VideoReceiveStream2::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
rtp_video_stream_receiver_.SetFrameDecryptor(std::move(frame_decryptor));
}
void VideoReceiveStream2::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
rtp_video_stream_receiver_.SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
}
void VideoReceiveStream2::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) {
RTC_DCHECK(buffering_allowed);
rtp_video_stream_receiver_.RequestPacketRetransmit(sequence_numbers);
}
void VideoReceiveStream2::RequestKeyFrame(int64_t timestamp_ms) {
rtp_video_stream_receiver_.RequestKeyFrame();
last_keyframe_request_ms_ = timestamp_ms;
}
void VideoReceiveStream2::OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&network_sequence_checker_);
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
int64_t time_now_ms = clock_->TimeInMilliseconds();
if (last_complete_frame_time_ms_ > 0 &&
time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {
frame_buffer_->Clear();
}
last_complete_frame_time_ms_ = time_now_ms;
// TODO(bugs.webrtc.org/11489): We grab the playout_delay_lock_ lock
// potentially twice. Consider checking both min/max and posting to worker if
// there's a change. If we always update playout delays on the worker, we
// don't need a lock.
const PlayoutDelay& playout_delay = frame->EncodedImage().playout_delay_;
if (playout_delay.min_ms >= 0) {
rtc::CritScope cs(&playout_delay_lock_);
frame_minimum_playout_delay_ms_ = playout_delay.min_ms;
UpdatePlayoutDelays();
}
if (playout_delay.max_ms >= 0) {
rtc::CritScope cs(&playout_delay_lock_);
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);
}
void VideoReceiveStream2::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
frame_buffer_->UpdateRtt(max_rtt_ms);
rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
stats_proxy_.OnRttUpdate(avg_rtt_ms);
}
uint32_t VideoReceiveStream2::id() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
return config_.rtp.remote_ssrc;
}
absl::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
absl::optional<Syncable::Info> info =
rtp_video_stream_receiver_.GetSyncInfo();
if (!info)
return absl::nullopt;
info->current_delay_ms = timing_->TargetVideoDelay();
return info;
}
bool VideoReceiveStream2::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const {
RTC_NOTREACHED();
return 0;
}
void VideoReceiveStream2::SetEstimatedPlayoutNtpTimestampMs(
int64_t ntp_timestamp_ms,
int64_t time_ms) {
RTC_NOTREACHED();
}
void VideoReceiveStream2::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// TODO(bugs.webrtc.org/11489): See if we can't get rid of the
// |playout_delay_lock_|
rtc::CritScope cs(&playout_delay_lock_);
syncable_minimum_playout_delay_ms_ = delay_ms;
UpdatePlayoutDelays();
}
int64_t VideoReceiveStream2::GetWaitMs() const {
return keyframe_required_ ? max_wait_for_keyframe_ms_
: max_wait_for_frame_ms_;
}
void VideoReceiveStream2::StartNextDecode() {
TRACE_EVENT0("webrtc", "VideoReceiveStream2::StartNextDecode");
frame_buffer_->NextFrame(
GetWaitMs(), keyframe_required_, &decode_queue_,
/* encoded frame handler */
[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([this, frame = std::move(frame)]() mutable {
RTC_DCHECK_RUN_ON(&decode_queue_);
if (decoder_stopped_)
return;
if (frame) {
HandleEncodedFrame(std::move(frame));
} else {
HandleFrameBufferTimeout();
}
StartNextDecode();
});
});
}
void VideoReceiveStream2::HandleEncodedFrame(
std::unique_ptr<EncodedFrame> frame) {
// Running on |decode_queue_|.
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);
HandleKeyFrameGeneration(frame->FrameType() == VideoFrameType::kVideoFrameKey,
now_ms);
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(now_ms);
} 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(now_ms);
}
if (encoded_frame_buffer_function_) {
frame->Retain();
encoded_frame_buffer_function_(WebRtcRecordableEncodedFrame(*frame));
}
}
void VideoReceiveStream2::HandleKeyFrameGeneration(
bool received_frame_is_keyframe,
int64_t now_ms) {
// Repeat sending keyframe requests if we've requested a keyframe.
if (!keyframe_generation_requested_) {
return;
}
if (received_frame_is_keyframe) {
keyframe_generation_requested_ = false;
} else if (last_keyframe_request_ms_ + max_wait_for_keyframe_ms_ <= now_ms) {
if (!IsReceivingKeyFrame(now_ms)) {
RequestKeyFrame(now_ms);
}
} else {
// It hasn't been long enough since the last keyframe request, do nothing.
}
}
void VideoReceiveStream2::HandleFrameBufferTimeout() {
// Running on |decode_queue_|.
int64_t now_ms = clock_->TimeInMilliseconds();
absl::optional<int64_t> last_packet_ms =
rtp_video_stream_receiver_.LastReceivedPacketMs();
// 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) {
worker_thread_->PostTask(ToQueuedTask(task_safety_, [this]() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
stats_proxy_.OnStreamInactive();
}));
}
if (stream_is_active && !IsReceivingKeyFrame(now_ms) &&
(!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(now_ms);
}
}
bool VideoReceiveStream2::IsReceivingKeyFrame(int64_t timestamp_ms) const {
absl::optional<int64_t> last_keyframe_packet_ms =
rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
// 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 &&
timestamp_ms - *last_keyframe_packet_ms < max_wait_for_keyframe_ms_;
return receiving_keyframe;
}
void VideoReceiveStream2::UpdatePlayoutDelays() const {
const int minimum_delay_ms =
std::max({frame_minimum_playout_delay_ms_, base_minimum_playout_delay_ms_,
syncable_minimum_playout_delay_ms_});
if (minimum_delay_ms >= 0) {
timing_->set_min_playout_delay(minimum_delay_ms);
}
const int maximum_delay_ms = frame_maximum_playout_delay_ms_;
if (maximum_delay_ms >= 0) {
timing_->set_max_playout_delay(maximum_delay_ms);
}
}
std::vector<webrtc::RtpSource> VideoReceiveStream2::GetSources() const {
return source_tracker_.GetSources();
}
VideoReceiveStream2::RecordingState
VideoReceiveStream2::SetAndGetRecordingState(RecordingState state,
bool generate_key_frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtc::Event event;
RecordingState old_state;
decode_queue_.PostTask([this, &event, &old_state, generate_key_frame,
state = std::move(state)] {
RTC_DCHECK_RUN_ON(&decode_queue_);
// Save old state.
old_state.callback = std::move(encoded_frame_buffer_function_);
old_state.keyframe_needed = keyframe_generation_requested_;
old_state.last_keyframe_request_ms = last_keyframe_request_ms_;
// Set new state.
encoded_frame_buffer_function_ = std::move(state.callback);
if (generate_key_frame) {
RequestKeyFrame(clock_->TimeInMilliseconds());
keyframe_generation_requested_ = true;
} else {
keyframe_generation_requested_ = state.keyframe_needed;
last_keyframe_request_ms_ = state.last_keyframe_request_ms.value_or(0);
}
event.Set();
});
event.Wait(rtc::Event::kForever);
return old_state;
}
void VideoReceiveStream2::GenerateKeyFrame() {
decode_queue_.PostTask([this]() {
RTC_DCHECK_RUN_ON(&decode_queue_);
RequestKeyFrame(clock_->TimeInMilliseconds());
keyframe_generation_requested_ = true;
});
}
} // namespace internal
} // namespace webrtc
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