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webrtc/video/video_receive_stream2.cc
Per K 217b384c1b Remove rtp header extension from config of Call audio and video receivers 
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.

Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
2023-01-31 11:58:43 +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/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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 "call/rtp_stream_receiver_controller_interface.h"
#include "call/rtx_receive_stream.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/timing.h"
#include "modules/video_coding/utility/vp8_header_parser.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "video/call_stats2.h"
#include "video/frame_dumping_decoder.h"
#include "video/receive_statistics_proxy2.h"
#include "video/render/incoming_video_stream.h"
#include "video/task_queue_frame_decode_scheduler.h"
namespace webrtc {
namespace internal {
namespace {
// The default delay before re-requesting a key frame to be sent.
constexpr TimeDelta kMinBaseMinimumDelay = TimeDelta::Zero();
constexpr TimeDelta kMaxBaseMinimumDelay = TimeDelta::Seconds(10);
// Concrete instance of RecordableEncodedFrame wrapping needed content
// from EncodedFrame.
class WebRtcRecordableEncodedFrame : public RecordableEncodedFrame {
public:
explicit WebRtcRecordableEncodedFrame(
const EncodedFrame& frame,
RecordableEncodedFrame::EncodedResolution resolution)
: buffer_(frame.GetEncodedData()),
render_time_ms_(frame.RenderTime()),
codec_(frame.CodecSpecific()->codecType),
is_key_frame_(frame.FrameType() == VideoFrameType::kVideoFrameKey),
resolution_(resolution) {
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_;
};
RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) {
FieldTrialOptional<int> width("w");
FieldTrialOptional<int> height("h");
ParseFieldTrial({&width, &height},
field_trials.Lookup("WebRTC-Video-InitialDecoderResolution"));
if (width && height) {
return RenderResolution(width.Value(), height.Value());
}
return RenderResolution(320, 180);
}
// 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:
bool Configure(const Settings& settings) override {
RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
return true;
}
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"; }
};
bool IsKeyFrameAndUnspecifiedResolution(const EncodedFrame& frame) {
return frame.FrameType() == VideoFrameType::kVideoFrameKey &&
frame.EncodedImage()._encodedWidth == 0 &&
frame.EncodedImage()._encodedHeight == 0;
}
std::string OptionalDelayToLogString(const absl::optional<TimeDelta> opt) {
return opt.has_value() ? ToLogString(*opt) : "<unset>";
}
} // namespace
TimeDelta DetermineMaxWaitForFrame(TimeDelta rtp_history, bool is_keyframe) {
// A (arbitrary) conversion factor between the remotely signalled NACK buffer
// time (if not present defaults to 1000ms) and the maximum time we wait for a
// remote frame. Chosen to not change existing defaults when using not
// rtx-time.
const int conversion_factor = 3;
if (rtp_history > TimeDelta::Zero() &&
conversion_factor * rtp_history < kMaxWaitForFrame) {
return is_keyframe ? rtp_history : conversion_factor * rtp_history;
}
return is_keyframe ? kMaxWaitForKeyFrame : kMaxWaitForFrame;
}
VideoReceiveStream2::VideoReceiveStream2(
TaskQueueFactory* task_queue_factory,
Call* call,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStreamInterface::Config config,
CallStats* call_stats,
Clock* clock,
std::unique_ptr<VCMTiming> timing,
NackPeriodicProcessor* nack_periodic_processor,
DecodeSynchronizer* decode_sync,
RtcEventLog* event_log)
: task_queue_factory_(task_queue_factory),
transport_adapter_(config.rtcp_send_transport),
config_(std::move(config)),
num_cpu_cores_(num_cpu_cores),
call_(call),
clock_(clock),
call_stats_(call_stats),
source_tracker_(clock_),
stats_proxy_(remote_ssrc(), clock_, call->worker_thread()),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
timing_(std::move(timing)),
video_receiver_(clock_, timing_.get(), call->trials()),
rtp_video_stream_receiver_(call->worker_thread(),
clock_,
&transport_adapter_,
call_stats->AsRtcpRttStats(),
packet_router,
&config_,
rtp_receive_statistics_.get(),
&stats_proxy_,
&stats_proxy_,
nack_periodic_processor,
this, // OnCompleteFrameCallback
std::move(config_.frame_decryptor),
std::move(config_.frame_transformer),
call->trials(),
event_log),
rtp_stream_sync_(call->worker_thread(), this),
max_wait_for_keyframe_(DetermineMaxWaitForFrame(
TimeDelta::Millis(config_.rtp.nack.rtp_history_ms),
true)),
max_wait_for_frame_(DetermineMaxWaitForFrame(
TimeDelta::Millis(config_.rtp.nack.rtp_history_ms),
false)),
decode_queue_(task_queue_factory_->CreateTaskQueue(
"DecodingQueue",
TaskQueueFactory::Priority::HIGH)) {
RTC_LOG(LS_INFO) << "VideoReceiveStream2: " << config_.ToString();
RTC_DCHECK(call_->worker_thread());
RTC_DCHECK(config_.renderer);
RTC_DCHECK(call_stats_);
packet_sequence_checker_.Detach();
RTC_DCHECK(!config_.decoders.empty());
RTC_CHECK(config_.decoder_factory);
std::set<int> decoder_payload_types;
for (const Decoder& decoder : config_.decoders) {
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(TimeDelta::Millis(config_.render_delay_ms));
std::unique_ptr<FrameDecodeScheduler> scheduler =
decode_sync ? decode_sync->CreateSynchronizedFrameScheduler()
: std::make_unique<TaskQueueFrameDecodeScheduler>(
clock, call_->worker_thread());
buffer_ = std::make_unique<VideoStreamBufferController>(
clock_, call_->worker_thread(), timing_.get(), &stats_proxy_, this,
max_wait_for_keyframe_, max_wait_for_frame_, std::move(scheduler),
call_->trials());
if (rtx_ssrc()) {
rtx_receive_stream_ = std::make_unique<RtxReceiveStream>(
&rtp_video_stream_receiver_,
std::move(config_.rtp.rtx_associated_payload_types), remote_ssrc(),
rtp_receive_statistics_.get());
} else {
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc(), true);
}
}
VideoReceiveStream2::~VideoReceiveStream2() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_LOG(LS_INFO) << "~VideoReceiveStream2: " << config_.ToString();
RTC_DCHECK(!media_receiver_);
RTC_DCHECK(!rtx_receiver_);
Stop();
}
void VideoReceiveStream2::RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(!media_receiver_);
RTC_DCHECK(!rtx_receiver_);
// Register with RtpStreamReceiverController.
media_receiver_ = receiver_controller->CreateReceiver(
remote_ssrc(), &rtp_video_stream_receiver_);
if (rtx_ssrc()) {
RTC_DCHECK(rtx_receive_stream_);
rtx_receiver_ = receiver_controller->CreateReceiver(
rtx_ssrc(), rtx_receive_stream_.get());
}
}
void VideoReceiveStream2::UnregisterFromTransport() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
media_receiver_.reset();
rtx_receiver_.reset();
}
const std::string& VideoReceiveStream2::sync_group() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return config_.sync_group;
}
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) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return rtp_video_stream_receiver_.DeliverRtcp(packet, length);
}
void VideoReceiveStream2::SetSync(Syncable* audio_syncable) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_stream_sync_.ConfigureSync(audio_syncable);
}
void VideoReceiveStream2::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (config_.rtp.local_ssrc == local_ssrc)
return;
// TODO(tommi): Make sure we don't rely on local_ssrc via the config struct.
const_cast<uint32_t&>(config_.rtp.local_ssrc) = local_ssrc;
rtp_video_stream_receiver_.OnLocalSsrcChange(local_ssrc);
}
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_.ulpfec_payload_type() != -1;
if (config_.rtp.nack.rtp_history_ms > 0 && protected_by_fec) {
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) {
VideoDecoder::Settings settings;
settings.set_codec_type(
PayloadStringToCodecType(decoder.video_format.name));
settings.set_max_render_resolution(
InitialDecoderResolution(call_->trials()));
settings.set_number_of_cores(num_cpu_cores_);
const bool raw_payload =
config_.rtp.raw_payload_types.count(decoder.payload_type) > 0;
{
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.AddReceiveCodec(
decoder.payload_type, settings.codec_type(),
decoder.video_format.parameters, raw_payload);
}
video_receiver_.RegisterReceiveCodec(decoder.payload_type, settings);
}
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.
stats_proxy_.DecoderThreadStarting();
decode_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
decoder_stopped_ = false;
});
buffer_->StartNextDecode(true);
decoder_running_ = true;
{
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.StartReceive();
}
}
void VideoReceiveStream2::Stop() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
// Also call `GetUniqueFramesSeen()` at the same time (since it's a counter
// that's updated on the network thread).
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.StopReceive();
stats_proxy_.OnUniqueFramesCounted(
rtp_video_stream_receiver_.GetUniqueFramesSeen());
buffer_->Stop();
call_stats_->DeregisterStatsObserver(this);
if (decoder_running_) {
rtc::Event done;
decode_queue_.PostTask([this, &done] {
RTC_DCHECK_RUN_ON(&decode_queue_);
// Set `decoder_stopped_` before deregistering all decoders. This means
// that any pending encoded frame will return early without trying to
// access the decoder database.
decoder_stopped_ = true;
for (const Decoder& decoder : config_.decoders) {
video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
}
done.Set();
});
done.Wait(rtc::Event::kForever);
decoder_running_ = false;
stats_proxy_.DecoderThreadStopped();
UpdateHistograms();
}
// TODO(bugs.webrtc.org/11993): Make these calls on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.RemoveReceiveCodecs();
video_receiver_.DeregisterReceiveCodecs();
video_stream_decoder_.reset();
incoming_video_stream_.reset();
transport_adapter_.Disable();
}
void VideoReceiveStream2::SetRtcpMode(RtcpMode mode) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<RtcpMode&>(config_.rtp.rtcp_mode) = mode;
rtp_video_stream_receiver_.SetRtcpMode(mode);
}
void VideoReceiveStream2::SetFlexFecProtection(
RtpPacketSinkInterface* flexfec_sink) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.SetPacketSink(flexfec_sink);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<RtpPacketSinkInterface*&>(config_.rtp.packet_sink_) = flexfec_sink;
const_cast<bool&>(config_.rtp.protected_by_flexfec) =
(flexfec_sink != nullptr);
}
void VideoReceiveStream2::SetLossNotificationEnabled(bool enabled) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<bool&>(config_.rtp.lntf.enabled) = enabled;
rtp_video_stream_receiver_.SetLossNotificationEnabled(enabled);
}
void VideoReceiveStream2::SetNackHistory(TimeDelta history) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_GE(history.ms(), 0);
if (config_.rtp.nack.rtp_history_ms == history.ms())
return;
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<int&>(config_.rtp.nack.rtp_history_ms) = history.ms();
const bool protected_by_fec =
config_.rtp.protected_by_flexfec ||
rtp_video_stream_receiver_.ulpfec_payload_type() != -1;
buffer_->SetProtectionMode(history.ms() > 0 && protected_by_fec
? kProtectionNackFEC
: kProtectionNack);
rtp_video_stream_receiver_.SetNackHistory(history);
TimeDelta max_wait_for_keyframe = DetermineMaxWaitForFrame(history, true);
TimeDelta max_wait_for_frame = DetermineMaxWaitForFrame(history, false);
max_wait_for_keyframe_ = max_wait_for_keyframe;
max_wait_for_frame_ = max_wait_for_frame;
buffer_->SetMaxWaits(max_wait_for_keyframe, max_wait_for_frame);
}
void VideoReceiveStream2::SetProtectionPayloadTypes(int red_payload_type,
int ulpfec_payload_type) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.SetProtectionPayloadTypes(red_payload_type,
ulpfec_payload_type);
}
void VideoReceiveStream2::SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.SetReferenceTimeReport(
rtcp_xr.receiver_reference_time_report);
}
void VideoReceiveStream2::SetAssociatedPayloadTypes(
std::map<int, int> associated_payload_types) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// For setting the associated payload types after construction, we currently
// assume that the rtx_ssrc cannot change. In such a case we can know that
// if the ssrc is non-0, a `rtx_receive_stream_` instance has previously been
// created and configured (and is referenced by `rtx_receiver_`) and we can
// simply reconfigure it.
// If rtx_ssrc is 0 however, we ignore this call.
if (!rtx_ssrc())
return;
rtx_receive_stream_->SetAssociatedPayloadTypes(
std::move(associated_payload_types));
}
void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
const Decoder& decoder) {
TRACE_EVENT0("webrtc",
"VideoReceiveStream2::CreateAndRegisterExternalDecoder");
std::unique_ptr<VideoDecoder> video_decoder =
config_.decoder_factory->CreateVideoDecoder(decoder.video_format);
// 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 =
call_->trials().Lookup("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_" << remote_ssrc()
<< "-" << rtc::TimeMicros() << ".ivf";
video_decoder = CreateFrameDumpingDecoderWrapper(
std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
}
video_receiver_.RegisterExternalDecoder(std::move(video_decoder),
decoder.payload_type);
}
VideoReceiveStreamInterface::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 (rtx_ssrc()) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(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(remote_ssrc());
if (statistician) {
fraction_lost = statistician->GetFractionLostInPercent();
rtp_stats = statistician->GetReceiveStreamDataCounters();
}
if (rtx_ssrc()) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(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);
}
bool VideoReceiveStream2::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
TimeDelta delay = TimeDelta::Millis(delay_ms);
if (delay < kMinBaseMinimumDelay || delay > kMaxBaseMinimumDelay) {
return false;
}
base_minimum_playout_delay_ = delay;
UpdatePlayoutDelays();
return true;
}
int VideoReceiveStream2::GetBaseMinimumPlayoutDelayMs() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
constexpr TimeDelta kDefaultBaseMinPlayoutDelay = TimeDelta::Millis(-1);
// Unset must be -1.
static_assert(-1 == kDefaultBaseMinPlayoutDelay.ms(), "");
return base_minimum_playout_delay_.value_or(kDefaultBaseMinPlayoutDelay).ms();
}
void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) {
source_tracker_.OnFrameDelivered(video_frame.packet_infos());
config_.renderer->OnFrame(video_frame);
// TODO(bugs.webrtc.org/10739): we should set local capture clock offset for
// `video_frame.packet_infos`. But VideoFrame is const qualified here.
// For frame delay metrics, calculated in `OnRenderedFrame`, to better reflect
// user experience measurements must be done as close as possible to frame
// rendering moment. Capture current time, which is used for calculation of
// delay metrics in `OnRenderedFrame`, right after frame is passed to
// renderer. Frame may or may be not rendered by this time. This results in
// inaccuracy but is still the best we can do in the absence of "frame
// rendered" callback from the renderer.
VideoFrameMetaData frame_meta(video_frame, clock_->CurrentTime());
call_->worker_thread()->PostTask(
SafeTask(task_safety_.flag(), [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);
}));
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (pending_resolution_.has_value()) {
if (!pending_resolution_->empty() &&
(video_frame.width() != static_cast<int>(pending_resolution_->width) ||
video_frame.height() !=
static_cast<int>(pending_resolution_->height))) {
RTC_LOG(LS_WARNING)
<< "Recordable encoded frame stream resolution was reported as "
<< pending_resolution_->width << "x" << pending_resolution_->height
<< " but the stream is now " << video_frame.width()
<< video_frame.height();
}
pending_resolution_ = RecordableEncodedFrame::EncodedResolution{
static_cast<unsigned>(video_frame.width()),
static_cast<unsigned>(video_frame.height())};
}
}
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::RequestKeyFrame(Timestamp now) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// Called from RtpVideoStreamReceiver (rtp_video_stream_receiver_ is
// ultimately responsible).
rtp_video_stream_receiver_.RequestKeyFrame();
last_keyframe_request_ = now;
}
void VideoReceiveStream2::OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
const VideoPlayoutDelay& playout_delay = frame->EncodedImage().playout_delay_;
if (playout_delay.min_ms >= 0) {
frame_minimum_playout_delay_ = TimeDelta::Millis(playout_delay.min_ms);
UpdatePlayoutDelays();
}
if (playout_delay.max_ms >= 0) {
frame_maximum_playout_delay_ = TimeDelta::Millis(playout_delay.max_ms);
UpdatePlayoutDelays();
}
auto last_continuous_pid = buffer_->InsertFrame(std::move(frame));
if (last_continuous_pid.has_value()) {
{
// TODO(bugs.webrtc.org/11993): Call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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_);
// TODO(bugs.webrtc.org/13757): Replace with TimeDelta.
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 remote_ssrc();
}
absl::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
absl::optional<Syncable::Info> info =
rtp_video_stream_receiver_.GetSyncInfo();
if (!info)
return absl::nullopt;
info->current_delay_ms = timing_->TargetVideoDelay().ms();
return info;
}
bool VideoReceiveStream2::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const {
RTC_DCHECK_NOTREACHED();
return false;
}
void VideoReceiveStream2::SetEstimatedPlayoutNtpTimestampMs(
int64_t ntp_timestamp_ms,
int64_t time_ms) {
RTC_DCHECK_NOTREACHED();
}
bool VideoReceiveStream2::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
syncable_minimum_playout_delay_ = TimeDelta::Millis(delay_ms);
UpdatePlayoutDelays();
return true;
}
void VideoReceiveStream2::OnEncodedFrame(std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
Timestamp now = clock_->CurrentTime();
const bool keyframe_request_is_due =
!last_keyframe_request_ ||
now >= (*last_keyframe_request_ + max_wait_for_keyframe_);
const bool received_frame_is_keyframe =
frame->FrameType() == VideoFrameType::kVideoFrameKey;
// 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);
decode_queue_.PostTask([this, now, keyframe_request_is_due,
received_frame_is_keyframe, frame = std::move(frame),
keyframe_required = keyframe_required_]() mutable {
RTC_DCHECK_RUN_ON(&decode_queue_);
if (decoder_stopped_)
return;
DecodeFrameResult result = HandleEncodedFrameOnDecodeQueue(
std::move(frame), keyframe_request_is_due, keyframe_required);
// TODO(bugs.webrtc.org/11993): Make this PostTask to the network thread.
call_->worker_thread()->PostTask(
SafeTask(task_safety_.flag(),
[this, now, result = std::move(result),
received_frame_is_keyframe, keyframe_request_is_due]() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
keyframe_required_ = result.keyframe_required;
if (result.decoded_frame_picture_id) {
rtp_video_stream_receiver_.FrameDecoded(
*result.decoded_frame_picture_id);
}
HandleKeyFrameGeneration(received_frame_is_keyframe, now,
result.force_request_key_frame,
keyframe_request_is_due);
buffer_->StartNextDecode(keyframe_required_);
}));
});
}
void VideoReceiveStream2::OnDecodableFrameTimeout(TimeDelta wait) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
Timestamp now = clock_->CurrentTime();
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.
constexpr TimeDelta kInactiveDuration = TimeDelta::Seconds(5);
const bool stream_is_active =
last_packet_ms &&
now - Timestamp::Millis(*last_packet_ms) < kInactiveDuration;
if (!stream_is_active)
stats_proxy_.OnStreamInactive();
if (stream_is_active && !IsReceivingKeyFrame(now) &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
RTC_LOG(LS_WARNING) << "No decodable frame in " << wait
<< ", requesting keyframe.";
RequestKeyFrame(now);
}
buffer_->StartNextDecode(keyframe_required_);
}
VideoReceiveStream2::DecodeFrameResult
VideoReceiveStream2::HandleEncodedFrameOnDecodeQueue(
std::unique_ptr<EncodedFrame> frame,
bool keyframe_request_is_due,
bool keyframe_required) {
RTC_DCHECK_RUN_ON(&decode_queue_);
bool force_request_key_frame = false;
absl::optional<int64_t> decoded_frame_picture_id;
if (!video_receiver_.IsExternalDecoderRegistered(frame->PayloadType())) {
// Look for the decoder with this payload type.
for (const Decoder& decoder : config_.decoders) {
if (decoder.payload_type == frame->PayloadType()) {
CreateAndRegisterExternalDecoder(decoder);
break;
}
}
}
int64_t frame_id = frame->Id();
int decode_result = DecodeAndMaybeDispatchEncodedFrame(std::move(frame));
if (decode_result == WEBRTC_VIDEO_CODEC_OK ||
decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
keyframe_required = false;
frame_decoded_ = true;
decoded_frame_picture_id = frame_id;
if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
force_request_key_frame = true;
} else if (!frame_decoded_ || !keyframe_required || keyframe_request_is_due) {
keyframe_required = true;
// TODO(philipel): Remove this keyframe request when downstream project
// has been fixed.
force_request_key_frame = true;
}
return DecodeFrameResult{
.force_request_key_frame = force_request_key_frame,
.decoded_frame_picture_id = std::move(decoded_frame_picture_id),
.keyframe_required = keyframe_required,
};
}
int VideoReceiveStream2::DecodeAndMaybeDispatchEncodedFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&decode_queue_);
// If `buffered_encoded_frames_` grows out of control (=60 queued frames),
// maybe due to a stuck decoder, we just halt the process here and log the
// error.
const bool encoded_frame_output_enabled =
encoded_frame_buffer_function_ != nullptr &&
buffered_encoded_frames_.size() < kBufferedEncodedFramesMaxSize;
EncodedFrame* frame_ptr = frame.get();
if (encoded_frame_output_enabled) {
// If we receive a key frame with unset resolution, hold on dispatching the
// frame and following ones until we know a resolution of the stream.
// NOTE: The code below has a race where it can report the wrong
// resolution for keyframes after an initial keyframe of other resolution.
// However, the only known consumer of this information is the W3C
// MediaRecorder and it will only use the resolution in the first encoded
// keyframe from WebRTC, so misreporting is fine.
buffered_encoded_frames_.push_back(std::move(frame));
if (buffered_encoded_frames_.size() == kBufferedEncodedFramesMaxSize)
RTC_LOG(LS_ERROR) << "About to halt recordable encoded frame output due "
"to too many buffered frames.";
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (IsKeyFrameAndUnspecifiedResolution(*frame_ptr) &&
!pending_resolution_.has_value())
pending_resolution_.emplace();
}
int decode_result = video_receiver_.Decode(frame_ptr);
if (encoded_frame_output_enabled) {
absl::optional<RecordableEncodedFrame::EncodedResolution>
pending_resolution;
{
// Fish out `pending_resolution_` to avoid taking the mutex on every lap
// or dispatching under the mutex in the flush loop.
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (pending_resolution_.has_value())
pending_resolution = *pending_resolution_;
}
if (!pending_resolution.has_value() || !pending_resolution->empty()) {
// Flush the buffered frames.
for (const auto& frame : buffered_encoded_frames_) {
RecordableEncodedFrame::EncodedResolution resolution{
frame->EncodedImage()._encodedWidth,
frame->EncodedImage()._encodedHeight};
if (IsKeyFrameAndUnspecifiedResolution(*frame)) {
RTC_DCHECK(!pending_resolution->empty());
resolution = *pending_resolution;
}
encoded_frame_buffer_function_(
WebRtcRecordableEncodedFrame(*frame, resolution));
}
buffered_encoded_frames_.clear();
}
}
return decode_result;
}
void VideoReceiveStream2::HandleKeyFrameGeneration(
bool received_frame_is_keyframe,
Timestamp now,
bool always_request_key_frame,
bool keyframe_request_is_due) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
bool request_key_frame = always_request_key_frame;
// Repeat sending keyframe requests if we've requested a keyframe.
if (keyframe_generation_requested_) {
if (received_frame_is_keyframe) {
keyframe_generation_requested_ = false;
} else if (keyframe_request_is_due) {
if (!IsReceivingKeyFrame(now)) {
request_key_frame = true;
}
} else {
// It hasn't been long enough since the last keyframe request, do nothing.
}
}
if (request_key_frame) {
// HandleKeyFrameGeneration is initiated from the decode thread -
// RequestKeyFrame() triggers a call back to the decode thread.
// Perhaps there's a way to avoid that.
RequestKeyFrame(now);
}
}
bool VideoReceiveStream2::IsReceivingKeyFrame(Timestamp now) const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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 &&
now - Timestamp::Millis(*last_keyframe_packet_ms) <
max_wait_for_keyframe_;
return receiving_keyframe;
}
void VideoReceiveStream2::UpdatePlayoutDelays() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
const std::initializer_list<absl::optional<TimeDelta>> min_delays = {
frame_minimum_playout_delay_, base_minimum_playout_delay_,
syncable_minimum_playout_delay_};
// Since nullopt < anything, this will return the largest of the minumum
// delays, or nullopt if all are nullopt.
absl::optional<TimeDelta> minimum_delay = std::max(min_delays);
if (minimum_delay) {
auto num_playout_delays_set =
absl::c_count_if(min_delays, [](auto opt) { return opt.has_value(); });
if (num_playout_delays_set > 1 &&
timing_->min_playout_delay() != minimum_delay) {
RTC_LOG(LS_WARNING)
<< "Multiple playout delays set. Actual delay value set to "
<< *minimum_delay << " frame min delay="
<< OptionalDelayToLogString(frame_maximum_playout_delay_)
<< " base min delay="
<< OptionalDelayToLogString(base_minimum_playout_delay_)
<< " sync min delay="
<< OptionalDelayToLogString(syncable_minimum_playout_delay_);
}
timing_->set_min_playout_delay(*minimum_delay);
if (frame_minimum_playout_delay_ == TimeDelta::Zero() &&
frame_maximum_playout_delay_ > TimeDelta::Zero()) {
// TODO(kron): Estimate frame rate from video stream.
constexpr Frequency kFrameRate = Frequency::Hertz(60);
// Convert playout delay in ms to number of frames.
int max_composition_delay_in_frames =
std::lrint(*frame_maximum_playout_delay_ * kFrameRate);
// Subtract frames in buffer.
max_composition_delay_in_frames =
std::max(max_composition_delay_in_frames - buffer_->Size(), 0);
timing_->SetMaxCompositionDelayInFrames(max_composition_delay_in_frames);
}
}
if (frame_maximum_playout_delay_) {
timing_->set_max_playout_delay(*frame_maximum_playout_delay_);
}
}
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;
// Save old state, set the new state.
RecordingState old_state;
absl::optional<Timestamp> last_keyframe_request;
{
// TODO(bugs.webrtc.org/11993): Post this to the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
last_keyframe_request = last_keyframe_request_;
last_keyframe_request_ =
generate_key_frame
? clock_->CurrentTime()
: Timestamp::Millis(state.last_keyframe_request_ms.value_or(0));
}
decode_queue_.PostTask(
[this, &event, &old_state, callback = std::move(state.callback),
last_keyframe_request = std::move(last_keyframe_request)] {
RTC_DCHECK_RUN_ON(&decode_queue_);
old_state.callback = std::move(encoded_frame_buffer_function_);
encoded_frame_buffer_function_ = std::move(callback);
old_state.last_keyframe_request_ms =
last_keyframe_request.value_or(Timestamp::Zero()).ms();
event.Set();
});
if (generate_key_frame) {
rtp_video_stream_receiver_.RequestKeyFrame();
{
// TODO(bugs.webrtc.org/11993): Post this to the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
keyframe_generation_requested_ = true;
}
}
event.Wait(rtc::Event::kForever);
return old_state;
}
void VideoReceiveStream2::GenerateKeyFrame() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RequestKeyFrame(clock_->CurrentTime());
keyframe_generation_requested_ = true;
}
} // namespace internal
} // namespace webrtc
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