blob: efdf39bf2443385191067bbed22edcac6ee6b880 [file] [log] [blame]
// Copyright 2015 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "remoting/protocol/webrtc_video_stream.h"
#include <memory>
#include <utility>
#include "base/bind.h"
#include "base/logging.h"
#include "build/build_config.h"
#include "remoting/base/constants.h"
#include "remoting/codec/webrtc_video_encoder_proxy.h"
#include "remoting/codec/webrtc_video_encoder_vpx.h"
#include "remoting/protocol/frame_stats.h"
#include "remoting/protocol/host_video_stats_dispatcher.h"
#include "remoting/protocol/webrtc_frame_scheduler_simple.h"
#include "remoting/protocol/webrtc_transport.h"
#include "third_party/webrtc/api/media_stream_interface.h"
#include "third_party/webrtc/api/notifier.h"
#include "third_party/webrtc/api/peer_connection_interface.h"
#include "third_party/webrtc/media/base/vp9_profile.h"
#if defined(USE_H264_ENCODER)
#include "remoting/codec/webrtc_video_encoder_gpu.h"
#endif
namespace remoting {
namespace protocol {
namespace {
const char kStreamLabel[] = "screen_stream";
const char kVideoLabel[] = "screen_video";
std::string EncodeResultToString(WebrtcVideoEncoder::EncodeResult result) {
using EncodeResult = WebrtcVideoEncoder::EncodeResult;
switch (result) {
case EncodeResult::SUCCEEDED:
return "Succeeded";
case EncodeResult::FRAME_SIZE_EXCEEDS_CAPABILITY:
return "Frame size exceeds capability";
case EncodeResult::UNKNOWN_ERROR:
return "Unknown error";
}
NOTREACHED();
return "";
}
class DummyVideoTrackSource
: public webrtc::Notifier<webrtc::VideoTrackSourceInterface> {
public:
SourceState state() const override { return kLive; }
bool remote() const override { return false; }
bool is_screencast() const override { return true; }
absl::optional<bool> needs_denoising() const override {
return absl::nullopt;
}
bool GetStats(Stats* stats) override { return false; }
void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override {}
void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {}
bool SupportsEncodedOutput() const override { return false; }
void GenerateKeyFrame() override {}
void AddEncodedSink(
rtc::VideoSinkInterface<webrtc::RecordableEncodedFrame>* sink) override {}
void RemoveEncodedSink(
rtc::VideoSinkInterface<webrtc::RecordableEncodedFrame>* sink) override {}
};
} // namespace
struct WebrtcVideoStream::FrameStats {
// The following fields are non-null only for one frame after each incoming
// input event.
InputEventTimestamps input_event_timestamps;
base::TimeTicks capture_started_time;
base::TimeTicks capture_ended_time;
base::TimeDelta capture_delay;
base::TimeTicks encode_started_time;
base::TimeTicks encode_ended_time;
uint32_t capturer_id = 0;
int frame_quality = -1;
};
WebrtcVideoStream::WebrtcVideoStream(const SessionOptions& session_options)
: video_stats_dispatcher_(kStreamLabel), session_options_(session_options) {
// WeakPtr can't be used here, as the Create...() methods return a value
// which would be undefined if the pointer were invalidated. But
// Unretained(this) is safe because |encoder_selector_| is owned by this
// object.
encoder_selector_.RegisterEncoder(
base::BindRepeating(&WebrtcVideoEncoderVpx::IsSupportedByVP8),
base::BindRepeating(&WebrtcVideoStream::CreateVP8Encoder,
base::Unretained(this)));
encoder_selector_.RegisterEncoder(
base::BindRepeating(&WebrtcVideoEncoderVpx::IsSupportedByVP9),
base::BindRepeating(&WebrtcVideoStream::CreateVP9Encoder,
base::Unretained(this),
/*lossless_color=*/false));
encoder_selector_.RegisterEncoder(
base::BindRepeating(&WebrtcVideoEncoderVpx::IsSupportedByVP9),
base::BindRepeating(&WebrtcVideoStream::CreateVP9Encoder,
base::Unretained(this),
/*lossless_color=*/true));
#if defined(USE_H264_ENCODER)
encoder_selector_.RegisterEncoder(
base::BindRepeating(&WebrtcVideoEncoderGpu::IsSupportedByH264),
base::BindRepeating(&WebrtcVideoEncoderGpu::CreateForH264));
#endif
}
WebrtcVideoStream::~WebrtcVideoStream() {
DCHECK(thread_checker_.CalledOnValidThread());
}
void WebrtcVideoStream::Start(
std::unique_ptr<webrtc::DesktopCapturer> desktop_capturer,
WebrtcTransport* webrtc_transport,
scoped_refptr<base::SequencedTaskRunner> encode_task_runner) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(webrtc_transport);
DCHECK(desktop_capturer);
DCHECK(encode_task_runner);
scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_connection_factory(
webrtc_transport->peer_connection_factory());
peer_connection_ = webrtc_transport->peer_connection();
DCHECK(peer_connection_factory);
DCHECK(peer_connection_);
encode_task_runner_ = std::move(encode_task_runner);
capturer_ = std::move(desktop_capturer);
webrtc_transport_ = webrtc_transport;
webrtc_transport_->video_encoder_factory()->RegisterEncoderSelectedCallback(
base::BindRepeating(&WebrtcVideoStream::OnEncoderCreated,
weak_factory_.GetWeakPtr()));
capturer_->Start(this);
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> src =
new rtc::RefCountedObject<DummyVideoTrackSource>();
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
peer_connection_factory->CreateVideoTrack(kVideoLabel, src);
webrtc::RtpTransceiverInit init;
init.stream_ids = {kStreamLabel};
// value() DCHECKs if AddTransceiver() fails, which only happens if a track
// was already added with the stream label.
auto transceiver =
peer_connection_->AddTransceiver(video_track, init).value();
webrtc_transport_->OnVideoTransceiverCreated(transceiver);
scheduler_.reset(new WebrtcFrameSchedulerSimple(session_options_));
scheduler_->Start(webrtc_transport_->video_encoder_factory(),
base::BindRepeating(&WebrtcVideoStream::CaptureNextFrame,
base::Unretained(this)));
video_stats_dispatcher_.Init(webrtc_transport_->CreateOutgoingChannel(
video_stats_dispatcher_.channel_name()),
this);
}
void WebrtcVideoStream::SelectSource(int id) {
capturer_->SelectSource(id);
}
void WebrtcVideoStream::SetEventTimestampsSource(
scoped_refptr<InputEventTimestampsSource> event_timestamps_source) {
event_timestamps_source_ = event_timestamps_source;
}
void WebrtcVideoStream::Pause(bool pause) {
DCHECK(thread_checker_.CalledOnValidThread());
scheduler_->Pause(pause);
}
void WebrtcVideoStream::SetLosslessEncode(bool want_lossless) {
lossless_encode_ = want_lossless;
if (encoder_) {
encoder_->SetLosslessEncode(want_lossless);
}
}
void WebrtcVideoStream::SetLosslessColor(bool want_lossless) {
NOTIMPLEMENTED() << "Changing lossless-color for VP9 requires SDP "
"offer/answer exchange.";
}
void WebrtcVideoStream::SetObserver(Observer* observer) {
DCHECK(thread_checker_.CalledOnValidThread());
observer_ = observer;
}
void WebrtcVideoStream::OnCaptureResult(
webrtc::DesktopCapturer::Result result,
std::unique_ptr<webrtc::DesktopFrame> frame) {
DCHECK(thread_checker_.CalledOnValidThread());
current_frame_stats_->capture_ended_time = base::TimeTicks::Now();
current_frame_stats_->capture_delay =
base::TimeDelta::FromMilliseconds(frame ? frame->capture_time_ms() : 0);
if (!frame) {
scheduler_->OnFrameCaptured(nullptr, nullptr);
return;
}
// TODO(sergeyu): Handle ERROR_PERMANENT result here.
webrtc::DesktopVector dpi =
frame->dpi().is_zero() ? webrtc::DesktopVector(kDefaultDpi, kDefaultDpi)
: frame->dpi();
if (!frame_size_.equals(frame->size()) || !frame_dpi_.equals(dpi)) {
frame_size_ = frame->size();
frame_dpi_ = dpi;
if (observer_)
observer_->OnVideoSizeChanged(this, frame_size_, frame_dpi_);
}
current_frame_stats_->capturer_id = frame->capturer_id();
if (recreate_encoder_) {
recreate_encoder_ = false;
encoder_.reset();
}
if (!encoder_) {
encoder_selector_.SetDesktopFrame(*frame);
encoder_ = encoder_selector_.CreateEncoder();
encoder_->SetLosslessEncode(lossless_encode_);
// TODO(zijiehe): Permanently stop the video stream if we cannot create an
// encoder for the |frame|.
}
WebrtcVideoEncoder::FrameParams frame_params;
if (!scheduler_->OnFrameCaptured(frame.get(), &frame_params)) {
return;
}
if (encoder_) {
current_frame_stats_->encode_started_time = base::TimeTicks::Now();
encoder_->Encode(std::move(frame), frame_params,
base::BindOnce(&WebrtcVideoStream::OnFrameEncoded,
base::Unretained(this)));
}
}
void WebrtcVideoStream::OnChannelInitialized(
ChannelDispatcherBase* channel_dispatcher) {
DCHECK(&video_stats_dispatcher_ == channel_dispatcher);
}
void WebrtcVideoStream::OnChannelClosed(
ChannelDispatcherBase* channel_dispatcher) {
DCHECK(&video_stats_dispatcher_ == channel_dispatcher);
LOG(WARNING) << "video_stats channel was closed.";
}
void WebrtcVideoStream::CaptureNextFrame() {
DCHECK(thread_checker_.CalledOnValidThread());
current_frame_stats_.reset(new FrameStats());
current_frame_stats_->capture_started_time = base::TimeTicks::Now();
current_frame_stats_->input_event_timestamps =
event_timestamps_source_->TakeLastEventTimestamps();
capturer_->CaptureFrame();
}
void WebrtcVideoStream::OnFrameEncoded(
WebrtcVideoEncoder::EncodeResult encode_result,
std::unique_ptr<WebrtcVideoEncoder::EncodedFrame> frame) {
DCHECK(thread_checker_.CalledOnValidThread());
current_frame_stats_->encode_ended_time = base::TimeTicks::Now();
// Convert the frame quantizer to a measure of frame quality between 0 and
// 100, for a simple visualization of quality over time. The quantizer from
// VP8/VP9 encoder lies within 0-63, with 0 representing a lossless
// frame.
// TODO(crbug.com/891571): Remove |quantizer| from the WebrtcVideoEncoder
// interface, and move this logic to the encoders.
if (frame) {
current_frame_stats_->frame_quality = (63 - frame->quantizer) * 100 / 63;
}
HostFrameStats stats;
scheduler_->OnFrameEncoded(frame.get(), &stats);
if (encode_result != WebrtcVideoEncoder::EncodeResult::SUCCEEDED) {
LOG(ERROR) << "Video encoder returns error "
<< EncodeResultToString(encode_result);
// TODO(zijiehe): Restart the video stream.
encoder_.reset();
return;
}
if (!frame) {
return;
}
if (recreate_encoder_) {
// Don't send the encoded frame if the new SDP-negotiated encoder might be
// different from the current one. This would trigger a crash in WebRTC.
return;
}
webrtc::EncodedImageCallback::Result result =
webrtc_transport_->video_encoder_factory()->SendEncodedFrame(
*frame, current_frame_stats_->capture_started_time,
current_frame_stats_->encode_started_time,
current_frame_stats_->encode_ended_time);
if (result.error != webrtc::EncodedImageCallback::Result::OK) {
// TODO(sergeyu): Stop the stream.
LOG(ERROR) << "Failed to send video frame.";
return;
}
// Send FrameStats message.
if (video_stats_dispatcher_.is_connected()) {
stats.frame_size = frame ? frame->data.size() : 0;
if (!current_frame_stats_->input_event_timestamps.is_null()) {
stats.capture_pending_delay =
current_frame_stats_->capture_started_time -
current_frame_stats_->input_event_timestamps.host_timestamp;
stats.latest_event_timestamp =
current_frame_stats_->input_event_timestamps.client_timestamp;
}
stats.capture_delay = current_frame_stats_->capture_delay;
// Total overhead time for IPC and threading when capturing frames.
stats.capture_overhead_delay =
(current_frame_stats_->capture_ended_time -
current_frame_stats_->capture_started_time) -
stats.capture_delay;
stats.encode_pending_delay = current_frame_stats_->encode_started_time -
current_frame_stats_->capture_ended_time;
stats.encode_delay = current_frame_stats_->encode_ended_time -
current_frame_stats_->encode_started_time;
stats.capturer_id = current_frame_stats_->capturer_id;
stats.frame_quality = current_frame_stats_->frame_quality;
video_stats_dispatcher_.OnVideoFrameStats(result.frame_id, stats);
}
}
void WebrtcVideoStream::OnEncoderCreated(
webrtc::VideoCodecType codec_type,
const webrtc::SdpVideoFormat::Parameters& parameters) {
DCHECK(thread_checker_.CalledOnValidThread());
// This method is called when SDP has been negotiated and WebRTC has
// created a preferred encoder via the WebrtcDummyVideoEncoderFactory
// implementation.
// Trigger recreating the encoder in case a previous one was being used and
// SDP renegotiation selected a different one. The proper encoder will be
// created after the next frame is captured.
// Note that this flag controls the encoder created by this class, and not
// the encoder that was just "created" by WebRTC.
// An optimization would be to do this only if the new encoder is different
// from the current one. However, SDP renegotiation is expected to occur
// infrequently (only when the user changes a setting), and should typically
// not cause the same codec to be repeatedly selected.
recreate_encoder_ = true;
// The preferred codec id depends on the order of
// |encoder_selector_|.RegisterEncoder().
if (codec_type == webrtc::kVideoCodecVP8) {
VLOG(0) << "VP8 video codec is preferred.";
encoder_selector_.SetPreferredCodec(0);
} else if (codec_type == webrtc::kVideoCodecVP9) {
encoder_selector_.SetPreferredCodec(1);
const auto iter = parameters.find(webrtc::kVP9FmtpProfileId);
bool losslessColor = iter != parameters.end() && iter->second == "1";
VLOG(0) << "VP9 video codec is preferred, losslessColor="
<< (losslessColor ? "true" : "false");
encoder_selector_.SetPreferredCodec(losslessColor ? 2 : 1);
} else if (codec_type == webrtc::kVideoCodecH264) {
#if defined(USE_H264_ENCODER)
VLOG(0) << "H264 video codec is preferred.";
encoder_selector_.SetPreferredCodec(3);
#else
NOTIMPLEMENTED();
#endif
} else {
LOG(FATAL) << "Unknown codec type: " << codec_type;
}
}
std::unique_ptr<WebrtcVideoEncoder> WebrtcVideoStream::CreateVP8Encoder() {
return std::make_unique<WebrtcVideoEncoderProxy>(
WebrtcVideoEncoderVpx::CreateForVP8(), encode_task_runner_);
}
std::unique_ptr<WebrtcVideoEncoder> WebrtcVideoStream::CreateVP9Encoder(
bool lossless_color) {
auto vp9encoder = WebrtcVideoEncoderVpx::CreateForVP9();
vp9encoder->SetLosslessColor(lossless_color);
return std::make_unique<WebrtcVideoEncoderProxy>(std::move(vp9encoder),
encode_task_runner_);
}
} // namespace protocol
} // namespace remoting