Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source removal.
BUG=webrtc:6371,webrtc:6983
Review-Url: https://codereview.webrtc.org/2469993003
Cr-Commit-Position: refs/heads/master@{#16048}
diff --git a/webrtc/media/base/videoengine_unittest.h b/webrtc/media/base/videoengine_unittest.h
index a673d35..a571bbc 100644
--- a/webrtc/media/base/videoengine_unittest.h
+++ b/webrtc/media/base/videoengine_unittest.h
@@ -830,14 +830,12 @@
rtc::Thread::Current()->ProcessMessages(30);
// Remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
- // Wait for one black frame for removing the capturer.
- EXPECT_FRAME_WAIT(2, kVideoWidth, kVideoHeight, kTimeout);
// No capturer was added, so this SetVideoSend shouldn't do anything.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
rtc::Thread::Current()->ProcessMessages(300);
// Verify no more frames were sent.
- EXPECT_EQ(2, renderer_.num_rendered_frames());
+ EXPECT_EQ(1, renderer_.num_rendered_frames());
}
// Tests that we can add and remove capturer as unique sources.
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index f09b254..ca65e65 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -1553,8 +1553,7 @@
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithOneEncoding()),
allocated_encoder_(nullptr, cricket::VideoCodec(), false),
- sending_(false),
- last_frame_timestamp_us_(0) {
+ sending_(false) {
parameters_.config.rtp.max_packet_size = kVideoMtu;
parameters_.conference_mode = send_params.conference_mode;
@@ -1612,43 +1611,6 @@
DestroyVideoEncoder(&allocated_encoder_);
}
-void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
- const webrtc::VideoFrame& frame) {
- TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame");
- webrtc::VideoFrame video_frame(frame.video_frame_buffer(),
- frame.rotation(),
- frame.timestamp_us());
-
- rtc::CritScope cs(&lock_);
-
- if (video_frame.width() != last_frame_info_.width ||
- video_frame.height() != last_frame_info_.height ||
- video_frame.rotation() != last_frame_info_.rotation ||
- video_frame.is_texture() != last_frame_info_.is_texture) {
- last_frame_info_.width = video_frame.width();
- last_frame_info_.height = video_frame.height();
- last_frame_info_.rotation = video_frame.rotation();
- last_frame_info_.is_texture = video_frame.is_texture();
-
- LOG(LS_INFO) << "Video frame parameters changed: dimensions="
- << last_frame_info_.width << "x" << last_frame_info_.height
- << ", rotation=" << last_frame_info_.rotation
- << ", texture=" << last_frame_info_.is_texture;
- }
-
- if (encoder_sink_ == NULL) {
- // Frame input before send codecs are configured, dropping frame.
- return;
- }
-
- last_frame_timestamp_us_ = video_frame.timestamp_us();
-
- // Forward frame to the encoder regardless if we are sending or not. This is
- // to ensure that the encoder can be reconfigured with the correct frame size
- // as quickly as possible.
- encoder_sink_->OnFrame(video_frame);
-}
-
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoSend(
bool enable,
const VideoOptions* options,
@@ -1658,7 +1620,6 @@
// Ignore |options| pointer if |enable| is false.
bool options_present = enable && options;
- bool source_changing = source_ != source;
if (options_present) {
VideoOptions old_options = parameters_.options;
@@ -1668,29 +1629,6 @@
}
}
- if (source_changing) {
- rtc::CritScope cs(&lock_);
- if (source == nullptr && last_frame_info_.width > 0 && encoder_sink_) {
- LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
- // Force this black frame not to be dropped due to timestamp order
- // check. As IncomingCapturedFrame will drop the frame if this frame's
- // timestamp is less than or equal to last frame's timestamp, it is
- // necessary to give this black frame a larger timestamp than the
- // previous one.
- last_frame_timestamp_us_ += rtc::kNumMicrosecsPerMillisec;
- rtc::scoped_refptr<webrtc::I420Buffer> black_buffer(
- webrtc::I420Buffer::Create(last_frame_info_.width,
- last_frame_info_.height));
- webrtc::I420Buffer::SetBlack(black_buffer);
-
- encoder_sink_->OnFrame(webrtc::VideoFrame(
- black_buffer, last_frame_info_.rotation, last_frame_timestamp_us_));
- }
- }
-
- // TODO(perkj, nisse): Remove |source_| and directly call
- // |stream_|->SetSource(source) once the video frame types have been
- // merged.
if (source_ && stream_) {
stream_->SetSource(
nullptr, webrtc::VideoSendStream::DegradationPreference::kBalanced);
@@ -1700,6 +1638,8 @@
if (source && stream_) {
// Do not adapt resolution for screen content as this will likely
// result in blurry and unreadable text.
+ // |this| acts like a VideoSource to make sure SinkWants are handled on the
+ // correct thread.
stream_->SetSource(
this, enable_cpu_overuse_detection_ &&
!parameters_.options.is_screencast.value_or(false)
@@ -1973,45 +1913,35 @@
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::RemoveSink(
- VideoSinkInterface<webrtc::VideoFrame>* sink) {
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
RTC_DCHECK_RUN_ON(&thread_checker_);
- {
- rtc::CritScope cs(&lock_);
- RTC_DCHECK(encoder_sink_ == sink);
- encoder_sink_ = nullptr;
- }
- source_->RemoveSink(this);
+ RTC_DCHECK(encoder_sink_ == sink);
+ encoder_sink_ = nullptr;
+ source_->RemoveSink(sink);
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::AddOrUpdateSink(
- VideoSinkInterface<webrtc::VideoFrame>* sink,
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) {
if (worker_thread_ == rtc::Thread::Current()) {
// AddOrUpdateSink is called on |worker_thread_| if this is the first
// registration of |sink|.
RTC_DCHECK_RUN_ON(&thread_checker_);
- {
- rtc::CritScope cs(&lock_);
- encoder_sink_ = sink;
- }
- source_->AddOrUpdateSink(this, wants);
+ encoder_sink_ = sink;
+ source_->AddOrUpdateSink(encoder_sink_, wants);
} else {
// Subsequent calls to AddOrUpdateSink will happen on the encoder task
// queue.
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [this, wants] {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- bool encoder_sink_valid = true;
- {
- rtc::CritScope cs(&lock_);
- encoder_sink_valid = encoder_sink_ != nullptr;
- }
- // Since |source_| is still valid after a call to RemoveSink, check if
- // |encoder_sink_| is still valid to check if this call should be
- // cancelled.
- if (source_ && encoder_sink_valid) {
- source_->AddOrUpdateSink(this, wants);
- }
- });
+ invoker_.AsyncInvoke<void>(
+ RTC_FROM_HERE, worker_thread_, [this, sink, wants] {
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+ // |sink| may be invalidated after this task was posted since
+ // RemoveSink is called on the worker thread.
+ bool encoder_sink_valid = (sink == encoder_sink_);
+ if (source_ && encoder_sink_valid) {
+ source_->AddOrUpdateSink(encoder_sink_, wants);
+ }
+ });
}
}
@@ -2135,12 +2065,10 @@
parameters_.encoder_config.encoder_specific_settings = NULL;
if (source_) {
- // TODO(perkj, nisse): Remove |source_| and directly call
- // |stream_|->SetSource(source) once the video frame types have been
- // merged and |stream_| internally reconfigure the encoder on frame
- // resolution change.
// Do not adapt resolution for screen content as this will likely result in
// blurry and unreadable text.
+ // |this| acts like a VideoSource to make sure SinkWants are handled on the
+ // correct thread.
stream_->SetSource(
this, enable_cpu_overuse_detection_ &&
!parameters_.options.is_screencast.value_or(false)
diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
index 1fb794e..b50c5a1 100644
--- a/webrtc/media/engine/webrtcvideoengine2.h
+++ b/webrtc/media/engine/webrtcvideoengine2.h
@@ -231,11 +231,9 @@
static std::string CodecSettingsVectorToString(
const std::vector<VideoCodecSettings>& codecs);
- // Wrapper for the sender part, this is where the source is connected and
- // frames are then converted from cricket frames to webrtc frames.
+ // Wrapper for the sender part.
class WebRtcVideoSendStream
- : public rtc::VideoSinkInterface<webrtc::VideoFrame>,
- public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+ : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
public:
WebRtcVideoSendStream(
webrtc::Call* call,
@@ -256,14 +254,12 @@
// Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
// WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream
- // in |stream_|.
- // TODO(perkj, nisse): Refactor WebRtcVideoSendStream to directly connect
- // the camera input |source_|
- void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+ // in |stream_|. This is done to proxy VideoSinkWants from the encoder to
+ // the worker thread.
+ void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override;
- void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+ void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
- void OnFrame(const webrtc::VideoFrame& frame) override;
bool SetVideoSend(bool mute,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
@@ -306,21 +302,6 @@
bool external;
};
- // TODO(perkj): VideoFrameInfo is currently used for sending a black frame
- // when the video source is removed. Consider moving that logic to
- // VieEncoder or remove it.
- struct VideoFrameInfo {
- VideoFrameInfo()
- : width(0),
- height(0),
- rotation(webrtc::kVideoRotation_0),
- is_texture(false) {}
- int width;
- int height;
- webrtc::VideoRotation rotation;
- bool is_texture;
- };
-
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
ConfigureVideoEncoderSettings(const VideoCodec& codec);
AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec);
@@ -348,10 +329,9 @@
WebRtcVideoEncoderFactory* const external_encoder_factory_
ACCESS_ON(&thread_checker_);
- rtc::CriticalSection lock_;
webrtc::VideoSendStream* stream_ ACCESS_ON(&thread_checker_);
rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
- GUARDED_BY(lock_);
+ ACCESS_ON(&thread_checker_);
// Contains settings that are the same for all streams in the MediaChannel,
// such as codecs, header extensions, and the global bitrate limit for the
// entire channel.
@@ -363,13 +343,8 @@
// one stream per MediaChannel.
webrtc::RtpParameters rtp_parameters_ ACCESS_ON(&thread_checker_);
AllocatedEncoder allocated_encoder_ ACCESS_ON(&thread_checker_);
- VideoFrameInfo last_frame_info_ GUARDED_BY(lock_);
bool sending_ ACCESS_ON(&thread_checker_);
-
- // The timestamp of the last frame received
- // Used to generate timestamp for the black frame when source is removed
- int64_t last_frame_timestamp_us_ GUARDED_BY(lock_);
};
// Wrapper for the receiver part, contains configs etc. that are needed to
diff --git a/webrtc/media/engine/webrtcvideoengine2_unittest.cc b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
index 8349469..852eedd 100644
--- a/webrtc/media/engine/webrtcvideoengine2_unittest.cc
+++ b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
@@ -1654,8 +1654,7 @@
<< "Non-screenshare shouldn't use min-transmit bitrate.";
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
- // Removing a capturer triggers a black frame to be sent.
- EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
+ EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
VideoOptions screencast_options;
screencast_options.is_screencast = rtc::Optional<bool>(true);
EXPECT_TRUE(
@@ -1663,7 +1662,7 @@
EXPECT_TRUE(capturer.CaptureFrame());
// Send stream not recreated after option change.
ASSERT_EQ(send_stream, fake_call_->GetVideoSendStreams().front());
- EXPECT_EQ(3, send_stream->GetNumberOfSwappedFrames());
+ EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
// Verify screencast settings.
encoder_config = send_stream->GetEncoderConfig().Copy();
