Index: webrtc/media/engine/webrtcvideoengine2.cc |
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc |
index 82d32b0e1865dff74586b16030a8731a6634a5ec..432c025e30f2a9d85106fcdde2dc0489cfeaf159 100644 |
--- a/webrtc/media/engine/webrtcvideoengine2.cc |
+++ b/webrtc/media/engine/webrtcvideoengine2.cc |
@@ -748,7 +748,7 @@ bool WebRtcVideoChannel2::GetChangedSendParameters( |
// Handle RTP header extensions. |
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true); |
- if (send_rtp_extensions_ != filtered_extensions) { |
+ if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) { |
changed_params->rtp_header_extensions = |
rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
} |
@@ -796,7 +796,7 @@ bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) { |
} |
if (changed_params.rtp_header_extensions) { |
- send_rtp_extensions_ = *changed_params.rtp_header_extensions; |
+ send_rtp_extensions_ = changed_params.rtp_header_extensions; |
} |
if (changed_params.codec || changed_params.max_bandwidth_bps) { |
@@ -1520,7 +1520,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( |
bool enable_cpu_overuse_detection, |
int max_bitrate_bps, |
const rtc::Optional<VideoCodecSettings>& codec_settings, |
- const std::vector<webrtc::RtpExtension>& rtp_extensions, |
+ const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions, |
// TODO(deadbeef): Don't duplicate information between send_params, |
// rtp_extensions, options, etc. |
const VideoSendParameters& send_params) |
@@ -1546,15 +1546,23 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( |
sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, |
¶meters_.config.rtp.rtx.ssrcs); |
parameters_.config.rtp.c_name = sp.cname; |
- parameters_.config.rtp.extensions = rtp_extensions; |
+ if (rtp_extensions) { |
+ parameters_.config.rtp.extensions = *rtp_extensions; |
+ } |
parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size |
? webrtc::RtcpMode::kReducedSize |
: webrtc::RtcpMode::kCompound; |
parameters_.config.overuse_callback = |
enable_cpu_overuse_detection ? this : nullptr; |
- sink_wants_.rotation_applied = !ContainsHeaderExtension( |
- rtp_extensions, webrtc::RtpExtension::kVideoRotationUri); |
+ // Only request rotation at the source when we positively know that the remote |
+ // side doesn't support the rotation extension. This allows us to prepare the |
+ // encoder in the expectation that rotation is supported - which is the common |
+ // case. |
+ sink_wants_.rotation_applied = |
+ rtp_extensions && |
+ !ContainsHeaderExtension(*rtp_extensions, |
+ webrtc::RtpExtension::kVideoRotationUri); |
if (codec_settings) { |
SetCodec(*codec_settings); |
@@ -1593,6 +1601,23 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame( |
webrtc::VideoFrame video_frame(frame.video_frame_buffer(), 0, 0, |
frame.rotation()); |
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(); |
+ pending_encoder_reconfiguration_ = true; |
+ |
+ 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 (stream_ == NULL) { |
// Frame input before send codecs are configured, dropping frame. |
return; |
@@ -1609,9 +1634,11 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame( |
last_frame_timestamp_ms_ = *first_frame_timestamp_ms_ + frame_delta_ms; |
video_frame.set_render_time_ms(last_frame_timestamp_ms_); |
- // Reconfigure codec if necessary. |
- SetDimensions(video_frame.width(), video_frame.height()); |
- last_rotation_ = video_frame.rotation(); |
+ |
+ if (pending_encoder_reconfiguration_) { |
+ ReconfigureEncoder(); |
+ pending_encoder_reconfiguration_ = false; |
+ } |
// Not sending, abort after reconfiguration. Reconfiguration should still |
// occur to permit sending this input as quickly as possible once we start |
@@ -1664,9 +1691,9 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoSend( |
// necessary to give this black frame a larger timestamp than the |
// previous one. |
last_frame_timestamp_ms_ += 1; |
- stream_->Input()->IncomingCapturedFrame( |
- CreateBlackFrame(last_dimensions_.width, last_dimensions_.height, |
- last_frame_timestamp_ms_, last_rotation_)); |
+ stream_->Input()->IncomingCapturedFrame(CreateBlackFrame( |
+ last_frame_info_.width, last_frame_info_.height, |
+ last_frame_timestamp_ms_, last_frame_info_.rotation)); |
} |
source_ = source; |
} |
@@ -1758,8 +1785,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder( |
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec( |
const VideoCodecSettings& codec_settings) { |
- parameters_.encoder_config = |
- CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec); |
+ parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec); |
RTC_DCHECK(!parameters_.encoder_config.streams.empty()); |
AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec); |
@@ -1898,7 +1924,6 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() { |
webrtc::VideoEncoderConfig |
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
- const Dimensions& dimensions, |
const VideoCodec& codec) const { |
webrtc::VideoEncoderConfig encoder_config; |
bool is_screencast = parameters_.options.is_screencast.value_or(false); |
@@ -1914,8 +1939,8 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
} |
// Restrict dimensions according to codec max. |
- int width = dimensions.width; |
- int height = dimensions.height; |
+ int width = last_frame_info_.width; |
+ int height = last_frame_info_.height; |
if (!is_screencast) { |
if (codec.width < width) |
width = codec.width; |
@@ -1940,6 +1965,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
parameters_.max_bitrate_bps); |
encoder_config.streams = CreateVideoStreams( |
clamped_codec, parameters_.options, stream_max_bitrate, stream_count); |
+ encoder_config.expect_encode_from_texture = last_frame_info_.is_texture; |
// Conference mode screencast uses 2 temporal layers split at 100kbit. |
if (parameters_.conference_mode && is_screencast && |
@@ -1964,25 +1990,14 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
return encoder_config; |
} |
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( |
- int width, |
- int height) { |
- if (last_dimensions_.width == width && last_dimensions_.height == height && |
- !pending_encoder_reconfiguration_) { |
- // Configured using the same parameters, do not reconfigure. |
- return; |
- } |
- |
- last_dimensions_.width = width; |
- last_dimensions_.height = height; |
- |
+void WebRtcVideoChannel2::WebRtcVideoSendStream::ReconfigureEncoder() { |
RTC_DCHECK(!parameters_.encoder_config.streams.empty()); |
RTC_CHECK(parameters_.codec_settings); |
VideoCodecSettings codec_settings = *parameters_.codec_settings; |
webrtc::VideoEncoderConfig encoder_config = |
- CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec); |
+ CreateVideoEncoderConfig(codec_settings.codec); |
encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings( |
codec_settings.codec); |
@@ -1990,7 +2005,6 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( |
stream_->ReconfigureVideoEncoder(encoder_config); |
encoder_config.encoder_specific_settings = NULL; |
- pending_encoder_reconfiguration_ = false; |
parameters_.encoder_config = encoder_config; |
} |
@@ -2035,7 +2049,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) { |
// equal to |max_pixel_count| depending on how the source can scale the |
// input frame size. |
max_pixel_count = rtc::Optional<int>( |
- (last_dimensions_.height * last_dimensions_.width * 3) / 5); |
+ (last_frame_info_.height * last_frame_info_.width * 3) / 5); |
// Increase |number_of_cpu_adapt_changes_| if |
// sink_wants_.max_pixel_count will be changed since |
// last time |source_->AddOrUpdateSink| was called. That is, this will |
@@ -2050,8 +2064,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) { |
// The input video frame size will have a resolution with "one step up" |
// pixels than |max_pixel_count_step_up| where "one step up" depends on |
// how the source can scale the input frame size. |
- max_pixel_count_step_up = rtc::Optional<int>(last_dimensions_.height * |
- last_dimensions_.width); |
+ max_pixel_count_step_up = |
+ rtc::Optional<int>(last_frame_info_.height * last_frame_info_.width); |
// Increase |number_of_cpu_adapt_changes_| if |
// sink_wants_.max_pixel_count_step_up will be changed since |
// last time |source_->AddOrUpdateSink| was called. That is, this will |