Replace SetCapturer and SetCaptureDevice by SetSource.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/media/engine/webrtcvideoengine2.h"
 
 #include <stdio.h>
 #include <algorithm>
 #include <set>
 #include <string>
 
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
-#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/engine/constants.h"
 #include "webrtc/media/engine/simulcast.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvideoencoderfactory.h"
 #include "webrtc/media/engine/webrtcvideoframe.h"
 #include "webrtc/media/engine/webrtcvoiceengine.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/video_decoder.h"
@@ skipping 959 matching lines @@
   {
     rtc::CritScope stream_lock(&stream_crit_);
     for (const auto& kv : send_streams_) {
       kv.second->SetSend(send);
     }
   }
   sending_ = send;
   return true;
 }
 
+// TODO(nisse): The enable argument was used for mute logic which has
+// been moved to VideoBroadcaster. So delete this method, and use
+// SetOptions instead.
 bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
                                        const VideoOptions* options) {
   TRACE_EVENT0("webrtc", "SetVideoSend");
   LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
                << "options: " << (options ? options->ToString() : "nullptr")
                << ").";
 
-  // TODO(solenberg): The state change should be fully rolled back if any one of
-  //                  these calls fail.
-  if (!MuteStream(ssrc, !enable)) {
-    return false;
-  }
   if (enable && options) {
     SetOptions(ssrc, *options);
   }
   return true;
 }
 
 bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
     const StreamParams& sp) const {
   for (uint32_t ssrc : sp.ssrcs) {
     if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
@@ skipping 282 matching lines @@
   // Get send stream bitrate stats.
   rtc::CritScope stream_lock(&stream_crit_);
   for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
            send_streams_.begin();
        stream != send_streams_.end(); ++stream) {
     stream->second->FillBandwidthEstimationInfo(&bwe_info);
   }
   video_media_info->bw_estimations.push_back(bwe_info);
 }
 
-bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
-  LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
-               << (capturer != NULL ? "(capturer)" : "NULL");
+void WebRtcVideoChannel2::SetSource(
+    uint32_t ssrc,
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  LOG(LS_INFO) << "SetSource: " << ssrc << " -> "
+               << (source ? "(source)" : "NULL");
   RTC_DCHECK(ssrc != 0);
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    const auto& kv = send_streams_.find(ssrc);
-    if (kv == send_streams_.end()) {
-      LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-      return false;
-    }
-    if (!kv->second->SetCapturer(capturer)) {
-      return false;
-    }
+
+  rtc::CritScope stream_lock(&stream_crit_);
+  const auto& kv = send_streams_.find(ssrc);
+  if (kv == send_streams_.end()) {
+    // Allow unknown ssrc only if source is null.
+    RTC_CHECK(source == nullptr);
+  } else {
+    kv->second->SetSource(source);
   }
-  return true;
 }
 
 void WebRtcVideoChannel2::OnPacketReceived(
     rtc::CopyOnWriteBuffer* packet,
     const rtc::PacketTime& packet_time) {
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   const webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(
           webrtc::MediaType::VIDEO,
@@ skipping 69 matching lines @@
       ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 void WebRtcVideoChannel2::OnNetworkRouteChanged(
     const std::string& transport_name,
     const NetworkRoute& network_route) {
   // TODO(honghaiz): uncomment this once the function in call is implemented.
   // call_->OnNetworkRouteChanged(transport_name, network_route);
 }
 
-bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
-  LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
-                  << (mute ? "mute" : "unmute");
-  RTC_DCHECK(ssrc != 0);
-  rtc::CritScope stream_lock(&stream_crit_);
-  const auto& kv = send_streams_.find(ssrc);
-  if (kv == send_streams_.end()) {
-    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-    return false;
-  }
-
-  kv->second->MuteStream(mute);
-  return true;
-}
-
 // TODO(pbos): Remove SetOptions in favor of SetSendParameters.
 void WebRtcVideoChannel2::SetOptions(uint32_t ssrc,
                                      const VideoOptions& options) {
   LOG(LS_INFO) << "SetOptions: ssrc " << ssrc << ": " << options.ToString();
 
   rtc::CritScope stream_lock(&stream_crit_);
   const auto& kv = send_streams_.find(ssrc);
   if (kv == send_streams_.end()) {
     return;
   }
@@ skipping 68 matching lines @@
     const std::vector<webrtc::RtpExtension>& rtp_extensions,
     // TODO(deadbeef): Don't duplicate information between send_params,
     // rtp_extensions, options, etc.
     const VideoSendParameters& send_params)
     : worker_thread_(rtc::Thread::Current()),
       ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
       call_(call),
       cpu_restricted_counter_(0),
       number_of_cpu_adapt_changes_(0),
-      capturer_(nullptr),
+      source_(nullptr),
       external_encoder_factory_(external_encoder_factory),
       stream_(nullptr),
       parameters_(config, options, max_bitrate_bps, codec_settings),
       rtp_parameters_(CreateRtpParametersWithOneEncoding()),
       pending_encoder_reconfiguration_(false),
       allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
       sending_(false),
-      muted_(false),
       first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
   parameters_.conference_mode = send_params.conference_mode;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
   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, kRtpVideoRotationHeaderExtension);
 
   if (codec_settings) {
     SetCodec(*codec_settings);
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
-  DisconnectCapturer();
+  DisconnectSource();
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
   DestroyVideoEncoder(&allocated_encoder_);
 }
 
 static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
                              int width,
                              int height,
                              webrtc::VideoRotation rotation) {
@@ skipping 12 matching lines @@
     const VideoFrame& frame) {
   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame");
   webrtc::VideoFrame video_frame(frame.GetVideoFrameBuffer(), 0, 0,
                                  frame.GetVideoRotation());
   rtc::CritScope cs(&lock_);
   if (stream_ == NULL) {
     // Frame input before send codecs are configured, dropping frame.
     return;
   }
 
-  if (muted_) {
-    // Create a black frame to transmit instead.
-    CreateBlackFrame(&video_frame,
-                     frame.width(),
-                     frame.height(),
-                     video_frame.rotation());
-  }
-
   int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
   // frame->GetTimeStamp() is essentially a delta, align to webrtc time
   if (first_frame_timestamp_ms_ == 0) {
     first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
   }
 
   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();
 
   // Not sending, abort after reconfiguration. Reconfiguration should still
   // occur to permit sending this input as quickly as possible once we start
   // sending (without having to reconfigure then).
   if (!sending_) {
     return;
   }
 
   stream_->Input()->IncomingCapturedFrame(video_frame);
 }
 
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
-    VideoCapturer* capturer) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSource(
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetSource");
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!DisconnectCapturer() && capturer == NULL) {
-    return false;
-  }
+
+  if (!source && !source_)
+    return;
+  DisconnectSource();
 
   {
     rtc::CritScope cs(&lock_);
 
     // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
     // new capturer may have a different timestamp delta than the previous one.
     first_frame_timestamp_ms_ = 0;
 
-    if (capturer == NULL) {
+    if (source == NULL) {
       if (stream_ != NULL) {
         LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
         webrtc::VideoFrame black_frame;
 
         CreateBlackFrame(&black_frame, last_dimensions_.width,
                          last_dimensions_.height, last_rotation_);
 
         // 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_ms_ += 1;
         black_frame.set_render_time_ms(last_frame_timestamp_ms_);
         stream_->Input()->IncomingCapturedFrame(black_frame);
       }
-
-      capturer_ = NULL;
-      return true;
     }
   }
-  capturer_ = capturer;
-  // |capturer_->AddOrUpdateSink| may not be called while holding |lock_| since
+  source_ = source;
+  // |source_->AddOrUpdateSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->AddOrUpdateSink(this, sink_wants_);
-  return true;
+  if (source_) {
+    source_->AddOrUpdateSink(this, sink_wants_);
+  }
 }
 
-void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
-  rtc::CritScope cs(&lock_);
-  muted_ = mute;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
+void WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectSource() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (capturer_ == NULL) {
-    return false;
+  if (source_ == NULL) {
+    return;
   }
 
-  // |capturer_->RemoveSink| may not be called while holding |lock_| since
+  // |source_->RemoveSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->RemoveSink(this);
-  capturer_ = NULL;
+  source_->RemoveSink(this);
+  source_ = nullptr;
   // Reset |cpu_restricted_counter_| if the capturer is changed. It is not
   // possible to know if the video resolution is restricted by CPU usage after
   // the capturer is changed since the next capturer might be screen capture
   // with another resolution and frame rate.
   cpu_restricted_counter_ = 0;
-  return true;
 }
 
 const std::vector<uint32_t>&
 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
   return ssrcs_;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
     const VideoOptions& options) {
   rtc::CritScope cs(&lock_);
@@ skipping 140 matching lines @@
           << "RecreateWebRtcStream (send) because of SetSendParameters";
       RecreateWebRtcStream();
     }
   }  // release |lock_|
 
   // |capturer_->AddOrUpdateSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
   if (params.rtp_header_extensions) {
     sink_wants_.rotation_applied = !ContainsHeaderExtension(
         *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension);
-    if (capturer_) {
-      capturer_->AddOrUpdateSink(this, sink_wants_);
+    if (source_) {
+      source_->AddOrUpdateSink(this, sink_wants_);
     }
   }
 }
 
 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters(
     const webrtc::RtpParameters& new_parameters) {
   if (!ValidateRtpParameters(new_parameters)) {
     return false;
   }
 
@@ skipping 144 matching lines @@
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
   if (worker_thread_ != rtc::Thread::Current()) {
     invoker_.AsyncInvoke<void>(
         worker_thread_,
         rtc::Bind(&WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate,
                   this, load));
     return;
   }
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!capturer_) {
+  if (!source_) {
     return;
   }
   {
     rtc::CritScope cs(&lock_);
     LOG(LS_INFO) << "OnLoadUpdate " << load << ", is_screencast: "
                  << (parameters_.options.is_screencast
                          ? (*parameters_.options.is_screencast ? "true"
                                                                : "false")
                          : "unset");
     // Do not adapt resolution for screen content as this will likely result in
@@ skipping 35 matching lines @@
       if (sink_wants_.max_pixel_count ||
           (sink_wants_.max_pixel_count_step_up &&
            *sink_wants_.max_pixel_count_step_up < *max_pixel_count_step_up)) {
         ++number_of_cpu_adapt_changes_;
         --cpu_restricted_counter_;
       }
     }
     sink_wants_.max_pixel_count = max_pixel_count;
     sink_wants_.max_pixel_count_step_up = max_pixel_count_step_up;
   }
-  // |capturer_->AddOrUpdateSink| may not be called while holding |lock_| since
+  // |source_->AddOrUpdateSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->AddOrUpdateSink(this, sink_wants_);
+  source_->AddOrUpdateSink(this, sink_wants_);
 }
 
 VideoSenderInfo
 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
   VideoSenderInfo info;
   webrtc::VideoSendStream::Stats stats;
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   {
     rtc::CritScope cs(&lock_);
     for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
@@ skipping 517 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket