Add diffsrv for screen sharing.

Author: taste1981

talk/owt/sdk/base/globalconfiguration.cc

@@ -30,5 +30,6 @@ int GlobalConfiguration::screen_max_ = 0;
 int GlobalConfiguration::data_min_ = 0;
 int GlobalConfiguration::data_max_ = 0;
 bool GlobalConfiguration::pre_decode_dump_enabled_ = false;
+bool GlobalConfiguration::post_encode_dump_enabled_ = false;
 }  // namespace base
 }

talk/owt/sdk/base/peerconnectiondependencyfactory.cc

@@ -115,6 +115,11 @@ void PeerConnectionDependencyFactory::
     field_trial_ += "WebRTC-DecoderDataDumpDirectory/./";
   }
 
+  bool post_encode_dump = GlobalConfiguration::GetPostEncodeDumpEnabled();
+  if (post_encode_dump) {
+    field_trial_ += "WebRTC-EncoderDataDumpDirectory/./";
+  }
+
   // Set H.264 temporal layers. Ideally it should be set via RtpSenderParam
   int h264_temporal_layers = GlobalConfiguration::GetH264TemporalLayers();
   field_trial_ +=

talk/owt/sdk/base/sdputils.cc

@@ -42,7 +42,7 @@ std::string SdpUtils::SetPreferAudioCodecs(const std::string& original_sdp,
   return cur_sdp;
 }
 std::string SdpUtils::SetPreferVideoCodecs(const std::string& original_sdp,
-                                          std::vector<VideoCodec>& codec) {
+                                          std::vector<VideoCodec>& codec, bool set_quality) {
   std::string cur_sdp(original_sdp);
   if (codec.size() == 0)
     return cur_sdp;
@@ -56,7 +56,7 @@ std::string SdpUtils::SetPreferVideoCodecs(const std::string& original_sdp,
     }
     codec_names.push_back(codec_it->second);
   }
-  cur_sdp = SdpUtils::SetPreferCodecs(cur_sdp, codec_names, false);
+  cur_sdp = SdpUtils::SetPreferCodecs(cur_sdp, codec_names, false, set_quality);
   return cur_sdp;
 }
 std::vector<std::string> SdpUtils::GetCodecValues(const std::string& sdp,
@@ -81,7 +81,7 @@ std::vector<std::string> SdpUtils::GetCodecValues(const std::string& sdp,
 // TODO: unify to std::regex impl for Linux builds.
 std::string SdpUtils::SetPreferCodecs(const std::string& sdp,
     std::vector<std::string>& codec_names,
-    bool is_audio) {
+    bool is_audio, bool set_quality) {
   // Search all rtx maps in the sdp.
   std::regex reg_fmtp_apt(
       "a=fmtp:(\\d+) apt=(\\d+)(?=[\r]?[\n]?)",
@@ -193,9 +193,28 @@ std::string SdpUtils::SetPreferCodecs(const std::string& sdp,
   for (auto& codec_value : kept_codec_values) {
     m_line_stream << " " << codec_value;
   }
+
   RTC_LOG(LS_INFO) << "New m-line: " << m_line_stream.str();
   std::string before_strip = std::regex_replace(sdp, reg_m_line, m_line_stream.str());
   std::string after_strip = before_strip;
+
+  if (!is_audio && set_quality) {
+    // Search for c-line and add a=priority line after it.
+    std::regex reg_c_line("c=IN .*\\r\\n");
+    std::smatch c_line_match;
+    auto cline_search_result =
+        std::regex_search(after_strip, c_line_match, reg_c_line);
+    if (!cline_search_result || c_line_match.size() == 0) {
+      RTC_LOG(LS_WARNING) << "c-line is not found. SDP: " << sdp;
+      return sdp;
+    }
+    std::string c_line(c_line_match[0]);
+    std::stringstream c_line_stream(c_line);
+    std::string new_cline = c_line_stream.str() + "a=quality:10\r\n";
+
+    after_strip = std::regex_replace(before_strip, reg_c_line, new_cline);
+    before_strip = after_strip;
+  }
   // Remove all a=fmtp:xx, a=rtpmap:xx and a=rtcp-fb:xx where xx is not in m-line,
   // this includes the a=fmtp:xx apt:yy lines for rtx.
   for (size_t i = 3, m_line_vector_size = m_line_vector.size(); i < m_line_vector_size; i++) {

talk/owt/sdk/base/sdputils.h

@@ -14,7 +14,7 @@ class SdpUtils {
   static std::string SetPreferAudioCodecs(const std::string& sdp,
                                          std::vector<AudioCodec>& codec);
   static std::string SetPreferVideoCodecs(const std::string& sdp,
-                                         std::vector<VideoCodec>& codec);
+                                         std::vector<VideoCodec>& codec, bool set_quality = false);
  private:
   /**
    @brief Replace SDP for preferred codec.
@@ -24,7 +24,7 @@ class SdpUtils {
    */
   static std::string SetPreferCodecs(const std::string& sdp,
                                      std::vector<std::string>& codec_name,
-                                     bool is_audio);
+                                     bool is_audio, bool set_quality = false);
   static std::vector<std::string> GetCodecValues(const std::string& sdp,
                                                  std::string& codec_name,
                                                  bool is_audio);

talk/owt/sdk/base/win/msdkvideoencoder.cc

@@ -6,20 +6,23 @@
 #include <vector>
 #include "libyuv/convert_from.h"
 #include "mfxcommon.h"
+#include "absl/algorithm/container.h"
 #include "talk/owt/sdk/base/mediautils.h"
 #include "talk/owt/sdk/base/win/d3d_allocator.h"
 #include "talk/owt/sdk/base/win/msdkvideobase.h"
 #include "talk/owt/sdk/base/win/msdkvideoencoder.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h"
-#include "webrtc/rtc_base/bind.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/thread.h"
-#include "webrtc/media/base/vp9_profile.h"
 #include "webrtc/common_video/h264/h264_common.h"
 #ifndef DISABLE_H265
 #include "webrtc/common_video/h265/h265_common.h"
 #endif
+#include "webrtc/media/base/vp9_profile.h"
+#include "webrtc/rtc_base/bind.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/system/file_wrapper.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/thread.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 #include "vp9/ratectrl_rtc.h"
 
 using namespace rtc;
@@ -95,6 +98,26 @@ MSDKVideoEncoder::MSDKVideoEncoder(const cricket::VideoCodec& format)
   gof.SetGofInfoVP9(webrtc::TemporalStructureMode::kTemporalStructureMode1);
   gof_idx = 0;
   rtp_codec_parameters = format;
+
+  encoder_dump_file_name =
+      webrtc::field_trial::FindFullName("WebRTC-EncoderDataDumpDirectory");
+  // Because '/' can't be used inside a field trial parameter, we use ';'
+  // instead.
+  // This is only relevant to WebRTC-EncoderDataDumpDirectory
+  // 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(encoder_dump_file_name, ';', '/');
+  if (!encoder_dump_file_name.empty()) {
+    enable_bitstream_dump = true;
+    char filename_buffer[256];
+    rtc::SimpleStringBuilder ssb(filename_buffer);
+    ssb << encoder_dump_file_name << "/webrtc_send_stream_"
+        << rtc::TimeMicros() << ".ivf";
+    dump_writer =
+        webrtc::IvfFileWriter::Wrap(webrtc::FileWrapper::OpenWriteOnly(
+            ssb.str()), /* byte_limit= */ 100000000);
+  }
 }
 
 MSDKVideoEncoder::~MSDKVideoEncoder() {
@@ -400,7 +423,7 @@ int MSDKVideoEncoder::InitEncodeOnEncoderThread(
   vp9_rate_ctrl = VP9RateControl::Create(vp9_rc_config);
   memset(&frame_params, 0, sizeof(frame_params));
 
-  // MSDK is using ExtAvcTemporalLayers for HEVC as well. This should be fixed.
+  // TODO: MSDK is using ExtAvcTemporalLayers for HEVC as well. This should be fixed.
   if ((codec_id == MFX_CODEC_AVC || codec_id == MFX_CODEC_HEVC) && num_temporal_layers > 1) {
     mfxExtAvcTemporalLayers extAvcTemporalLayers;
     MSDK_ZERO_MEMORY(extAvcTemporalLayers);
@@ -823,6 +846,11 @@ int MSDKVideoEncoder::Encode(
     bs.DataOffset = 0;
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
+
+  if (enable_bitstream_dump && dump_writer.get()) {
+    dump_writer->WriteFrame(encodedFrame, codec_type);
+  }
+
   if (pbsData != nullptr) {
     delete[] pbsData;
     pbsData = nullptr;

talk/owt/sdk/base/win/msdkvideoencoder.h

@@ -14,6 +14,7 @@
 #include "mfxvideo++.h"
 #include "mfxvideo.h"
 #include "sysmem_allocator.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/api/video_codecs/video_codec.h"
 #include "webrtc/api/video_codecs/video_encoder.h"
 #include "webrtc/media/base/codec.h"
@@ -113,6 +114,9 @@ class MSDKVideoEncoder : public webrtc::VideoEncoder {
   // Gof related information for VP9 codec specific info.
   uint8_t gof_idx;
   webrtc::GofInfoVP9 gof;
+  std::unique_ptr<webrtc::IvfFileWriter> dump_writer;
+  bool enable_bitstream_dump = false;
+  std::string encoder_dump_file_name;
 };
 }  // namespace base
 }  // namespace owt

talk/owt/sdk/conference/conferencepeerconnectionchannel.cc

@@ -307,7 +307,13 @@ void ConferencePeerConnectionChannel::OnCreateSessionDescriptionSuccess(
   for (auto& video_enc_param : configuration_.video) {
     video_codecs.push_back(video_enc_param.codec.name);
   }
-  sdp_string = SdpUtils::SetPreferVideoCodecs(sdp_string, video_codecs);
+  bool is_screen = published_stream_.get() ? (published_stream_->Source().video ==
+                          owt::base::VideoSourceInfo::kScreenCast)
+                       : (subscribed_stream_.get()
+                          ? (subscribed_stream_->Source().video ==
+                                    owt::base::VideoSourceInfo::kScreenCast)
+                          : false);
+  sdp_string = SdpUtils::SetPreferVideoCodecs(sdp_string, video_codecs, is_screen);
   webrtc::SessionDescriptionInterface* new_desc(
       webrtc::CreateSessionDescription(desc->type(), sdp_string, nullptr));
   peer_connection_->SetLocalDescription(observer, new_desc);

talk/owt/sdk/include/cpp/owt/base/globalconfiguration.h

@@ -110,6 +110,12 @@ class GlobalConfiguration {
     pre_decode_dump_enabled_ = enabled;
   }
 
+  /**
+   @brief This function enables stream dump after encoder.
+  */
+  static void SetPostEncodeDumpEnabled(bool enabled) {
+    post_encode_dump_enabled_ = enabled;
+  }
   /**
    @brief This function sets the temporal layers for H.264.
 
@@ -235,11 +241,24 @@ class GlobalConfiguration {
   static int data_min_;
   static int data_max_;
 
+  /**
+   @brief This function enables dumping of bitstream before decoding.
+  */
   static bool GetPreDecodeDumpEnabled() {
     return pre_decode_dump_enabled_;
   }
 
   static bool pre_decode_dump_enabled_;
+
+  /**
+   @brief This function enables dumping of bitstream after encoding.
+  */
+  static bool GetPostEncodeDumpEnabled() {
+    return post_encode_dump_enabled_;
+  }
+
+  static bool post_encode_dump_enabled_;
+
   /**
    @brief This function gets whether auto echo cancellation is enabled or not.
    @return true or false.