enalbing low latency mode for RTC

Author: taste1981

BUILD.gn

@@ -62,7 +62,7 @@ if (!build_with_chromium) {
         "modules/audio_processing:audio_processing_tests",
         "modules/remote_bitrate_estimator:rtp_to_text",
         "modules/rtp_rtcp:test_packet_masks_metrics",
-        "modules/video_capture:video_capture_internal_impl",
+        "modules/video_capture:video_capture",
         "net/dcsctp:dcsctp_unittests",
         "pc:peerconnection_unittests",
         "pc:rtc_pc_unittests",
@@ -513,7 +513,7 @@ if (!build_with_chromium) {
       "logging:rtc_event_log_api",
       "media",
       "modules",
-      "modules/video_capture:video_capture_internal_impl",
+      "modules/video_capture:video_capture",
       "p2p:rtc_p2p",
       "pc:libjingle_peerconnection",
       "pc:rtc_pc",

api/audio_codecs/opus/audio_decoder_opus.cc

@@ -44,7 +44,7 @@ absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
         return absl::nullopt;  // Bad stereo parameter.
       }
     }
-    return 1;  // Default to mono.
+    return 2;  // Default to mono.
   }();
   if (absl::EqualsIgnoreCase(format.name, "opus") &&
       format.clockrate_hz == 48000 && format.num_channels == 2 &&

api/rtp_headers.h

@@ -144,6 +144,9 @@ struct RTPHeaderExtension {
   std::string mid;
 
   absl::optional<ColorSpace> color_space;
+
+  // Required for Temporal scalability support
+  uint16_t picture_id;
 };
 
 enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13

api/rtp_parameters.cc

@@ -133,6 +133,7 @@ constexpr char RtpExtension::kRidUri[];
 constexpr char RtpExtension::kRepairedRidUri[];
 constexpr char RtpExtension::kVideoFrameTrackingIdUri[];
 constexpr char RtpExtension::kCsrcAudioLevelsUri[];
+constexpr char RtpExtension::kPictureIdUri[];
 
 constexpr int RtpExtension::kMinId;
 constexpr int RtpExtension::kMaxId;
@@ -168,7 +169,8 @@ bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
          uri == webrtc::RtpExtension::kRidUri ||
          uri == webrtc::RtpExtension::kRepairedRidUri ||
          uri == webrtc::RtpExtension::kVideoLayersAllocationUri ||
-         uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri;
+         uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri ||
+		 uri == webrtc::RtpExtension::kPictureIdUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {