Fix build issues after merging changes for low latency mode.

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",
+        "modules/video_capture:video_capture_internal_impl",
         "net/dcsctp:dcsctp_unittests",
         "pc:peerconnection_unittests",
         "pc:rtc_pc_unittests",
@@ -293,10 +293,6 @@ config("common_config") {
     defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
   }
 
-  if (owt_use_gpra) {
-    defines += [ "INTEL_GPRA" ]
-  }
-
   if (rtc_include_internal_audio_device) {
     defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
   }
@@ -517,7 +513,7 @@ if (!build_with_chromium) {
       "logging:rtc_event_log_api",
       "media",
       "modules",
-      "modules/video_capture:video_capture",
+      "modules/video_capture:video_capture_internal_impl",
       "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 2;  // Default to mono.
+    return 1;  // Default to mono.
   }();
   if (absl::EqualsIgnoreCase(format.name, "opus") &&
       format.clockrate_hz == 48000 && format.num_channels == 2 &&

api/stats/rtc_stats_report.h

@@ -23,7 +23,6 @@
 #include "api/ref_counted_base.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats.h"
-#include "rtc_base/constructor_magic.h"
 // TODO(tommi): Remove this include after fixing iwyu issue in chromium.
 // See: third_party/blink/renderer/platform/peerconnection/rtc_stats.cc
 #include "rtc_base/ref_counted_object.h"
@@ -58,15 +57,14 @@ class RTC_EXPORT RTCStatsReport final
     // Reference report to make sure it is kept alive.
     rtc::scoped_refptr<const RTCStatsReport> report_;
     StatsMap::const_iterator it_;
-    RTC_DISALLOW_COPY_AND_ASSIGN(ConstIterator);
   };
 
   // TODO(hbos): Remove "= 0" once Chromium unittest has been updated to call
   // with a parameter. crbug.com/627816
   static rtc::scoped_refptr<RTCStatsReport> Create(int64_t timestamp_us = 0);
 
   explicit RTCStatsReport(int64_t timestamp_us);
-  //RTCStatsReport(const RTCStatsReport& other) = delete;
+  RTCStatsReport(const RTCStatsReport& other) = delete;
   rtc::scoped_refptr<RTCStatsReport> Copy() const;
 
   int64_t timestamp_us() const { return timestamp_us_; }
@@ -132,7 +130,6 @@ class RTC_EXPORT RTCStatsReport final
  private:
   int64_t timestamp_us_;
   StatsMap stats_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(RTCStatsReport);
 };
 
 }  // namespace webrtc

api/transport/bitrate_settings.h

@@ -42,7 +42,7 @@ struct BitrateConstraints {
   int max_bitrate_bps = -1;
 
  private:
-  static constexpr int kDefaultStartBitrateBps = 5000000;  // Update from 300k to 5MB for low latency mode
+  static constexpr int kDefaultStartBitrateBps = 300000;
 };
 
 }  // namespace webrtc

api/video_codecs/video_codec.cc

@@ -53,7 +53,6 @@ bool VideoCodecH264::operator==(const VideoCodecH264& other) const {
           numberOfTemporalLayers == other.numberOfTemporalLayers);
 }
 
-#ifndef DISABLE_H265
 bool VideoCodecH265::operator==(const VideoCodecH265& other) const {
   return (frameDroppingOn == other.frameDroppingOn &&
           keyFrameInterval == other.keyFrameInterval &&
@@ -62,18 +61,6 @@ bool VideoCodecH265::operator==(const VideoCodecH265& other) const {
           (spsLen == 0 || memcmp(spsData, other.spsData, spsLen) == 0) &&
           (ppsLen == 0 || memcmp(ppsData, other.ppsData, ppsLen) == 0));
 }
-#endif
-
-#ifdef WEBRTC_USE_H265
-bool VideoCodecH265::operator==(const VideoCodecH265& other) const {
-  return (frameDroppingOn == other.frameDroppingOn &&
-          keyFrameInterval == other.keyFrameInterval &&
-          vpsLen == other.vpsLen && spsLen == other.spsLen &&
-          ppsLen == other.ppsLen &&
-          (spsLen == 0 || memcmp(spsData, other.spsData, spsLen) == 0) &&
-          (ppsLen == 0 || memcmp(ppsData, other.ppsData, ppsLen) == 0));
-}
-#endif
 
 VideoCodec::VideoCodec()
     : codecType(kVideoCodecGeneric),
@@ -134,7 +121,6 @@ const VideoCodecH265& VideoCodec::H265() const {
   RTC_DCHECK_EQ(codecType, kVideoCodecH265);
   return codec_specific_.H265;
 }
-#endif
 
 const char* CodecTypeToPayloadString(VideoCodecType type) {
   switch (type) {
@@ -148,7 +134,6 @@ const char* CodecTypeToPayloadString(VideoCodecType type) {
       return kPayloadNameH264;
     case kVideoCodecH265:
       return kPayloadNameH265;
-#endif
     case kVideoCodecMultiplex:
       return kPayloadNameMultiplex;
     case kVideoCodecGeneric:
@@ -172,7 +157,6 @@ VideoCodecType PayloadStringToCodecType(const std::string& name) {
     return kVideoCodecMultiplex;
   if (absl::EqualsIgnoreCase(name, kPayloadNameH265))
     return kVideoCodecH265;
-#endif
   return kVideoCodecGeneric;
 }
 

api/video_codecs/video_codec.h

@@ -112,23 +112,6 @@ struct VideoCodecH265 {
   size_t ppsLen;
 };
 
-#ifndef DISABLE_H265
-struct VideoCodecH265 {
-  bool operator==(const VideoCodecH265& other) const;
-  bool operator!=(const VideoCodecH265& other) const {
-    return !(*this == other);
-  }
-  bool frameDroppingOn;
-  int keyFrameInterval;
-  const uint8_t* vpsData;
-  size_t vpsLen;
-  const uint8_t* spsData;
-  size_t spsLen;
-  const uint8_t* ppsData;
-  size_t ppsLen;
-};
-#endif
-
 // Translates from name of codec to codec type and vice versa.
 RTC_EXPORT const char* CodecTypeToPayloadString(VideoCodecType type);
 RTC_EXPORT VideoCodecType PayloadStringToCodecType(const std::string& name);
@@ -137,9 +120,7 @@ union VideoCodecUnion {
   VideoCodecVP8 VP8;
   VideoCodecVP9 VP9;
   VideoCodecH264 H264;
-#ifndef DISABLE_H265
   VideoCodecH265 H265;
-#endif
 };
 
 enum class VideoCodecMode { kRealtimeVideo, kScreensharing };

call/call_config.h

@@ -38,7 +38,6 @@ struct CallConfig {
   RtpTransportConfig ExtractTransportConfig() const;
   ~CallConfig();
 
-  RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
   // Bitrate config used until valid bitrate estimates are calculated. Also
   // used to cap total bitrate used. This comes from the remote connection.
   BitrateConstraints bitrate_config;

call/rtp_payload_params.cc

@@ -19,9 +19,7 @@
 #include "absl/types/variant.h"
 #include "api/video/video_timing.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
-#ifndef DISABLE_H265
 #include "modules/video_coding/codecs/h265/include/h265_globals.h"
-#endif
 #include "modules/video_coding/codecs/interface/common_constants.h"
 #include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
@@ -317,32 +315,25 @@ void RtpPayloadParams::SetGeneric(const CodecSpecificInfo* codec_specific_info,
                                   int64_t frame_id,
                                   bool is_keyframe,
                                   RTPVideoHeader* rtp_video_header) {
-
-#ifdef WEBRTC_USE_H265
   if (codec_specific_info->codecType == webrtc::kVideoCodecH265 &&
       codec_specific_info->codecSpecific.H265.picture_id > 0) {
     // H265ToGeneric implementation. Only set it when picture id is valid.
     rtp_video_header->generic->frame_id =
         codec_specific_info->codecSpecific.H265.picture_id;
-    rtp_video_header->generic->discardable =
-        codec_specific_info->codecSpecific.H265.discardable;
-    rtp_video_header->generic->spatial_index = 0;  // Not enabled at present.
+    rtp_video_header->generic->spatial_index = 0;   // Not enabled at present.
     rtp_video_header->generic->temporal_index = 0;  // Not enabled at present.
     for (int dep_idx = 0; dep_idx < 5; dep_idx++) {
       if (codec_specific_info->codecSpecific.H265.dependencies[dep_idx] <= 0)
         break;
       rtp_video_header->generic->dependencies[dep_idx] =
           codec_specific_info->codecSpecific.H265.dependencies[dep_idx];
     }
-    // Not filling DTIs at present.
-    rtp_video_header->generic->discardable =
-        codec_specific_info->codecSpecific.H265.discardable;
     return;
   }
-#else
-   if (codec_specific_info && codec_specific_info->generic_frame_info &&
+  if (codec_specific_info && codec_specific_info->generic_frame_info &&
       !codec_specific_info->generic_frame_info->encoder_buffers.empty()) {
-    // If generic frame info is provided for other codecs, use generic frame info.
+    // If generic frame info is provided for other codecs, use generic frame
+    // info.
     if (is_keyframe) {
       // Key frame resets all chains it is in.
       chains_calculator_.Reset(
@@ -352,8 +343,6 @@ void RtpPayloadParams::SetGeneric(const CodecSpecificInfo* codec_specific_info,
         *codec_specific_info->generic_frame_info, frame_id);
     return;
   }
-#endif
-
 
   switch (rtp_video_header->codec) {
     case VideoCodecType::kVideoCodecGeneric:
@@ -380,10 +369,8 @@ void RtpPayloadParams::SetGeneric(const CodecSpecificInfo* codec_specific_info,
                       is_keyframe, rtp_video_header);
       }
       return;
-#ifndef DISABLE_H265
     // No further special handling for H.265
     case VideoCodecType::kVideoCodecH265:
-#endif
     case VideoCodecType::kVideoCodecMultiplex:
       return;
   }

call/rtp_transport_controller_send.cc

@@ -36,7 +36,6 @@ static const int64_t kRetransmitWindowSizeMs = 500;
 static const size_t kMaxOverheadBytes = 500;
 
 constexpr TimeDelta kPacerQueueUpdateInterval = TimeDelta::Millis(25);
-const char kLowLatencyStreaming[] = "OWT-LowLatencyMode";
 
 TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
                                          int max_bitrate_bps,
@@ -55,11 +54,6 @@ TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
   return msg;
 }
 
-bool LowLatencyStreamingEnabled() {
-  std::string trial = webrtc::field_trial::FindFullName(kLowLatencyStreaming);
-  return trial.find("Enabled") == 0;
-}
-
 TargetRateConstraints ConvertConstraints(const BitrateConstraints& contraints,
                                          Clock* clock) {
   return ConvertConstraints(contraints.min_bitrate_bps,

call/rtp_video_sender.cc

@@ -613,17 +613,13 @@ EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
       codec_specific_info->codecSpecific.H264.last_fragment_in_frame)
     absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header)
         .has_last_fragement = true;
-#ifndef DISABLE_H265
-  else if (codec_specific_info->codecType ==
-           kVideoCodecH265 && codec_specific_info->codecSpecific.H265
-               .last_fragment_in_frame)
+  else if (codec_specific_info->codecType == kVideoCodecH265 &&
+           codec_specific_info->codecSpecific.H265.last_fragment_in_frame)
     absl::get<RTPVideoHeaderH265>(rtp_video_header.video_type_header)
         .has_last_fragement = true;
-#endif
   bool send_result = rtp_streams_[stream_index].sender_video->SendEncodedImage(
       rtp_config_.payload_type, codec_type_, rtp_timestamp, encoded_image,
-      rtp_video_header,
-      expected_retransmission_time_ms);
+      rtp_video_header, expected_retransmission_time_ms);
   if (frame_count_observer_) {
     FrameCounts& counts = frame_counts_[stream_index];
     if (encoded_image._frameType == VideoFrameType::kVideoFrameKey) {

media/BUILD.gn

@@ -357,7 +357,7 @@ rtc_library("rtc_audio_video") {
   public_configs = []
   if (!build_with_chromium) {
     public_configs += [ ":rtc_media_defines_config" ]
-    deps += [ "../modules/video_capture:video_capture" ]
+    deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   if (rtc_enable_protobuf) {
     deps += [

media/base/media_constants.cc

@@ -105,9 +105,7 @@ const char kVp8CodecName[] = "VP8";
 const char kVp9CodecName[] = "VP9";
 const char kAv1CodecName[] = "AV1";
 const char kH264CodecName[] = "H264";
-#ifdef WEBRTC_USE_H265
 const char kH265CodecName[] = "H265";
-#endif
 
 // RFC 6184 RTP Payload Format for H.264 video
 const char kH264FmtpProfileLevelId[] = "profile-level-id";
@@ -122,7 +120,6 @@ const char kH265FmtpProfileSpace[] = "profile-space";
 const char kH265FmtpProfileId[] = "profile-id";
 const char kH265FmtpTierFlag[] = "tier-flag";
 const char kH265FmtpLevelId[] = "level-id";
-#endif
 
 const char kVP9ProfileId[] = "profile-id";
 

media/base/media_constants.h

@@ -127,9 +127,7 @@ RTC_EXPORT extern const char kVp8CodecName[];
 RTC_EXPORT extern const char kVp9CodecName[];
 RTC_EXPORT extern const char kAv1CodecName[];
 RTC_EXPORT extern const char kH264CodecName[];
-#ifdef WEBRTC_USE_H265
 RTC_EXPORT extern const char kH265CodecName[];
-#endif
 
 // RFC 6184 RTP Payload Format for H.264 video
 RTC_EXPORT extern const char kH264FmtpProfileLevelId[];

media/engine/encoder_simulcast_proxy.h

@@ -25,7 +25,6 @@
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "modules/video_coding/include/video_codec_interface.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -58,7 +57,6 @@ class RTC_EXPORT EncoderSimulcastProxy : public VideoEncoder {
   SdpVideoFormat video_format_;
   std::unique_ptr<VideoEncoder> encoder_;
   EncodedImageCallback* callback_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(EncoderSimulcastProxy);
 };
 
 }  // namespace webrtc

media/engine/webrtc_video_engine.cc

@@ -40,8 +40,7 @@
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
-
-static const int kRtpBufferSizeIncreased = 4 * 1024 * 1024;
+#include "system_wrappers/include/field_trial.h"
 
 namespace cricket {
 
@@ -68,15 +67,6 @@ const char* StreamTypeToString(
   }
   return nullptr;
 }
-// If this field trial is enabled, the "flexfec-03" codec will be advertised
-// as being supported. This means that "flexfec-03" will appear in the default
-// SDP offer, and we therefore need to be ready to receive FlexFEC packets from
-// the remote. It also means that FlexFEC SSRCs will be generated by
-// MediaSession and added as "a=ssrc:" and "a=ssrc-group:" lines in the local
-// SDP.
-bool IsFlexfecAdvertisedFieldTrialEnabled() {
-  return webrtc::field_trial::IsEnabled("OWT-FlexFEC");
-}
 
 bool IsEnabled(const webrtc::FieldTrialsView& trials, absl::string_view name) {
   return absl::StartsWith(trials.Lookup(name), "Enabled");
@@ -102,11 +92,10 @@ void AddDefaultFeedbackParams(VideoCodec* codec,
   codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
   codec->AddFeedbackParam(
       FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamRpsi));
-  if ((codec->name == kVp8CodecName || codec->name == kH264CodecName
-       || codec->name == kH265CodecName
-      ) &&
+  if ((codec->name == kVp8CodecName || codec->name == kH264CodecName ||
+       codec->name == kH265CodecName) &&
       (webrtc::field_trial::IsEnabled("WebRTC-RtcpLossNotification") ||
-      (webrtc::field_trial::IsEnabled("OWT-LowLatencyMode")))) {
+       (webrtc::field_trial::IsEnabled("OWT-LowLatencyMode")))) {
     codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty));
   }
 }

media/engine/webrtc_voice_engine.cc

@@ -52,6 +52,7 @@
 #include "rtc_base/third_party/base64/base64.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/metrics.h"
+#include "system_wrappers/include/field_trial.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
@@ -403,8 +404,6 @@ void WebRtcVoiceEngine::Init() {
       options.auto_gain_control = false;
       options.noise_suppression = false;
       options.highpass_filter = false;
-      options.typing_detection = false;
-      options.residual_echo_detector = false;
     }
     ApplyOptions(options);
   }