Export symbols needed by the Chromium component build (part 8).

This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}

api/BUILD.gn

@@ -466,6 +466,7 @@ rtc_source_set("ice_transport_factory") {
     ":scoped_refptr",
     "../p2p:rtc_p2p",
     "../rtc_base",
+    "../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
   ]
 }

api/ice_transport_factory.h

@@ -13,6 +13,7 @@
 
 #include "api/ice_transport_interface.h"
 #include "api/scoped_refptr.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace cricket {
 class PortAllocator;
@@ -25,7 +26,7 @@ namespace webrtc {
 // The returned object must be accessed and destroyed on the thread that
 // created it.
 // The PortAllocator must outlive the created IceTransportInterface object.
-rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
+RTC_EXPORT rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
     cricket::PortAllocator* port_allocator);
 
 }  // namespace webrtc

api/jsep_ice_candidate.h

@@ -21,11 +21,12 @@
 #include "api/candidate.h"
 #include "api/jsep.h"
 #include "rtc_base/constructor_magic.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 // Implementation of IceCandidateInterface.
-class JsepIceCandidate : public IceCandidateInterface {
+class RTC_EXPORT JsepIceCandidate : public IceCandidateInterface {
  public:
   JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index);
   JsepIceCandidate(const std::string& sdp_mid,

api/media_stream_interface.h

@@ -30,6 +30,7 @@
 #include "api/video/video_source_interface.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
 #include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -52,8 +53,8 @@ class NotifierInterface {
 
 // Base class for sources. A MediaStreamTrack has an underlying source that
 // provides media. A source can be shared by multiple tracks.
-class MediaSourceInterface : public rtc::RefCountInterface,
-                             public NotifierInterface {
+class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
+                                        public NotifierInterface {
  public:
   enum SourceState { kInitializing, kLive, kEnded, kMuted };
 
@@ -74,8 +75,8 @@ class MediaSourceInterface : public rtc::RefCountInterface,
 
 // C++ version of MediaStreamTrack.
 // See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
-class MediaStreamTrackInterface : public rtc::RefCountInterface,
-                                  public NotifierInterface {
+class RTC_EXPORT MediaStreamTrackInterface : public rtc::RefCountInterface,
+                                             public NotifierInterface {
  public:
   enum TrackState {
     kLive,
@@ -190,7 +191,7 @@ class AudioTrackSinkInterface {
 
 // AudioSourceInterface is a reference counted source used for AudioTracks.
 // The same source can be used by multiple AudioTracks.
-class AudioSourceInterface : public MediaSourceInterface {
+class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
  public:
   class AudioObserver {
    public:

api/media_types.h

@@ -13,6 +13,8 @@
 
 #include <string>
 
+#include "rtc_base/system/rtc_export.h"
+
 // The cricket and webrtc have separate definitions for what a media type is.
 // They're not compatible. Watch out for this.
 
@@ -20,7 +22,7 @@ namespace cricket {
 
 enum MediaType { MEDIA_TYPE_AUDIO, MEDIA_TYPE_VIDEO, MEDIA_TYPE_DATA };
 
-std::string MediaTypeToString(MediaType type);
+RTC_EXPORT std::string MediaTypeToString(MediaType type);
 // Aborts on invalid string. Only expected to be used on strings that are
 // guaranteed to be valid, such as MediaStreamTrackInterface::kind().
 MediaType MediaTypeFromString(const std::string& type_str);

api/peer_connection_interface.h

@@ -151,7 +151,7 @@ class StatsObserver : public rtc::RefCountInterface {
 
 enum class SdpSemantics { kPlanB, kUnifiedPlan };
 
-class PeerConnectionInterface : public rtc::RefCountInterface {
+class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
  public:
   // See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
   enum SignalingState {

api/rtc_error.h

@@ -20,6 +20,7 @@
 #include "absl/strings/string_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -80,7 +81,7 @@ enum class RTCErrorType {
 //
 // Doesn't contain anything beyond a type and message now, but will in the
 // future as more errors are implemented.
-class RTCError {
+class RTC_EXPORT RTCError {
  public:
   // Constructors.
 

api/rtp_parameters.h

@@ -87,7 +87,7 @@ enum class DegradationPreference {
   BALANCED,
 };
 
-extern const double kDefaultBitratePriority;
+RTC_EXPORT extern const double kDefaultBitratePriority;
 
 struct RtcpFeedback {
   RtcpFeedbackType type = RtcpFeedbackType::CCM;
@@ -362,7 +362,7 @@ struct RtpRtxParameters {
   bool operator!=(const RtpRtxParameters& o) const { return !(*this == o); }
 };
 
-struct RtpEncodingParameters {
+struct RTC_EXPORT RtpEncodingParameters {
   RtpEncodingParameters();
   RtpEncodingParameters(const RtpEncodingParameters&);
   ~RtpEncodingParameters();
@@ -561,7 +561,7 @@ struct RtpCodecParameters {
 // RtpCapabilities is used to represent the static capabilities of an
 // endpoint. An application can use these capabilities to construct an
 // RtpParameters.
-struct RtpCapabilities {
+struct RTC_EXPORT RtpCapabilities {
   RtpCapabilities();
   ~RtpCapabilities();
 

api/rtp_transceiver_interface.h

@@ -22,6 +22,7 @@
 #include "api/rtp_sender_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -36,7 +37,7 @@ enum class RtpTransceiverDirection {
 // Structure for initializing an RtpTransceiver in a call to
 // PeerConnectionInterface::AddTransceiver.
 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
-struct RtpTransceiverInit final {
+struct RTC_EXPORT RtpTransceiverInit final {
   RtpTransceiverInit();
   RtpTransceiverInit(const RtpTransceiverInit&);
   ~RtpTransceiverInit();

api/sctp_transport_interface.h

@@ -33,7 +33,7 @@ enum class SctpTransportState {
 // SctpTransport.
 // It reflects the readonly attributes of the object in the specification.
 // http://w3c.github.io/webrtc-pc/#rtcsctptransport-interface
-class SctpTransportInformation {
+class RTC_EXPORT SctpTransportInformation {
  public:
   explicit SctpTransportInformation(SctpTransportState state);
   SctpTransportInformation(

api/transport/BUILD.gn

@@ -15,6 +15,7 @@ rtc_source_set("bitrate_settings") {
     "bitrate_settings.h",
   ]
   deps = [
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }

api/transport/bitrate_settings.h

@@ -12,6 +12,7 @@
 #define API_TRANSPORT_BITRATE_SETTINGS_H_
 
 #include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -20,7 +21,7 @@ namespace webrtc {
 // estimator, and for initial configuration of the encoder. We may
 // want to create separate apis for those, and use a smaller struct
 // with only the min and max constraints.
-struct BitrateSettings {
+struct RTC_EXPORT BitrateSettings {
   BitrateSettings();
   ~BitrateSettings();
   BitrateSettings(const BitrateSettings&);

api/video/BUILD.gn

@@ -130,6 +130,7 @@ rtc_source_set("video_bitrate_allocation") {
     "../../rtc_base:checks",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }

api/video/video_bitrate_allocation.h

@@ -19,6 +19,7 @@
 
 #include "absl/types/optional.h"
 #include "api/video/video_codec_constants.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -27,7 +28,7 @@ namespace webrtc {
 // layers are dependent or not, it is up to the user to aggregate.
 // For each index, the bitrate can also both set and unset. This is used with a
 // set bps = 0 to signal an explicit "turn off" signal.
-class VideoBitrateAllocation {
+class RTC_EXPORT VideoBitrateAllocation {
  public:
   static constexpr uint32_t kMaxBitrateBps =
       std::numeric_limits<uint32_t>::max();

api/video_codecs/BUILD.gn

@@ -143,6 +143,7 @@ rtc_static_library("rtc_software_fallback_wrappers") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:fallthrough",
+    "../../rtc_base/system:rtc_export",
     "../../system_wrappers:field_trial",
     "../video:encoded_image",
     "../video:video_bitrate_allocation",

api/video_codecs/sdp_video_format.h

@@ -32,8 +32,10 @@ struct RTC_EXPORT SdpVideoFormat {
 
   ~SdpVideoFormat();
 
-  friend bool operator==(const SdpVideoFormat& a, const SdpVideoFormat& b);
-  friend bool operator!=(const SdpVideoFormat& a, const SdpVideoFormat& b) {
+  friend RTC_EXPORT bool operator==(const SdpVideoFormat& a,
+                                    const SdpVideoFormat& b);
+  friend RTC_EXPORT bool operator!=(const SdpVideoFormat& a,
+                                    const SdpVideoFormat& b) {
     return !(a == b);
   }
 

api/video_codecs/video_decoder_factory.h

@@ -15,6 +15,8 @@
 #include <string>
 #include <vector>
 
+#include "rtc_base/system/rtc_export.h"
+
 namespace webrtc {
 
 class VideoDecoder;
@@ -22,7 +24,7 @@ struct SdpVideoFormat;
 
 // A factory that creates VideoDecoders.
 // NOTE: This class is still under development and may change without notice.
-class VideoDecoderFactory {
+class RTC_EXPORT VideoDecoderFactory {
  public:
   // Returns a list of supported video formats in order of preference, to use
   // for signaling etc.

api/video_codecs/video_decoder_software_fallback_wrapper.h

@@ -14,13 +14,15 @@
 #include <memory>
 
 #include "api/video_codecs/video_decoder.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 // Used to wrap external VideoDecoders to provide a fallback option on
 // software decoding when a hardware decoder fails to decode a stream due to
 // hardware restrictions, such as max resolution.
-std::unique_ptr<VideoDecoder> CreateVideoDecoderSoftwareFallbackWrapper(
+RTC_EXPORT std::unique_ptr<VideoDecoder>
+CreateVideoDecoderSoftwareFallbackWrapper(
     std::unique_ptr<VideoDecoder> sw_fallback_decoder,
     std::unique_ptr<VideoDecoder> hw_decoder);
 

api/video_codecs/video_encoder_software_fallback_wrapper.h

@@ -14,13 +14,15 @@
 #include <memory>
 
 #include "api/video_codecs/video_encoder.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 // Used to wrap external VideoEncoders to provide a fallback option on
 // software encoding when a hardware encoder fails to encode a stream due to
 // hardware restrictions, such as max resolution.
-std::unique_ptr<VideoEncoder> CreateVideoEncoderSoftwareFallbackWrapper(
+RTC_EXPORT std::unique_ptr<VideoEncoder>
+CreateVideoEncoderSoftwareFallbackWrapper(
     std::unique_ptr<VideoEncoder> sw_fallback_encoder,
     std::unique_ptr<VideoEncoder> hw_encoder);