Export symbols needed by the Chromium component build (part 1).

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: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}

api/audio/BUILD.gn

@@ -39,6 +39,9 @@ rtc_source_set("aec3_config") {
     "echo_canceller3_config.cc",
     "echo_canceller3_config.h",
   ]
+  deps = [
+    "../../rtc_base/system:rtc_export",
+  ]
 }
 
 rtc_source_set("aec3_factory") {
@@ -54,6 +57,7 @@ rtc_source_set("aec3_factory") {
     ":echo_control",
     "../../modules/audio_processing/aec3",
     "../../rtc_base:rtc_base_approved",
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
   ]
 }

api/audio/echo_canceller3_config.h

@@ -13,10 +13,12 @@
 
 #include <stddef.h>  // size_t
 
+#include "rtc_base/system/rtc_export.h"
+
 namespace webrtc {
 
 // Configuration struct for EchoCanceller3
-struct EchoCanceller3Config {
+struct RTC_EXPORT EchoCanceller3Config {
   EchoCanceller3Config();
   EchoCanceller3Config(const EchoCanceller3Config& e);
   struct Delay {

api/audio/echo_canceller3_factory.h

@@ -15,10 +15,11 @@
 
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/echo_control.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
-class EchoCanceller3Factory : public EchoControlFactory {
+class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory {
  public:
   // Factory producing EchoCanceller3 instances with the default configuration.
   EchoCanceller3Factory();

api/audio_codecs/L16/BUILD.gn

@@ -25,6 +25,7 @@ rtc_static_library("audio_encoder_L16") {
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

api/audio_codecs/L16/audio_encoder_L16.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -25,7 +26,7 @@ namespace webrtc {
 // CreateAudioEncoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderL16 {
+struct RTC_EXPORT AudioEncoderL16 {
   struct Config {
     bool IsOk() const {
       return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||

api/audio_codecs/g711/BUILD.gn

@@ -25,6 +25,7 @@ rtc_static_library("audio_encoder_g711") {
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

api/audio_codecs/g711/audio_encoder_g711.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -25,7 +26,7 @@ namespace webrtc {
 // CreateAudioEncoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderG711 {
+struct RTC_EXPORT AudioEncoderG711 {
   struct Config {
     enum class Type { kPcmU, kPcmA };
     bool IsOk() const {

api/audio_codecs/g722/BUILD.gn

@@ -33,6 +33,7 @@ rtc_static_library("audio_encoder_g722") {
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

api/audio_codecs/g722/audio_encoder_g722.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -26,7 +27,7 @@ namespace webrtc {
 // CreateAudioEncoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderG722 {
+struct RTC_EXPORT AudioEncoderG722 {
   using Config = AudioEncoderG722Config;
   static absl::optional<AudioEncoderG722Config> SdpToConfig(
       const SdpAudioFormat& audio_format);

api/audio_codecs/isac/BUILD.gn

@@ -131,6 +131,7 @@ rtc_static_library("audio_decoder_isac_float") {
     "../../..:webrtc_common",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

api/audio_codecs/isac/audio_decoder_isac_float.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -25,7 +26,7 @@ namespace webrtc {
 // parameter to CreateAudioDecoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioDecoderIsacFloat {
+struct RTC_EXPORT AudioDecoderIsacFloat {
   struct Config {
     bool IsOk() const {
       return sample_rate_hz == 16000 || sample_rate_hz == 32000;

api/audio_codecs/opus/BUILD.gn

@@ -44,6 +44,7 @@ rtc_source_set("audio_encoder_opus") {
     "..:audio_codecs_api",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -60,6 +61,7 @@ rtc_static_library("audio_decoder_opus") {
     "../../..:webrtc_common",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

api/audio_codecs/opus/audio_decoder_opus.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -25,7 +26,7 @@ namespace webrtc {
 // CreateAudioDecoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioDecoderOpus {
+struct RTC_EXPORT AudioDecoderOpus {
   struct Config {
     int num_channels;
   };

api/audio_codecs/opus/audio_encoder_opus.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -26,7 +27,7 @@ namespace webrtc {
 // CreateAudioEncoderFactory<...>().
 //
 // NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderOpus {
+struct RTC_EXPORT AudioEncoderOpus {
   using Config = AudioEncoderOpusConfig;
   static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
       const SdpAudioFormat& audio_format);

media/BUILD.gn

@@ -136,6 +136,7 @@ rtc_static_library("rtc_media_base") {
     "../modules/audio_processing:audio_processing_statistics",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
+    "../rtc_base/system:rtc_export",
     "../rtc_base/third_party/sigslot",
     "//third_party/abseil-cpp/absl/types:optional",
   ]

media/base/rtputils.h

@@ -12,6 +12,7 @@
 #define MEDIA_BASE_RTPUTILS_H_
 
 #include "rtc_base/byteorder.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace rtc {
 struct PacketTimeUpdateParams;
@@ -68,7 +69,7 @@ bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
 const char* RtpRtcpStringLiteral(bool rtcp);
 
 // Verifies that a packet has a valid RTP header.
-bool ValidateRtpHeader(const uint8_t* rtp,
+bool RTC_EXPORT ValidateRtpHeader(const uint8_t* rtp,
                                   size_t length,
                                   size_t* header_length);
 
@@ -80,7 +81,8 @@ bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
 
 // Applies specified |options| to the packet. It updates the absolute send time
 // extension header if it is present present then updates HMAC.
-bool ApplyPacketOptions(uint8_t* data,
+bool RTC_EXPORT
+ApplyPacketOptions(uint8_t* data,
                    size_t length,
                    const rtc::PacketTimeUpdateParams& packet_time_params,
                    uint64_t time_us);

media/base/turnutils.h

@@ -14,13 +14,15 @@
 #include <cstddef>
 #include <cstdint>
 
+#include "rtc_base/system/rtc_export.h"
+
 namespace cricket {
 
 struct PacketOptions;
 
 // Finds data location within a TURN Channel Message or TURN Send Indication
 // message.
-bool UnwrapTurnPacket(const uint8_t* packet,
+bool RTC_EXPORT UnwrapTurnPacket(const uint8_t* packet,
                                  size_t packet_size,
                                  size_t* content_position,
                                  size_t* content_size);

modules/audio_processing/BUILD.gn

@@ -119,6 +119,7 @@ rtc_static_library("audio_processing") {
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
     "../../rtc_base/system:arch",
+    "../../rtc_base/system:rtc_export",
     "../../system_wrappers:cpu_features_api",
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
@@ -166,6 +167,7 @@ rtc_source_set("audio_processing_statistics") {
     "include/audio_processing_statistics.h",
   ]
   deps = [
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }

modules/audio_processing/aec_dump/BUILD.gn

@@ -17,6 +17,7 @@ rtc_source_set("aec_dump") {
   deps = [
     "../",
     "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base/system:rtc_export",
   ]
 }
 

modules/audio_processing/aec_dump/aec_dump_factory.h

@@ -16,6 +16,7 @@
 
 #include "modules/audio_processing/include/aec_dump.h"
 #include "rtc_base/platform_file.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace rtc {
 class TaskQueue;
@@ -23,7 +24,7 @@ class TaskQueue;
 
 namespace webrtc {
 
-class AecDumpFactory {
+class RTC_EXPORT AecDumpFactory {
  public:
   // The |worker_queue| may not be null and must outlive the created
   // AecDump instance. |max_log_size_bytes == -1| means the log size

modules/audio_processing/include/audio_processing.h

@@ -34,6 +34,7 @@
 #include "rtc_base/platform_file.h"
 #include "rtc_base/refcount.h"
 #include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -554,7 +555,7 @@ class AudioProcessing : public rtc::RefCountInterface {
     float minimum_ = 0.0f;  // Long-term minimum.
   };
 
-  struct AudioProcessingStatistics {
+  struct RTC_EXPORT AudioProcessingStatistics {
     AudioProcessingStatistics();
     AudioProcessingStatistics(const AudioProcessingStatistics& other);
     ~AudioProcessingStatistics();
@@ -653,7 +654,7 @@ class AudioProcessing : public rtc::RefCountInterface {
   static const int kChunkSizeMs = 10;
 };
 
-class AudioProcessingBuilder {
+class RTC_EXPORT AudioProcessingBuilder {
  public:
   AudioProcessingBuilder();
   ~AudioProcessingBuilder();

modules/audio_processing/include/audio_processing_statistics.h

@@ -12,11 +12,12 @@
 #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
 
 #include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 // This version of the stats uses Optionals, it will replace the regular
 // AudioProcessingStatistics struct.
-struct AudioProcessingStats {
+struct RTC_EXPORT AudioProcessingStats {
   AudioProcessingStats();
   AudioProcessingStats(const AudioProcessingStats& other);
   ~AudioProcessingStats();

modules/audio_processing/include/config.h

@@ -14,6 +14,7 @@
 #include <map>
 
 #include "rtc_base/constructormagic.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -57,7 +58,7 @@ enum class ConfigOptionID {
 //    config.Set<Algo1_CostFunction>(new SqrCost());
 //
 // Note: This class is thread-compatible (like STL containers).
-class Config {
+class RTC_EXPORT Config {
  public:
   // Returns the option if set or a default constructed one.
   // Callers that access options too often are encouraged to cache the result.

modules/audio_processing/typing_detection.h

@@ -11,9 +11,11 @@
 #ifndef MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 #define MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 
+#include "rtc_base/system/rtc_export.h"
+
 namespace webrtc {
 
-class TypingDetection {
+class RTC_EXPORT TypingDetection {
  public:
   TypingDetection();
   virtual ~TypingDetection();

modules/desktop_capture/BUILD.gn

@@ -30,6 +30,7 @@ rtc_static_library("primitives") {
   deps = [
     "../..:webrtc_common",
     "../../rtc_base:checks",
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
   ]
 
@@ -376,6 +377,7 @@ rtc_static_library("desktop_capture_generic") {
     "../../rtc_base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
     "../../rtc_base/synchronization:rw_lock_wrapper",
     "../../rtc_base/system:arch",
+    "../../rtc_base/system:rtc_export",
     "../../system_wrappers",
     "../../system_wrappers:cpu_features_api",
     "../../system_wrappers:metrics",

modules/desktop_capture/cropped_desktop_frame.h

@@ -12,6 +12,7 @@
 #define MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
 
 #include "modules/desktop_capture/desktop_frame.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -20,8 +21,8 @@ namespace webrtc {
 // |frame| should not be nullptr. |rect| is in |frame| coordinate, i.e.
 // |frame|->top_left() does not impact the area of |rect|.
 // Returns nullptr frame if |rect| is not contained by the bounds of |frame|.
-std::unique_ptr<DesktopFrame> CreateCroppedDesktopFrame(
-    std::unique_ptr<DesktopFrame> frame,
+std::unique_ptr<DesktopFrame> RTC_EXPORT
+CreateCroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame,
                           const DesktopRect& rect);
 
 }  // namespace webrtc

modules/desktop_capture/cropping_window_capturer.h

@@ -15,13 +15,14 @@
 
 #include "modules/desktop_capture/desktop_capture_options.h"
 #include "modules/desktop_capture/desktop_capturer.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 // WindowCapturer implementation that uses a screen capturer to capture the
 // whole screen and crops the video frame to the window area when the captured
 // window is on top.
-class CroppingWindowCapturer : public DesktopCapturer,
+class RTC_EXPORT CroppingWindowCapturer : public DesktopCapturer,
                                           public DesktopCapturer::Callback {
  public:
   static std::unique_ptr<DesktopCapturer> CreateCapturer(

modules/desktop_capture/desktop_and_cursor_composer.h

@@ -17,12 +17,14 @@
 #include "modules/desktop_capture/desktop_capturer.h"
 #include "modules/desktop_capture/mouse_cursor_monitor.h"
 #include "rtc_base/constructormagic.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 // A wrapper for DesktopCapturer that also captures mouse using specified
 // MouseCursorMonitor and renders it on the generated streams.
-class DesktopAndCursorComposer : public DesktopCapturer,
+class RTC_EXPORT DesktopAndCursorComposer
+    : public DesktopCapturer,
       public DesktopCapturer::Callback,
       public MouseCursorMonitor::Callback {
  public:

modules/desktop_capture/desktop_capture_options.h

@@ -12,6 +12,7 @@
 
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/system/rtc_export.h"
 
 #if defined(USE_X11)
 #include "modules/desktop_capture/x11/shared_x_display.h"
@@ -26,7 +27,7 @@ namespace webrtc {
 
 // An object that stores initialization parameters for screen and window
 // capturers.
-class DesktopCaptureOptions {
+class RTC_EXPORT DesktopCaptureOptions {
  public:
   // Returns instance of DesktopCaptureOptions with default parameters. On Linux
   // also initializes X window connection. x_display() will be set to null if

modules/desktop_capture/desktop_capturer.h

@@ -22,6 +22,7 @@
 #include "modules/desktop_capture/desktop_capture_types.h"
 #include "modules/desktop_capture/desktop_frame.h"
 #include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -29,7 +30,7 @@ class DesktopCaptureOptions;
 class DesktopFrame;
 
 // Abstract interface for screen and window capturers.
-class DesktopCapturer {
+class RTC_EXPORT DesktopCapturer {
  public:
   enum class Result {
     // The frame was captured successfully.

modules/desktop_capture/desktop_frame.h

@@ -18,13 +18,14 @@
 #include "modules/desktop_capture/desktop_region.h"
 #include "modules/desktop_capture/shared_memory.h"
 #include "rtc_base/constructormagic.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 const int kStandardDPI = 96;
 
 // DesktopFrame represents a video frame captured from the screen.
-class DesktopFrame {
+class RTC_EXPORT DesktopFrame {
  public:
   // DesktopFrame objects always hold RGBA data.
   static const int kBytesPerPixel = 4;
@@ -135,7 +136,7 @@ class DesktopFrame {
 };
 
 // A DesktopFrame that stores data in the heap.
-class BasicDesktopFrame : public DesktopFrame {
+class RTC_EXPORT BasicDesktopFrame : public DesktopFrame {
  public:
   explicit BasicDesktopFrame(DesktopSize size);
 

modules/desktop_capture/fake_desktop_capturer.h

@@ -17,6 +17,7 @@
 #include "modules/desktop_capture/desktop_capturer.h"
 #include "modules/desktop_capture/desktop_frame_generator.h"
 #include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
@@ -31,7 +32,7 @@ namespace webrtc {
 // Double buffering is guaranteed by the FrameGenerator. FrameGenerator
 // implements in desktop_frame_generator.h guarantee double buffering, they
 // creates a new instance of DesktopFrame each time.
-class FakeDesktopCapturer : public DesktopCapturer {
+class RTC_EXPORT FakeDesktopCapturer : public DesktopCapturer {
  public:
   FakeDesktopCapturer();
   ~FakeDesktopCapturer() override;

rtc_base/BUILD.gn

@@ -529,6 +529,7 @@ rtc_source_set("rtc_task_queue_api") {
   deps = [
     ":macromagic",
     ":ptr_util",
+    "system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
   ]
 }

rtc_base/task_queue.h

@@ -18,6 +18,7 @@
 #include "absl/memory/memory.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/system/rtc_export.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace rtc {
@@ -150,7 +151,7 @@ static std::unique_ptr<QueuedTask> NewClosure(Closure&& closure,
 // TaskQueue itself has been deleted or it may happen synchronously while the
 // TaskQueue instance is being deleted.  This may vary from one OS to the next
 // so assumptions about lifetimes of pending tasks should not be made.
-class RTC_LOCKABLE TaskQueue {
+class RTC_LOCKABLE RTC_EXPORT TaskQueue {
  public:
   // TaskQueue priority levels. On some platforms these will map to thread
   // priorities, on others such as Mac and iOS, GCD queue priorities.