Export symbols needed by the Chromium component build (part 2).
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).
Bug: webrtc:9419
Change-Id: I6f27003001548ea9d54412fdf62d5dd7a39cfd46
Reviewed-on: https://webrtc-review.googlesource.com/c/106022
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25187}diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
index e362067..a7d56d0 100644
--- a/api/audio_codecs/L16/BUILD.gn
+++ b/api/audio_codecs/L16/BUILD.gn
@@ -43,6 +43,7 @@
"../../..:webrtc_common",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
+ "../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
diff --git a/api/audio_codecs/L16/audio_decoder_L16.h b/api/audio_codecs/L16/audio_decoder_L16.h
index 184ec24..b38627d 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.h
+++ b/api/audio_codecs/L16/audio_decoder_L16.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 @@
// CreateAudioDecoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
-struct AudioDecoderL16 {
+struct RTC_EXPORT AudioDecoderL16 {
struct Config {
bool IsOk() const {
return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
index 860ba3d..ebd0b32 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -43,6 +43,7 @@
"../../..:webrtc_common",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
+ "../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
diff --git a/api/audio_codecs/g711/audio_decoder_g711.h b/api/audio_codecs/g711/audio_decoder_g711.h
index 8275a8c..0c67e5e 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.h
+++ b/api/audio_codecs/g711/audio_decoder_g711.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 @@
// CreateAudioDecoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
-struct AudioDecoderG711 {
+struct RTC_EXPORT AudioDecoderG711 {
struct Config {
enum class Type { kPcmU, kPcmA };
bool IsOk() const {
diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index d69596b..101c7a9 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -51,6 +51,7 @@
"../../..:webrtc_common",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
+ "../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
diff --git a/api/audio_codecs/g722/audio_decoder_g722.h b/api/audio_codecs/g722/audio_decoder_g722.h
index b7bb089..7601b3c 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.h
+++ b/api/audio_codecs/g722/audio_decoder_g722.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 @@
// CreateAudioDecoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
-struct AudioDecoderG722 {
+struct RTC_EXPORT AudioDecoderG722 {
struct Config {
bool IsOk() const { return num_channels == 1 || num_channels == 2; }
int num_channels;
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index 9c9bf33..af3cd7f 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -20,6 +20,7 @@
]
deps = [
"../../../rtc_base:rtc_base_approved",
+ "../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/types:optional",
]
defines = []
diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.h b/api/audio_codecs/opus/audio_encoder_opus_config.h
index c7067bb..98a6ef5 100644
--- a/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ b/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -16,11 +16,12 @@
#include <vector>
#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderOpusConfig {
+struct RTC_EXPORT AudioEncoderOpusConfig {
static constexpr int kDefaultFrameSizeMs = 20;
// Opus API allows a min bitrate of 500bps, but Opus documentation suggests