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}
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@ -25,6 +25,7 @@ rtc_static_library("audio_encoder_L16") {
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"../../../modules/audio_coding:pcm16b",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base:safe_minmax",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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@ -18,6 +18,7 @@
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -25,7 +26,7 @@ namespace webrtc {
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// CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderL16 {
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struct RTC_EXPORT AudioEncoderL16 {
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struct Config {
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bool IsOk() const {
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return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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@ -25,6 +25,7 @@ rtc_static_library("audio_encoder_g711") {
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"../../../modules/audio_coding:g711",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base:safe_minmax",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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@ -18,6 +18,7 @@
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -25,7 +26,7 @@ namespace webrtc {
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// CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderG711 {
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struct RTC_EXPORT AudioEncoderG711 {
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struct Config {
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enum class Type { kPcmU, kPcmA };
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bool IsOk() const {
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@ -33,6 +33,7 @@ rtc_static_library("audio_encoder_g722") {
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"../../../modules/audio_coding:g722",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base:safe_minmax",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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@ -19,6 +19,7 @@
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -26,7 +27,7 @@ namespace webrtc {
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// CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderG722 {
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struct RTC_EXPORT AudioEncoderG722 {
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using Config = AudioEncoderG722Config;
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static absl::optional<AudioEncoderG722Config> SdpToConfig(
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const SdpAudioFormat& audio_format);
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@ -131,6 +131,7 @@ rtc_static_library("audio_decoder_isac_float") {
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"../../..:webrtc_common",
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"../../../modules/audio_coding:isac",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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@ -18,6 +18,7 @@
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -25,7 +26,7 @@ namespace webrtc {
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// parameter to CreateAudioDecoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioDecoderIsacFloat {
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struct RTC_EXPORT AudioDecoderIsacFloat {
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struct Config {
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bool IsOk() const {
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return sample_rate_hz == 16000 || sample_rate_hz == 32000;
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@ -44,6 +44,7 @@ rtc_source_set("audio_encoder_opus") {
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_opus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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@ -60,6 +61,7 @@ rtc_static_library("audio_decoder_opus") {
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"../../..:webrtc_common",
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"../../../modules/audio_coding:webrtc_opus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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@ -18,6 +18,7 @@
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -25,7 +26,7 @@ namespace webrtc {
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// CreateAudioDecoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioDecoderOpus {
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struct RTC_EXPORT AudioDecoderOpus {
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struct Config {
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int num_channels;
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};
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@ -19,6 +19,7 @@
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -26,7 +27,7 @@ namespace webrtc {
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// CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderOpus {
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struct RTC_EXPORT AudioEncoderOpus {
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using Config = AudioEncoderOpusConfig;
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static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
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const SdpAudioFormat& audio_format);
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