base->rtc_base: Update .c, .mm and .java files.

TBR=kwiberg@webrtc.org BUG=webrtc:7634 Review-Url: https://codereview.webrtc.org/2974613003 Cr-Commit-Position: refs/heads/master@{#18926}
2017-07-07 03:09:51 -07:00
parent 0e9bd8e15b
commit eaaae9e91b
92 changed files with 137 additions and 138 deletions
--- a/webrtc/common_audio/signal_processing/auto_correlation.c
+++ b/webrtc/common_audio/signal_processing/auto_correlation.c
@ -10,7 +10,7 @@

 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
                                 size_t in_vector_length,
--- a/webrtc/common_audio/signal_processing/downsample_fast.c
+++ b/webrtc/common_audio/signal_processing/downsample_fast.c
@ -10,8 +10,8 @@

 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

-#include "webrtc/base/checks.h"
-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/sanitizer.h"

 // TODO(Bjornv): Change the function parameter order to WebRTC code style.
 // C version of WebRtcSpl_DownsampleFast() for generic platforms.
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // TODO(bjornv): Change the return type to report errors.
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
--- a/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
@ -17,7 +17,7 @@

 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/sanitizer.h"

 void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
                               int16_t* out_ptr,
--- a/webrtc/common_audio/signal_processing/min_max_operations.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations.c
@ -26,7 +26,7 @@

 #include <stdlib.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // TODO(bjorn/kma): Consolidate function pairs (e.g. combine
--- a/webrtc/common_audio/signal_processing/min_max_operations_mips.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations_mips.c
@ -16,7 +16,7 @@
 *
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Maximum absolute value of word16 vector.
--- a/webrtc/common_audio/signal_processing/min_max_operations_neon.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations_neon.c
@ -11,7 +11,7 @@
 #include <arm_neon.h>
 #include <stdlib.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Maximum absolute value of word16 vector. C version for generic platforms.
--- a/webrtc/common_audio/signal_processing/spl_sqrt.c
+++ b/webrtc/common_audio/signal_processing/spl_sqrt.c
@ -15,7 +15,7 @@
 *
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int32_t WebRtcSpl_SqrtLocal(int32_t in);
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@ -13,7 +13,7 @@
 *
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Maximum number of samples in a low/high-band frame.
--- a/webrtc/common_audio/vad/vad_core.c
+++ b/webrtc/common_audio/vad/vad_core.c
@ -10,7 +10,7 @@

 #include "webrtc/common_audio/vad/vad_core.h"

-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/sanitizer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/vad_filterbank.h"
 #include "webrtc/common_audio/vad/vad_gmm.h"
--- a/webrtc/common_audio/vad/vad_filterbank.c
+++ b/webrtc/common_audio/vad/vad_filterbank.c
@ -10,7 +10,7 @@

 #include "webrtc/common_audio/vad/vad_filterbank.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/typedefs.h"

--- a/webrtc/common_audio/vad/vad_sp.c
+++ b/webrtc/common_audio/vad/vad_sp.c
@ -10,7 +10,7 @@

 #include "webrtc/common_audio/vad/vad_sp.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/vad_core.h"
 #include "webrtc/typedefs.h"
--- a/webrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm
+++ b/webrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm
@ -13,7 +13,7 @@
 #import <QuartzCore/CoreAnimation.h>
 #import <XCTest/XCTest.h>

-#include "webrtc/base/ssladapter.h"
+#include "webrtc/rtc_base/ssladapter.h"

 #import "WebRTC/RTCMediaConstraints.h"
 #import "WebRTC/RTCPeerConnectionFactory.h"
--- a/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@ -18,7 +18,7 @@

 #include "defines.h"
 #include "constants.h"
-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/sanitizer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 /*----------------------------------------------------------------*
--- a/webrtc/modules/audio_coding/codecs/ilbc/encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
@ -29,7 +29,7 @@
 #include "index_conv_enc.h"
 #include "pack_bits.h"
 #include "hp_input.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 #ifdef SPLIT_10MS
 #include "unpack_bits.h"
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@ -22,7 +22,7 @@
 #include "encode.h"
 #include "init_decode.h"
 #include "decode.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include <stdlib.h>

 int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@ -21,7 +21,7 @@
 #include "bandwidth_estimator.h"

 #include "settings.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 /* array of quantization levels for bottle neck info; Matlab code: */
 /* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@ -15,7 +15,7 @@
 *
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"

 #include <stdio.h>
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@ -26,7 +26,7 @@
 #include "lpc_tables.h"
 #include "settings.h"
 #include "signal_processing_library.h"
-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/sanitizer.h"

 /*
 * Eenumerations for arguments to functions WebRtcIsacfix_MatrixProduct1()
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
@ -20,7 +20,7 @@
 #include <stddef.h>

 #include "signal_processing_library.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
                                      const int32_t matrix1[],
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
@ -23,7 +23,7 @@
 #include "codec.h"
 #include "filterbank_tables.h"
 #include "settings.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 // Declare a function pointer.
 AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16;
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
@ -15,7 +15,7 @@

 #include <arm_neon.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
    int16_t* data_ch1,  // Input and output in channel 1, in Q0
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"

 // Autocorrelation function in fixed point.
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
@ -10,7 +10,7 @@

 #include <arm_neon.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"

 // Autocorrelation function in fixed point.
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@ -19,7 +19,7 @@

 #include <stdlib.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
@ -17,7 +17,7 @@

 #include "codec.h"
 #include "settings.h"
-#include "webrtc/base/sanitizer.h"
+#include "webrtc/rtc_base/sanitizer.h"

 #define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32)                  \
  ((int32_t)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
@ -10,7 +10,7 @@

 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/base/compile_assert_c.h"
+#include "webrtc/rtc_base/compile_assert_c.h"

 /* log2[0.2, 0.5, 0.98] in Q8 */
 static const int16_t kLogLagWinQ8[3] = {
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
@ -15,7 +15,7 @@
 #endif

 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/base/compile_assert_c.h"
+#include "webrtc/rtc_base/compile_assert_c.h"

 extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);

--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
@ -10,7 +10,7 @@

 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/base/compile_assert_c.h"
+#include "webrtc/rtc_base/compile_assert_c.h"

 extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);

--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
@ -13,7 +13,7 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
-#include "webrtc/base/compile_assert_c.h"
+#include "webrtc/rtc_base/compile_assert_c.h"

 // Number of segments in a pitch subframe.
 static const int kSegments = 5;
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@ -19,7 +19,7 @@
 #include "bandwidth_estimator.h"
 #include "settings.h"
 #include "isac.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 #include <math.h>
 #include <string.h>
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@ -22,7 +22,7 @@
 #include <stdlib.h>
 #include <string.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h"
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@ -16,7 +16,7 @@

 #include "os_specific_inline.h"

-#include "webrtc/base/compile_assert_c.h"
+#include "webrtc/rtc_base/compile_assert_c.h"

 /*
 * We are implementing the following filters;
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@ -28,7 +28,7 @@
 /* include API */
 #include "isac.h"
 #include "utility.h"
-#include "webrtc/base/format_macros.h"
+#include "webrtc/rtc_base/format_macros.h"

 /* max number of samples per frame (= 60 ms frame) */
 #define MAX_FRAMESAMPLES_SWB 1920
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@ -10,7 +10,7 @@

 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"

 #include <stdlib.h>
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@ -15,16 +15,16 @@

 #include <cmath>

-#include "webrtc/base/array_view.h"
-#include "webrtc/base/atomicops.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
+#include "webrtc/rtc_base/array_view.h"
+#include "webrtc/rtc_base/atomicops.h"
+#include "webrtc/rtc_base/bind.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/thread.h"
+#include "webrtc/rtc_base/thread_annotations.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
 #include "webrtc/system_wrappers/include/metrics.h"

--- a/webrtc/modules/audio_device/ios/audio_device_not_implemented_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_not_implemented_ios.mm
@ -10,8 +10,8 @@

 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"

-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"

 namespace webrtc {

--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
@ -16,16 +16,16 @@
 #include <string>
 #include <vector>

-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/format_macros.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+#include "webrtc/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/format_macros.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
@ -475,7 +475,7 @@ class AudioDeviceTest : public ::testing::Test {
  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
    old_sev_ = rtc::LogMessage::GetLogToDebug();
    // Set suitable logging level here. Change to rtc::LS_INFO for more verbose
-    // output. See webrtc/base/logging.h for complete list of options.
+    // output. See webrtc/rtc_base/logging.h for complete list of options.
    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
    // Add extra logging fields here (timestamps and thread id).
    // rtc::LogMessage::LogTimestamps();
--- a/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
@ -10,7 +10,7 @@

 #import "webrtc/modules/audio_device/ios/voice_processing_audio_unit.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 #import "WebRTC/RTCLogging.h"
 #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
--- a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
@ -24,7 +24,7 @@
 #include <stdio.h>
 #endif

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 /* The slope of in Q13*/
 static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129,
--- a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
@ -19,7 +19,7 @@
 #include <stdio.h>
 #endif

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"

 // To generate the gaintable, copy&paste the following lines to a Matlab window:
--- a/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/webrtc/modules/audio_processing/ns/ns_core.c
@ -12,7 +12,7 @@
 #include <string.h>
 #include <stdlib.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/fft4g.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/ns/noise_suppression.h"
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@ -14,7 +14,7 @@
 #include <stdlib.h>
 #include <string.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/modules/audio_processing/ns/nsx_defines.h"
--- a/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@ -10,7 +10,7 @@

 #include <string.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"

--- a/webrtc/modules/audio_processing/ns/nsx_core_neon.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
@ -12,7 +12,7 @@

 #include <arm_neon.h>

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 // Constants to compensate for shifting signal log(2^shifts).
 const int16_t WebRtcNsx_kLogTable[9] = {
--- a/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
+++ b/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
@ -18,14 +18,14 @@
 #include <Cocoa/Cocoa.h>
 #include <CoreFoundation/CoreFoundation.h>

-#include "webrtc/base/macutils.h"
-#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h"
 #include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
+#include "webrtc/rtc_base/macutils.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"

 namespace webrtc {

--- a/webrtc/modules/desktop_capture/screen_capturer_mac.mm
+++ b/webrtc/modules/desktop_capture/screen_capturer_mac.mm
@ -21,13 +21,8 @@
 #include <OpenGL/CGLMacro.h>
 #include <OpenGL/OpenGL.h>

-#include "webrtc/base/checks.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/macutils.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
+#include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
@ -37,6 +32,11 @@
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
 #include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/macutils.h"
+#include "webrtc/rtc_base/timeutils.h"

 // Once Chrome no longer supports OSX 10.8, everything within this
 // preprocessor block can be removed. https://crbug.com/579255
--- a/webrtc/modules/desktop_capture/window_capturer_mac.mm
+++ b/webrtc/modules/desktop_capture/window_capturer_mac.mm
@ -13,16 +13,16 @@
 #include <Cocoa/Cocoa.h>
 #include <CoreFoundation/CoreFoundation.h>

-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/macutils.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
+#include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
 #include "webrtc/modules/desktop_capture/mac/window_list_utils.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/macutils.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"

 namespace webrtc {

--- a/webrtc/modules/video_capture/objc/video_capture.mm
+++ b/webrtc/modules/video_capture/objc/video_capture.mm
@ -12,10 +12,10 @@
 #error "This file requires ARC support."
 #endif

-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/video_capture/objc/device_info_objc.h"
 #include "webrtc/modules/video_capture/objc/rtc_video_capture_objc.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"

 using namespace webrtc;
 using namespace videocapturemodule;
--- a/webrtc/rtc_base/applefilesystem.mm
+++ b/webrtc/rtc_base/applefilesystem.mm
@ -16,8 +16,8 @@
 #import <Foundation/NSProcessInfo.h>
 #include <string.h>

-#include "webrtc/base/checks.h"
-#include "webrtc/base/pathutils.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/pathutils.h"

 // Return a new[]'d |char*| copy of the UTF8 representation of |s|.
 // Caller owns the returned memory and must use delete[] on it.
--- a/webrtc/rtc_base/java/src/org/webrtc/Logging.java
+++ b/webrtc/rtc_base/java/src/org/webrtc/Logging.java
@ -76,7 +76,7 @@ public class Logging {
    }
  }

-  // Keep in sync with webrtc/base/logging.h:LoggingSeverity.
+  // Keep in sync with webrtc/rtc_base/logging.h:LoggingSeverity.
  public enum Severity { LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR, LS_NONE }

  public static void enableLogThreads() {
--- a/webrtc/rtc_base/logging_mac.mm
+++ b/webrtc/rtc_base/logging_mac.mm
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/logging.h"
+#include "webrtc/rtc_base/logging.h"

 #import <Foundation/Foundation.h>

--- a/webrtc/rtc_base/thread_darwin.mm
+++ b/webrtc/rtc_base/thread_darwin.mm
@ -8,11 +8,11 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "webrtc/base/thread.h"
+#include "webrtc/rtc_base/thread.h"

 #import <Foundation/Foundation.h>

-#include "webrtc/base/platform_thread.h"
+#include "webrtc/rtc_base/platform_thread.h"

 /*
 * This file contains platform-specific implementations for several
--- a/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
+++ b/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
@ -40,7 +40,7 @@ public class PeerConnectionFactory {
  private EglBase remoteEglbase;

  public static class Options {
-    // Keep in sync with webrtc/base/network.h!
+    // Keep in sync with webrtc/rtc_base/network.h!
    static final int ADAPTER_TYPE_UNKNOWN = 0;
    static final int ADAPTER_TYPE_ETHERNET = 1 << 0;
    static final int ADAPTER_TYPE_WIFI = 1 << 1;
--- a/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
@ -12,10 +12,9 @@

 #import <UIKit/UIKit.h>

-#include "webrtc/base/atomicops.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/criticalsection.h"
-
+#include "webrtc/rtc_base/atomicops.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/criticalsection.h"

 #import "WebRTC/RTCAudioSessionConfiguration.h"
 #import "WebRTC/RTCLogging.h"
--- a/webrtc/sdk/objc/Framework/Classes/Common/RTCFileLogger.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Common/RTCFileLogger.mm
@ -12,10 +12,10 @@

 #include <memory>

-#include "webrtc/base/checks.h"
-#include "webrtc/base/filerotatingstream.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/logsinks.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/filerotatingstream.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/logsinks.h"

 NSString *const kDefaultLogDirName = @"webrtc_logs";
 NSUInteger const kDefaultMaxFileSize = 10 * 1024 * 1024; // 10MB.
--- a/webrtc/sdk/objc/Framework/Classes/Common/RTCLogging.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Common/RTCLogging.mm
@ -10,7 +10,7 @@

 #import "WebRTC/RTCLogging.h"

-#include "webrtc/base/logging.h"
+#include "webrtc/rtc_base/logging.h"

 rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
  switch (severity) {
--- a/webrtc/sdk/objc/Framework/Classes/Common/helpers.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Common/helpers.mm
@ -17,8 +17,8 @@

 #include <memory>

-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"

 namespace webrtc {
--- a/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm
@ -17,7 +17,7 @@
 #import "WebRTC/RTCVideoFrame.h"

 #include "webrtc/api/video/video_rotation.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 // As defined in shaderSource.
 static NSString *const vertexFunctionName = @"vertexPassthrough";
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm
@ -10,7 +10,7 @@

 #import "RTCAudioSource+Private.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

@implementation RTCAudioSource {
 }
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm
@ -15,7 +15,7 @@
 #import "RTCMediaStreamTrack+Private.h"
 #import "RTCPeerConnectionFactory+Private.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

@implementation RTCAudioTrack

--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
@ -15,8 +15,8 @@
 #import "RTCIceServer+Private.h"
 #import "WebRTC/RTCLogging.h"

-#include "webrtc/base/rtccertificategenerator.h"
-#include "webrtc/base/sslidentity.h"
+#include "webrtc/rtc_base/rtccertificategenerator.h"
+#include "webrtc/rtc_base/sslidentity.h"

@implementation RTCConfiguration

--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport.mm
@ -13,7 +13,7 @@
 #import "NSString+StdString.h"
 #import "WebRTC/RTCLogging.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

@implementation RTCLegacyStatsReport

--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
@ -10,7 +10,7 @@

 #import "RTCMediaSource+Private.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

@implementation RTCMediaSource {
  RTCMediaSourceType _type;
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm
@ -14,7 +14,7 @@
 #import "RTCMediaStreamTrack+Private.h"
 #import "RTCLegacyStatsReport+Private.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 namespace webrtc {

--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
@ -26,7 +26,7 @@
 #include <memory>

 #include "webrtc/api/jsepicecandidate.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

 NSString * const kRTCPeerConnectionErrorDomain =
    @"org.webrtc.RTCPeerConnection";
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
@ -13,8 +13,8 @@
 #import "NSString+StdString.h"
 #import "WebRTC/RTCMediaStreamTrack.h"  // For "kind" strings.

-#include "webrtc/base/checks.h"
 #include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/rtc_base/checks.h"

 const NSString * const kRTCRtxCodecName = @(cricket::kRtxCodecName);
 const NSString * const kRTCRedCodecName = @(cricket::kRedCodecName);
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm
@ -10,8 +10,8 @@

 #import "WebRTC/RTCSSLAdapter.h"

-#include "webrtc/base/checks.h"
-#include "webrtc/base/ssladapter.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/ssladapter.h"

 BOOL RTCInitializeSSL() {
  BOOL initialized = rtc::InitializeSSL();
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription.mm
@ -13,7 +13,7 @@
 #import "NSString+StdString.h"
 #import "WebRTC/RTCLogging.h"

-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"

@implementation RTCSessionDescription

--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm
@ -10,7 +10,7 @@

 #import "WebRTC/RTCTracing.h"

-#include "webrtc/base/event_tracer.h"
+#include "webrtc/rtc_base/event_tracer.h"

 void RTCSetupInternalTracer() {
  rtc::tracing::SetupInternalTracer();
--- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm
@ -11,7 +11,7 @@
 #import "RTCVideoSource+Private.h"

 #include "webrtc/api/videosourceproxy.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.h"

 static webrtc::ObjcVideoTrackSource *getObjcVideoSource(
--- a/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
@ -10,9 +10,9 @@

 #import "WebRTC/RTCVideoFrameBuffer.h"

-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"

@implementation RTCCVPixelBuffer {
  int _width;
--- a/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
@ -20,7 +20,7 @@
 #import "RTCShader.h"
 #import "WebRTC/RTCLogging.h"

-#include "webrtc/base/optional.h"
+#include "webrtc/rtc_base/optional.h"

 static const int kYTextureUnit = 0;
 static const int kUTextureUnit = 1;
--- a/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.mm
@ -22,8 +22,8 @@

 #import "RTCOpenGLDefines.h"

-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"

 // Vertex shader doesn't do anything except pass coordinates through.
 const char kRTCVertexShaderSource[] =
--- a/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.mm
@ -20,10 +20,10 @@
 #include "avfoundationformatmapper.h"

 #include "webrtc/api/video/video_rotation.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/thread.h"
+#include "webrtc/rtc_base/bind.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/thread.h"
 #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"

 namespace webrtc {
--- a/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.mm
+++ b/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.mm
@ -15,9 +15,9 @@

 #include "libyuv/convert.h"
 #include "webrtc/api/video/video_frame.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/common_video/include/video_frame.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
 #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"

--- a/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm
+++ b/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm
@ -21,9 +21,9 @@
 #endif
 #import "WebRTC/RTCVideoFrameBuffer.h"
 #include "libyuv/convert_from.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
 #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
 #include "webrtc/system_wrappers/include/clock.h"
--- a/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm
+++ b/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm
@ -9,9 +9,9 @@
 */
 #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h"

-#include "webrtc/base/logging.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/media/base/codec.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h"
 #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <OCMock/OCMock.h>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "RTCAudioSession+Private.h"

--- a/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm
@ -14,7 +14,7 @@
 #import <UIKit/UIKit.h>
 #endif

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import <WebRTC/RTCCameraVideoCapturer.h>
 #import <WebRTC/RTCDispatcher.h>
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
@ -12,7 +12,7 @@

 #include <vector>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCConfiguration+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm
@ -10,7 +10,7 @@

 #import <Foundation/Foundation.h>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCDataChannelConfiguration+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
@ -12,7 +12,7 @@

 #include <memory>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCIceCandidate+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
@ -12,7 +12,7 @@

 #include <vector>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCIceServer+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoViewTests.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoViewTests.mm
@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <OCMock/OCMock.h>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #include <Metal/RTCMTLNV12Renderer.h>
 #include <WebRTC/RTCMTLVideoView.h>
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
@ -12,7 +12,7 @@

 #include <memory>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCMediaConstraints+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
@ -12,7 +12,7 @@

 #include <vector>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCConfiguration+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
@ -10,7 +10,7 @@

 #import <Foundation/Foundation.h>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #import "NSString+StdString.h"
 #import "RTCSessionDescription+Private.h"
--- a/webrtc/sdk/objc/Framework/UnitTests/avformatmappertests.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/avformatmappertests.mm
@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <OCMock/OCMock.h>

-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"

 #include "Video/avfoundationformatmapper.h"

--- a/webrtc/test/testsupport/iosfileutils.mm
+++ b/webrtc/test/testsupport/iosfileutils.mm
@ -13,9 +13,9 @@
 #import <Foundation/Foundation.h>
 #include <string.h>

-#include "webrtc/base/checks.h"
-#include "webrtc/typedefs.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
+#include "webrtc/typedefs.h"

 namespace webrtc {
 namespace test {