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}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
index 8dcb6b7..e05b88b 100644
--- 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"
/*----------------------------------------------------------------*
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
index 8206687..fa4ffd0 100644
--- 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"
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index 6587e4f..18e0f84 100644
--- 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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index fa63b46..ab5aa0a 100644
--- 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)) */
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
index 248511f..ee07660 100644
--- 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>
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index dc8bcf3..69ee414 100644
--- 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()
diff --git 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
index 1e4812a..20437e5 100644
--- 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[],
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
index ce479e2..bc21617 100644
--- 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
index 5dd6e8f..cd0bb07 100644
--- 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
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c
index 2e666d6..31d4e08 100644
--- 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.
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
index df4ef6f..9ec2b13 100644
--- 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.
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 39ae8be..3377bf7 100644
--- 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"
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
index b69a885..1f8b2c2 100644
--- 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))))
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
index a6c9814..ac8511e 100644
--- 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] = {
diff --git 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
index 7485ad0..81ec1d6 100644
--- 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);
diff --git 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
index bbfc0fa..a76ed7d 100644
--- 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);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
index a64ab45..bcd6038 100644
--- 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index f3d9e1b..c2c166c 100644
--- 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>
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index bf1cff4..f83d4f8 100644
--- 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"
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
index d65f00e..3010fbc 100644
--- 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 5d7a92a..646cd48 100644
--- 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
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 59c6c9b..6427f0f 100644
--- 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>