Delete rtc_base/format_macros.h It defined RTC_PRIuS, which was needed for compatibility with MSVC prior to version 2015. Bug: webrtc:6424 Change-Id: I5668d473376201cad3e8da65927c967fc397804b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36814}

diff --git a/api/call/audio_sink.h b/api/call/audio_sink.h
index fa4c3f6..76feb13 100644
--- a/api/call/audio_sink.h
+++ b/api/call/audio_sink.h

@@ -11,13 +11,7 @@
 #ifndef API_CALL_AUDIO_SINK_H_
 #define API_CALL_AUDIO_SINK_H_
 
-#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
-// Avoid conflict with format_macros.h.
-#define __STDC_FORMAT_MACROS
-#endif
-
-#include <inttypes.h>
-#include <stddef.h>
+#include <stdint.h>
 
 namespace webrtc {
 

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index f2fd34a..93f534e 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc

@@ -40,7 +40,6 @@
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 6deb4f1..05341b9 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc

@@ -33,7 +33,6 @@
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"

diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc
index a80476e..30079bf 100644
--- a/audio/remix_resample_unittest.cc
+++ b/audio/remix_resample_unittest.cc

@@ -15,7 +15,6 @@
 #include "common_audio/resampler/include/push_resampler.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -140,7 +139,7 @@
       best_delay = delay;
     }
   }
-  printf("SNR=%.1f dB at delay=%" RTC_PRIuS "\n", best_snr, best_delay);
+  printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
   return best_snr;
 }
 

diff --git a/common_audio/audio_converter_unittest.cc b/common_audio/audio_converter_unittest.cc
index 018e7b8..97937c8 100644
--- a/common_audio/audio_converter_unittest.cc
+++ b/common_audio/audio_converter_unittest.cc

@@ -18,7 +18,6 @@
 #include "common_audio/channel_buffer.h"
 #include "common_audio/resampler/push_sinc_resampler.h"
 #include "rtc_base/arraysize.h"
-#include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -79,7 +78,7 @@
       best_delay = delay;
     }
   }
-  printf("SNR=%.1f dB at delay=%" RTC_PRIuS "\n", best_snr, best_delay);
+  printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
   return best_snr;
 }
 
@@ -131,8 +130,8 @@
                 PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
                 dst_sample_rate_hz);
   // SNR reported on the same line later.
-  printf("(%" RTC_PRIuS ", %d Hz) -> (%" RTC_PRIuS ", %d Hz) ", src_channels,
-         src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+  printf("(%zu, %d Hz) -> (%zu, %d Hz) ", src_channels, src_sample_rate_hz,
+         dst_channels, dst_sample_rate_hz);
 
   std::unique_ptr<AudioConverter> converter = AudioConverter::Create(
       src_channels, src_frames, dst_channels, dst_frames);

diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 9129df8f..57ebb54 100644
--- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc

@@ -22,7 +22,6 @@
 /* include API */
 #include "modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "modules/audio_coding/codecs/isac/main/util/utility.h"
-#include "rtc_base/format_macros.h"
 
 /* Defines */
 #define SEED_FILE                                             \
@@ -887,7 +886,7 @@
 #endif
   }
   printf("\n");
-  printf("total bits               = %" RTC_PRIuS " bits\n", totalbits);
+  printf("total bits               = %zu bits\n", totalbits);
   printf("measured average bitrate = %0.3f kbits/s\n",
          (double)totalbits * (sampFreqKHz) / totalsmpls);
   if (doTransCoding) {
@@ -906,14 +905,16 @@
          (100 * runtime / length_file));
 
   if (maxStreamLen30 != 0) {
-    printf("Maximum payload size 30ms Frames %" RTC_PRIuS
-           " bytes (%0.3f kbps)\n",
-           maxStreamLen30, maxStreamLen30 * 8 / 30.);
+    printf(
+        "Maximum payload size 30ms Frames %zu"
+        " bytes (%0.3f kbps)\n",
+        maxStreamLen30, maxStreamLen30 * 8 / 30.);
   }
   if (maxStreamLen60 != 0) {
-    printf("Maximum payload size 60ms Frames %" RTC_PRIuS
-           " bytes (%0.3f kbps)\n",
-           maxStreamLen60, maxStreamLen60 * 8 / 60.);
+    printf(
+        "Maximum payload size 60ms Frames %zu"
+        " bytes (%0.3f kbps)\n",
+        maxStreamLen60, maxStreamLen60 * 8 / 60.);
   }
   // fprintf(stderr, "\n");
 
@@ -921,12 +922,12 @@
   fprintf(stderr, "   %0.1f kbps",
           (double)totalbits * (sampFreqKHz) / totalsmpls);
   if (maxStreamLen30 != 0) {
-    fprintf(stderr, "   plmax-30ms %" RTC_PRIuS " bytes (%0.0f kbps)",
-            maxStreamLen30, maxStreamLen30 * 8 / 30.);
+    fprintf(stderr, "   plmax-30ms %zu bytes (%0.0f kbps)", maxStreamLen30,
+            maxStreamLen30 * 8 / 30.);
   }
   if (maxStreamLen60 != 0) {
-    fprintf(stderr, "   plmax-60ms %" RTC_PRIuS " bytes (%0.0f kbps)",
-            maxStreamLen60, maxStreamLen60 * 8 / 60.);
+    fprintf(stderr, "   plmax-60ms %zu bytes (%0.0f kbps)", maxStreamLen60,
+            maxStreamLen60 * 8 / 60.);
   }
   if (doTransCoding) {
     fprintf(stderr, "  transcoding rate %.0f kbps",

diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 116b051..96b9b23 100644
--- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c

@@ -28,7 +28,6 @@
 /* include API */
 #include "modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "modules/audio_coding/codecs/isac/main/util/utility.h"
-#include "rtc_base/format_macros.h"
 
 /* max number of samples per frame (= 60 ms frame) */
 #define MAX_FRAMESAMPLES_SWB 1920
@@ -420,7 +419,7 @@
   printf("\n");
   printf("Measured bit-rate........... %0.3f kbps\n", rate);
   printf("Measured RCU bit-ratre...... %0.3f kbps\n", rateRCU);
-  printf("Maximum bit-rate/payloadsize %0.3f / %" RTC_PRIuS "\n",
+  printf("Maximum bit-rate/payloadsize %0.3f / %zu\n",
          maxStreamLen * 8 / 0.03, maxStreamLen);
   printf("Measured packet-loss........ %0.1f%% \n",
          100.0f * (float)lostPacketCntr / (float)packetCntr);

diff --git a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
index dbb808c..6388f33 100644
--- a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc

@@ -10,7 +10,6 @@
 
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
 #include "modules/audio_coding/neteq/tools/audio_loop.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/time_utils.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"

diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 0636935..815f26e 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc

@@ -11,7 +11,6 @@
 #include <memory>
 
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
-#include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
@@ -70,8 +69,7 @@
 void OpusFecTest::SetUp() {
   channels_ = get<0>(GetParam());
   bit_rate_ = get<1>(GetParam());
-  printf("Coding %" RTC_PRIuS " channel signal at %d bps.\n", channels_,
-         bit_rate_);
+  printf("Coding %zu channel signal at %d bps.\n", channels_, bit_rate_);
 
   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
 

diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index f61aacc..537e6fc 100644
--- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc

@@ -11,7 +11,6 @@
 #include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
@@ -100,7 +99,7 @@
   size_t time_now_ms = 0;
   float time_ms;
 
-  printf("Coding %d kHz-sampled %" RTC_PRIuS "-channel audio at %d bps ...\n",
+  printf("Coding %d kHz-sampled %zu-channel audio at %d bps ...\n",
          input_sampling_khz_, channels_, bit_rate_);
 
   while (time_now_ms < audio_duration_sec * 1000) {

diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index ec0eccb..b85f9f4 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc

@@ -10,10 +10,8 @@
 
 #include "modules/audio_coding/test/Channel.h"
 
-
 #include <iostream>
 
-#include "rtc_base/format_macros.h"
 #include "rtc_base/time_utils.h"
 
 namespace webrtc {

diff --git a/modules/audio_device/android/audio_device_unittest.cc b/modules/audio_device/android/audio_device_unittest.cc
index 79cd69f..4e607bc 100644
--- a/modules/audio_device/android/audio_device_unittest.cc
+++ b/modules/audio_device/android/audio_device_unittest.cc

@@ -29,7 +29,6 @@
 #include "modules/audio_device/include/mock_audio_transport.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/event.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/time_utils.h"
 #include "test/gmock.h"
@@ -187,7 +186,7 @@
     const size_t size = fifo_->size();
     if (size > largest_size_) {
       largest_size_ = size;
-      PRINTD("(%" RTC_PRIuS ")", largest_size_);
+      PRINTD("(%zu)", largest_size_);
     }
     total_written_elements_ += size;
   }
@@ -532,13 +531,12 @@
 #ifdef ENABLE_PRINTF
     PRINT("file name: %s\n", file_name.c_str());
     const size_t bytes = test::GetFileSize(file_name);
-    PRINT("file size: %" RTC_PRIuS " [bytes]\n", bytes);
-    PRINT("file size: %" RTC_PRIuS " [samples]\n", bytes / kBytesPerSample);
+    PRINT("file size: %zu [bytes]\n", bytes);
+    PRINT("file size: %zu [samples]\n", bytes / kBytesPerSample);
     const int seconds =
         static_cast<int>(bytes / (sample_rate * kBytesPerSample));
     PRINT("file size: %d [secs]\n", seconds);
-    PRINT("file size: %" RTC_PRIuS " [callbacks]\n",
-          seconds * kNumCallbacksPerSecond);
+    PRINT("file size: %zu [callbacks]\n", seconds * kNumCallbacksPerSecond);
 #endif
     return file_name;
   }

diff --git a/modules/audio_device/android/audio_manager_unittest.cc b/modules/audio_device/android/audio_manager_unittest.cc
index 1b81904..093eddd 100644
--- a/modules/audio_device/android/audio_manager_unittest.cc
+++ b/modules/audio_device/android/audio_manager_unittest.cc

@@ -15,7 +15,6 @@
 #include "modules/audio_device/android/build_info.h"
 #include "modules/audio_device/android/ensure_initialized.h"
 #include "rtc_base/arraysize.h"
-#include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 
 #define PRINT(...) fprintf(stderr, __VA_ARGS__);
@@ -153,16 +152,16 @@
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
   PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
-  PRINT("%schannels: %" RTC_PRIuS "\n", kTag, playout_parameters_.channels());
-  PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
+  PRINT("%schannels: %zu\n", kTag, playout_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
         playout_parameters_.frames_per_buffer(),
         playout_parameters_.GetBufferSizeInMilliseconds());
   PRINT("RECORD: \n");
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
   PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
-  PRINT("%schannels: %" RTC_PRIuS "\n", kTag, record_parameters_.channels());
-  PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
+  PRINT("%schannels: %zu\n", kTag, record_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
         record_parameters_.frames_per_buffer(),
         record_parameters_.GetBufferSizeInMilliseconds());
 }

diff --git a/modules/audio_device/android/audio_record_jni.cc b/modules/audio_device/android/audio_record_jni.cc
index 9d7bf73..919eabb 100644
--- a/modules/audio_device/android/audio_record_jni.cc
+++ b/modules/audio_device/android/audio_record_jni.cc

@@ -16,7 +16,6 @@
 #include "modules/audio_device/android/audio_common.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"

diff --git a/modules/audio_device/android/audio_track_jni.cc b/modules/audio_device/android/audio_track_jni.cc
index 178ccad..5afa1ec 100644
--- a/modules/audio_device/android/audio_track_jni.cc
+++ b/modules/audio_device/android/audio_track_jni.cc

@@ -15,7 +15,6 @@
 #include "modules/audio_device/android/audio_manager.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/platform_thread.h"
 #include "system_wrappers/include/field_trial.h"

diff --git a/modules/audio_device/android/opensles_player.cc b/modules/audio_device/android/opensles_player.cc
index b5851f7..f2b3a37 100644
--- a/modules/audio_device/android/opensles_player.cc
+++ b/modules/audio_device/android/opensles_player.cc

@@ -20,7 +20,6 @@
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"
 
@@ -193,7 +192,7 @@
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
   const size_t channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%" RTC_PRIuS ")", channels);
+  ALOGD("SetPlayoutChannels(%zu)", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
   RTC_CHECK(audio_device_buffer_);
   AllocateDataBuffers();
@@ -214,7 +213,7 @@
   // which reduces jitter.
   const size_t buffer_size_in_samples =
       audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
-  ALOGD("native buffer size: %" RTC_PRIuS, buffer_size_in_samples);
+  ALOGD("native buffer size: %zu", buffer_size_in_samples);
   ALOGD("native buffer size in ms: %.2f",
         audio_parameters_.GetBufferSizeInMilliseconds());
   fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);

diff --git a/modules/audio_device/android/opensles_recorder.cc b/modules/audio_device/android/opensles_recorder.cc
index 8becd20..4e0c26d 100644
--- a/modules/audio_device/android/opensles_recorder.cc
+++ b/modules/audio_device/android/opensles_recorder.cc

@@ -20,7 +20,6 @@
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"
 
@@ -178,7 +177,7 @@
   // Ensure that the audio device buffer is informed about the number of
   // channels preferred by the OS on the recording side.
   const size_t channels = audio_parameters_.channels();
-  ALOGD("SetRecordingChannels(%" RTC_PRIuS ")", channels);
+  ALOGD("SetRecordingChannels(%zu)", channels);
   audio_device_buffer_->SetRecordingChannels(channels);
   // Allocated memory for internal data buffers given existing audio parameters.
   AllocateDataBuffers();
@@ -334,12 +333,10 @@
   // Create a modified audio buffer class which allows us to deliver any number
   // of samples (and not only multiple of 10ms) to match the native audio unit
   // buffer size.
-  ALOGD("frames per native buffer: %" RTC_PRIuS,
-        audio_parameters_.frames_per_buffer());
-  ALOGD("frames per 10ms buffer: %" RTC_PRIuS,
+  ALOGD("frames per native buffer: %zu", audio_parameters_.frames_per_buffer());
+  ALOGD("frames per 10ms buffer: %zu",
         audio_parameters_.frames_per_10ms_buffer());
-  ALOGD("bytes per native buffer: %" RTC_PRIuS,
-        audio_parameters_.GetBytesPerBuffer());
+  ALOGD("bytes per native buffer: %zu", audio_parameters_.GetBytesPerBuffer());
   ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
   RTC_DCHECK(audio_device_buffer_);
   fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);

diff --git a/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
index f99576f..c8f6faa 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
+++ b/modules/remote_bitrate_estimator/test/bwe_test_logging.cc

@@ -12,13 +12,13 @@
 
 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
 
+#include <inttypes.h>
 #include <stdarg.h>
 #include <stdio.h>
 
 #include <algorithm>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/strings/string_builder.h"
 

diff --git a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index 98f502a..e8dc59f 100644
--- a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc

@@ -16,7 +16,6 @@
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/rtp_file_reader.h"
 
@@ -52,10 +51,10 @@
       ss << static_cast<int64_t>(packet.time_ms) * 1000000;
       fprintf(stdout, "%s\n", ss.str().c_str());
     } else {
-      fprintf(stdout, "%u %u %d %u %u %d %u %" RTC_PRIuS " %" RTC_PRIuS "\n",
-              header.SequenceNumber(), header.Timestamp(), toffset,
-              abs_send_time, packet.time_ms, header.Marker(), header.Ssrc(),
-              packet.length, packet.original_length);
+      fprintf(stdout, "%u %u %d %u %u %d %u %zu %zu\n", header.SequenceNumber(),
+              header.Timestamp(), toffset, abs_send_time, packet.time_ms,
+              header.Marker(), header.Ssrc(), packet.length,
+              packet.original_length);
     }
     ++packet_counter;
   }

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 7765c61..b4cf481 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn

@@ -127,7 +127,6 @@
 rtc_source_set("macromagic") {
   sources = [
     "arraysize.h",
-    "format_macros.h",
     "thread_annotations.h",
   ]
   deps = [ "system:arch" ]

diff --git a/rtc_base/format_macros.h b/rtc_base/format_macros.h
deleted file mode 100644
index 83240fb..0000000
--- a/rtc_base/format_macros.h
+++ /dev/null

@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTC_BASE_FORMAT_MACROS_H_
-#define RTC_BASE_FORMAT_MACROS_H_
-
-// This file defines the format macros for some integer types and is derived
-// from Chromium's base/format_macros.h.
-
-// To print a 64-bit value in a portable way:
-//   int64_t value;
-//   printf("xyz:%" PRId64, value);
-// The "d" in the macro corresponds to %d; you can also use PRIu64 etc.
-//
-// To print a size_t value in a portable way:
-//   size_t size;
-//   printf("xyz: %" RTC_PRIuS, size);
-// The "u" in the macro corresponds to %u, and S is for "size".
-
-#if defined(WEBRTC_POSIX)
-
-#if (defined(_INTTYPES_H) || defined(_INTTYPES_H_)) && !defined(PRId64)
-#error "inttypes.h has already been included before this header file, but "
-#error "without __STDC_FORMAT_MACROS defined."
-#endif
-
-#if !defined(__STDC_FORMAT_MACROS)
-#define __STDC_FORMAT_MACROS
-#endif
-
-#include <inttypes.h>
-
-#include "rtc_base/system/arch.h"
-
-#define RTC_PRIuS "zu"
-
-#else  // WEBRTC_WIN
-
-#include <inttypes.h>
-
-#if !defined(PRId64) || !defined(PRIu64) || !defined(PRIx64)
-#error "inttypes.h provided by win toolchain should define these."
-#endif
-
-// PRI*64 were added in MSVC 2013, while "%zu" is supported since MSVC 2015
-// (so needs to be special-cased to "%Iu" instead).
-
-#define RTC_PRIuS "Iu"
-
-#endif
-
-#endif  // RTC_BASE_FORMAT_MACROS_H_

diff --git a/rtc_tools/rtc_event_log_visualizer/alerts.cc b/rtc_tools/rtc_event_log_visualizer/alerts.cc
index c0d8784..9ef5e9a 100644
--- a/rtc_tools/rtc_event_log_visualizer/alerts.cc
+++ b/rtc_tools/rtc_event_log_visualizer/alerts.cc

@@ -19,7 +19,6 @@
 
 #include "logging/rtc_event_log/rtc_event_processor.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/strings/string_builder.h"

diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 0185c7d..bdbb438 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc

@@ -48,7 +48,6 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/rate_statistics.h"

diff --git a/rtc_tools/unpack_aecdump/unpack.cc b/rtc_tools/unpack_aecdump/unpack.cc
index 0850e75..49b62d2 100644
--- a/rtc_tools/unpack_aecdump/unpack.cc
+++ b/rtc_tools/unpack_aecdump/unpack.cc

@@ -28,7 +28,6 @@
 #include "common_audio/wav_file.h"
 #include "modules/audio_processing/test/protobuf_utils.h"
 #include "modules/audio_processing/test/test_utils.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/ignore_wundef.h"
 #include "rtc_base/strings/string_builder.h"
 
@@ -480,14 +479,11 @@
       fprintf(settings_file, "  Reverse sample rate: %d\n",
               reverse_sample_rate);
       num_input_channels = msg.num_input_channels();
-      fprintf(settings_file, "  Input channels: %" RTC_PRIuS "\n",
-              num_input_channels);
+      fprintf(settings_file, "  Input channels: %zu\n", num_input_channels);
       num_output_channels = msg.num_output_channels();
-      fprintf(settings_file, "  Output channels: %" RTC_PRIuS "\n",
-              num_output_channels);
+      fprintf(settings_file, "  Output channels: %zu\n", num_output_channels);
       num_reverse_channels = msg.num_reverse_channels();
-      fprintf(settings_file, "  Reverse channels: %" RTC_PRIuS "\n",
-              num_reverse_channels);
+      fprintf(settings_file, "  Reverse channels: %zu\n", num_reverse_channels);
       if (msg.has_timestamp_ms()) {
         const int64_t timestamp = msg.timestamp_ms();
         fprintf(settings_file, "  Timestamp in millisecond: %" PRId64 "\n",

diff --git a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
index 717c074..7d582d4 100644
--- a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
+++ b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "modules/audio_device/include/audio_device.h"
+
 #include <list>
 #include <memory>
 #include <numeric>
 
 #include "api/scoped_refptr.h"
-#include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_device/include/mock_audio_transport.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/event.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/time_utils.h"
 #include "sdk/android/generated_native_unittests_jni/BuildInfo_jni.h"
@@ -184,7 +184,7 @@
     const size_t size = fifo_->size();
     if (size > largest_size_) {
       largest_size_ = size;
-      PRINTD("(%" RTC_PRIuS ")", largest_size_);
+      PRINTD("(%zu)", largest_size_);
     }
     total_written_elements_ += size;
   }
@@ -547,13 +547,12 @@
 #ifdef ENABLE_PRINTF
     PRINT("file name: %s\n", file_name.c_str());
     const size_t bytes = test::GetFileSize(file_name);
-    PRINT("file size: %" RTC_PRIuS " [bytes]\n", bytes);
-    PRINT("file size: %" RTC_PRIuS " [samples]\n", bytes / kBytesPerSample);
+    PRINT("file size: %zu [bytes]\n", bytes);
+    PRINT("file size: %zu [samples]\n", bytes / kBytesPerSample);
     const int seconds =
         static_cast<int>(bytes / (sample_rate * kBytesPerSample));
     PRINT("file size: %d [secs]\n", seconds);
-    PRINT("file size: %" RTC_PRIuS " [callbacks]\n",
-          seconds * kNumCallbacksPerSecond);
+    PRINT("file size: %zu [callbacks]\n", seconds * kNumCallbacksPerSecond);
 #endif
     return file_name;
   }
@@ -972,16 +971,16 @@
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
   PRINT("%ssample rate: %d Hz\n", kTag, output_parameters_.sample_rate());
-  PRINT("%schannels: %" RTC_PRIuS "\n", kTag, output_parameters_.channels());
-  PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
+  PRINT("%schannels: %zu\n", kTag, output_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
         output_parameters_.frames_per_buffer(),
         output_parameters_.GetBufferSizeInMilliseconds());
   PRINT("RECORD: \n");
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
   PRINT("%ssample rate: %d Hz\n", kTag, input_parameters_.sample_rate());
-  PRINT("%schannels: %" RTC_PRIuS "\n", kTag, input_parameters_.channels());
-  PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
+  PRINT("%schannels: %zu\n", kTag, input_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
         input_parameters_.frames_per_buffer(),
         input_parameters_.GetBufferSizeInMilliseconds());
 }

diff --git a/sdk/android/src/jni/audio_device/audio_record_jni.cc b/sdk/android/src/jni/audio_device/audio_record_jni.cc
index f5f1089..d206297 100644
--- a/sdk/android/src/jni/audio_device/audio_record_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_record_jni.cc

@@ -15,7 +15,6 @@
 
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"

diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc
index f2f22f9..c1ff4c3 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc

@@ -14,7 +14,6 @@
 
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/platform_thread.h"
 #include "sdk/android/generated_java_audio_device_module_native_jni/WebRtcAudioTrack_jni.h"

diff --git a/sdk/android/src/jni/audio_device/opensles_player.cc b/sdk/android/src/jni/audio_device/opensles_player.cc
index 5192acc..6300a3a 100644
--- a/sdk/android/src/jni/audio_device/opensles_player.cc
+++ b/sdk/android/src/jni/audio_device/opensles_player.cc

@@ -13,11 +13,11 @@
 #include <android/log.h>
 
 #include <memory>
+
 #include "api/array_view.h"
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"
 #include "sdk/android/src/jni/audio_device/audio_common.h"
@@ -202,7 +202,7 @@
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
   const size_t channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%" RTC_PRIuS ")", channels);
+  ALOGD("SetPlayoutChannels(%zu)", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
   RTC_CHECK(audio_device_buffer_);
   AllocateDataBuffers();
@@ -223,7 +223,7 @@
   // which reduces jitter.
   const size_t buffer_size_in_samples =
       audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
-  ALOGD("native buffer size: %" RTC_PRIuS, buffer_size_in_samples);
+  ALOGD("native buffer size: %zu", buffer_size_in_samples);
   ALOGD("native buffer size in ms: %.2f",
         audio_parameters_.GetBufferSizeInMilliseconds());
   fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);

diff --git a/sdk/android/src/jni/audio_device/opensles_recorder.cc b/sdk/android/src/jni/audio_device/opensles_recorder.cc
index d2eb2de..c426a8d9 100644
--- a/sdk/android/src/jni/audio_device/opensles_recorder.cc
+++ b/sdk/android/src/jni/audio_device/opensles_recorder.cc

@@ -13,11 +13,11 @@
 #include <android/log.h>
 
 #include <memory>
+
 #include "api/array_view.h"
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/time_utils.h"
 #include "sdk/android/src/jni/audio_device/audio_common.h"
@@ -188,7 +188,7 @@
   // Ensure that the audio device buffer is informed about the number of
   // channels preferred by the OS on the recording side.
   const size_t channels = audio_parameters_.channels();
-  ALOGD("SetRecordingChannels(%" RTC_PRIuS ")", channels);
+  ALOGD("SetRecordingChannels(%zu)", channels);
   audio_device_buffer_->SetRecordingChannels(channels);
   // Allocated memory for internal data buffers given existing audio parameters.
   AllocateDataBuffers();
@@ -345,12 +345,10 @@
   // Create a modified audio buffer class which allows us to deliver any number
   // of samples (and not only multiple of 10ms) to match the native audio unit
   // buffer size.
-  ALOGD("frames per native buffer: %" RTC_PRIuS,
-        audio_parameters_.frames_per_buffer());
-  ALOGD("frames per 10ms buffer: %" RTC_PRIuS,
+  ALOGD("frames per native buffer: %zu", audio_parameters_.frames_per_buffer());
+  ALOGD("frames per 10ms buffer: %zu",
         audio_parameters_.frames_per_10ms_buffer());
-  ALOGD("bytes per native buffer: %" RTC_PRIuS,
-        audio_parameters_.GetBytesPerBuffer());
+  ALOGD("bytes per native buffer: %zu", audio_parameters_.GetBytesPerBuffer());
   ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
   RTC_DCHECK(audio_device_buffer_);
   fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);

diff --git a/test/rtp_file_reader.cc b/test/rtp_file_reader.cc
index 15a0513..7fc91be 100644
--- a/test/rtp_file_reader.cc
+++ b/test/rtp_file_reader.cc

@@ -18,7 +18,6 @@
 
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/system/arch.h"
 
@@ -290,15 +289,15 @@
     }
 
     printf("Total packets in file: %d\n", total_packet_count);
-    printf("Total RTP/RTCP packets: %" RTC_PRIuS "\n", packets_.size());
+    printf("Total RTP/RTCP packets: %zu\n", packets_.size());
 
     for (SsrcMapIterator mit = packets_by_ssrc_.begin();
          mit != packets_by_ssrc_.end(); ++mit) {
       uint32_t ssrc = mit->first;
       const std::vector<uint32_t>& packet_indices = mit->second;
       int pt = packets_[packet_indices[0]].payload_type;
-      printf("SSRC: %08x, %" RTC_PRIuS " packets, pt=%d\n", ssrc,
-             packet_indices.size(), pt);
+      printf("SSRC: %08x, %zu packets, pt=%d\n", ssrc, packet_indices.size(),
+             pt);
     }
 
     // TODO(solenberg): Better validation of identified SSRC streams.

diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc
index 62ee7b4..c55bf96 100644
--- a/video/video_analyzer.cc
+++ b/video/video_analyzer.cc

@@ -9,6 +9,8 @@
  */
 #include "video/video_analyzer.h"
 
+#include <inttypes.h>
+
 #include <algorithm>
 #include <utility>
 
@@ -20,7 +22,6 @@
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/cpu_time.h"
-#include "rtc_base/format_macros.h"
 #include "rtc_base/memory_usage.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/task_utils/repeating_task.h"
@@ -854,7 +855,7 @@
   });
 
   fprintf(out, "%s\n", graph_title_.c_str());
-  fprintf(out, "%" RTC_PRIuS "\n", samples_.size());
+  fprintf(out, "%zu\n", samples_.size());
   fprintf(out,
           "dropped "
           "input_time_ms "
@@ -867,8 +868,7 @@
           "encode_time_ms\n");
   for (const Sample& sample : samples_) {
     fprintf(out,
-            "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" RTC_PRIuS
-            " %lf %lf\n",
+            "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %zu %lf %lf\n",
             sample.dropped, sample.input_time_ms, sample.send_time_ms,
             sample.recv_time_ms, sample.render_time_ms,
             sample.encoded_frame_size, sample.psnr, sample.ssim);