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Delete rtc_base/format_macros.h

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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}
This commit is contained in:
Niels Möller
2022-05-09 09:21:14 +02:00
committed by WebRTC LUCI CQ
parent cc50b04c02
commit ea1e6f44f8
31 changed files with 68 additions and 166 deletions
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21
sdk/android/native_unittests/audio_device/audio_device_unittest.cc
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@ -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 @@ class FifoAudioStream : public AudioStreamInterface {
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 @@ class AudioDeviceTest : public ::testing::Test {
#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 @@ TEST_F(AudioDeviceTest, ShowAudioParameterInfo) {
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());
}