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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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1
sdk/android/src/jni/audio_device/audio_record_jni.cc
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@ -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"

1
sdk/android/src/jni/audio_device/audio_track_jni.cc
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@ -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"

6
sdk/android/src/jni/audio_device/opensles_player.cc
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@ -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 @@ void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
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 @@ void OpenSLESPlayer::AllocateDataBuffers() {
// 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_);

12
sdk/android/src/jni/audio_device/opensles_recorder.cc
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@ -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 @@ void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
// 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 @@ void OpenSLESRecorder::AllocateDataBuffers() {
// 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_);