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Add RTC_ prefix to non-standard format specifier macro "PRIdNS"

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Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.

References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/
https://stackoverflow.com/a/2524673

Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
This commit is contained in:
Oleh Prypin 2019-08-02 09:36:47 +02:00 committed by Commit Bot
parent 12ebfa69ba
commit b1686786e8
18 changed files with 60 additions and 54 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
audio/remix_resample_unittest.cc
View File

@ -140,7 +140,7 @@ float ComputeSNR(const AudioFrame& ref_frame,
best_delay = delay;
}
}
printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
printf("SNR=%.1f dB at delay=%" RTC_PRIuS "\n", best_snr, best_delay);
return best_snr;
}

4
common_audio/audio_converter_unittest.cc
View File

@ -79,7 +79,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
best_delay = delay;
}
}
printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
printf("SNR=%.1f dB at delay=%" RTC_PRIuS "\n", best_snr, best_delay);
return best_snr;
}
@ -131,7 +131,7 @@ void RunAudioConverterTest(size_t src_channels,
PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
dst_sample_rate_hz);
// SNR reported on the same line later.
printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ", src_channels,
printf("(%" RTC_PRIuS ", %d Hz) -> (%" RTC_PRIuS ", %d Hz) ", src_channels,
src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
std::unique_ptr<AudioConverter> converter = AudioConverter::Create(

12
modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -906,7 +906,7 @@ int main(int argc, char* argv[]) {
#endif
}
printf("\n");
printf("total bits = %" PRIuS " bits\n", totalbits);
printf("total bits = %" RTC_PRIuS " bits\n", totalbits);
printf("measured average bitrate = %0.3f kbits/s\n",
(double)totalbits * (sampFreqKHz) / totalsmpls);
if (doTransCoding) {
@ -925,11 +925,13 @@ int main(int argc, char* argv[]) {
(100 * runtime / length_file));
if (maxStreamLen30 != 0) {
printf("Maximum payload size 30ms Frames %" PRIuS " bytes (%0.3f kbps)\n",
printf("Maximum payload size 30ms Frames %" RTC_PRIuS
" bytes (%0.3f kbps)\n",
maxStreamLen30, maxStreamLen30 * 8 / 30.);
}
if (maxStreamLen60 != 0) {
printf("Maximum payload size 60ms Frames %" PRIuS " bytes (%0.3f kbps)\n",
printf("Maximum payload size 60ms Frames %" RTC_PRIuS
" bytes (%0.3f kbps)\n",
maxStreamLen60, maxStreamLen60 * 8 / 60.);
}
// fprintf(stderr, "\n");
@ -938,11 +940,11 @@ int main(int argc, char* argv[]) {
fprintf(stderr, " %0.1f kbps",
(double)totalbits * (sampFreqKHz) / totalsmpls);
if (maxStreamLen30 != 0) {
fprintf(stderr, " plmax-30ms %" PRIuS " bytes (%0.0f kbps)",
fprintf(stderr, " plmax-30ms %" RTC_PRIuS " bytes (%0.0f kbps)",
maxStreamLen30, maxStreamLen30 * 8 / 30.);
}
if (maxStreamLen60 != 0) {
fprintf(stderr, " plmax-60ms %" PRIuS " bytes (%0.0f kbps)",
fprintf(stderr, " plmax-60ms %" RTC_PRIuS " bytes (%0.0f kbps)",
maxStreamLen60, maxStreamLen60 * 8 / 60.);
}
if (doTransCoding) {

2
modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -424,7 +424,7 @@ int main(int argc, char* argv[]) {
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 / %" PRIuS "\n",
printf("Maximum bit-rate/payloadsize %0.3f / %" RTC_PRIuS "\n",
maxStreamLen * 8 / 0.03, maxStreamLen);
printf("Measured packet-loss........ %0.1f%% \n",
100.0f * (float)lostPacketCntr / (float)packetCntr);

3
modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -70,7 +70,8 @@ class OpusFecTest : public TestWithParam<coding_param> {
void OpusFecTest::SetUp() {
channels_ = get<0>(GetParam());
bit_rate_ = get<1>(GetParam());
printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
printf("Coding %" RTC_PRIuS " channel signal at %d bps.\n", channels_,
bit_rate_);
in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));

2
modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
View File

@ -99,7 +99,7 @@ void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
size_t time_now_ms = 0;
float time_ms;
printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
printf("Coding %d kHz-sampled %" RTC_PRIuS "-channel audio at %d bps ...\n",
input_sampling_khz_, channels_, bit_rate_);
while (time_now_ms < audio_duration_sec * 1000) {

8
modules/audio_device/android/audio_device_unittest.cc
View File

@ -187,7 +187,7 @@ class FifoAudioStream : public AudioStreamInterface {
const size_t size = fifo_->size();
if (size > largest_size_) {
largest_size_ = size;
PRINTD("(%" PRIuS ")", largest_size_);
PRINTD("(%" RTC_PRIuS ")", largest_size_);
}
total_written_elements_ += size;
}
@ -532,12 +532,12 @@ 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: %" PRIuS " [bytes]\n", bytes);
PRINT("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
PRINT("file size: %" RTC_PRIuS " [bytes]\n", bytes);
PRINT("file size: %" RTC_PRIuS " [samples]\n", bytes / kBytesPerSample);
const int seconds =
static_cast<int>(bytes / (sample_rate * kBytesPerSample));
PRINT("file size: %d [secs]\n", seconds);
PRINT("file size: %" PRIuS " [callbacks]\n",
PRINT("file size: %" RTC_PRIuS " [callbacks]\n",
seconds * kNumCallbacksPerSecond);
#endif
return file_name;

8
modules/audio_device/android/audio_manager_unittest.cc
View File

@ -153,16 +153,16 @@ TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
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: %" PRIuS "\n", kTag, playout_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
PRINT("%schannels: %" RTC_PRIuS "\n", kTag, playout_parameters_.channels());
PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.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: %" PRIuS "\n", kTag, record_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
PRINT("%schannels: %" RTC_PRIuS "\n", kTag, record_parameters_.channels());
PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
record_parameters_.frames_per_buffer(),
record_parameters_.GetBufferSizeInMilliseconds());
}

4
modules/audio_device/android/opensles_player.cc
View File

@ -192,7 +192,7 @@ 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(%" PRIuS ")", channels);
ALOGD("SetPlayoutChannels(%" RTC_PRIuS ")", channels);
audio_device_buffer_->SetPlayoutChannels(channels);
RTC_CHECK(audio_device_buffer_);
AllocateDataBuffers();
@ -213,7 +213,7 @@ 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: %" PRIuS, buffer_size_in_samples);
ALOGD("native buffer size: %" RTC_PRIuS, buffer_size_in_samples);
ALOGD("native buffer size in ms: %.2f",
audio_parameters_.GetBufferSizeInMilliseconds());
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);

8
modules/audio_device/android/opensles_recorder.cc
View File

@ -177,7 +177,7 @@ 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(%" PRIuS ")", channels);
ALOGD("SetRecordingChannels(%" RTC_PRIuS ")", channels);
audio_device_buffer_->SetRecordingChannels(channels);
// Allocated memory for internal data buffers given existing audio parameters.
AllocateDataBuffers();
@ -333,11 +333,11 @@ 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: %" PRIuS,
ALOGD("frames per native buffer: %" RTC_PRIuS,
audio_parameters_.frames_per_buffer());
ALOGD("frames per 10ms buffer: %" PRIuS,
ALOGD("frames per 10ms buffer: %" RTC_PRIuS,
audio_parameters_.frames_per_10ms_buffer());
ALOGD("bytes per native buffer: %" PRIuS,
ALOGD("bytes per native buffer: %" RTC_PRIuS,
audio_parameters_.GetBytesPerBuffer());
ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
RTC_DCHECK(audio_device_buffer_);

2
modules/remote_bitrate_estimator/tools/rtp_to_text.cc
View File

@ -47,7 +47,7 @@ int main(int argc, char* argv[]) {
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 %" PRIuS " %" PRIuS "\n",
fprintf(stdout, "%u %u %d %u %u %d %u %" RTC_PRIuS " %" RTC_PRIuS "\n",
header.sequenceNumber, header.timestamp,
header.extension.transmissionTimeOffset,
header.extension.absoluteSendTime, packet.time_ms,

17
rtc_base/format_macros.h
View File

@ -21,7 +21,7 @@
//
// To print a size_t value in a portable way:
// size_t size;
// printf("xyz: %" PRIuS, size);
// printf("xyz: %" RTC_PRIuS, size);
// The "u" in the macro corresponds to %u, and S is for "size".
#if defined(WEBRTC_POSIX)
@ -39,14 +39,16 @@
#include "rtc_base/system/arch.h"
#if !defined(PRIuS)
#define PRIuS "zu"
#endif
#define RTC_PRIuS "zu"
#else // WEBRTC_WIN
#include <inttypes.h>
// These are being defined without the RTC_ prefix because this is just filling
// the holes from what's supposed to be already present as part of the C
// standard, but missing on older MSVC versions.
#if !defined(PRId64)
#define PRId64 "I64d"
#endif
@ -59,9 +61,10 @@
#define PRIx64 "I64x"
#endif
#if !defined(PRIuS)
#define PRIuS "Iu"
#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

6
rtc_tools/unpack_aecdump/unpack.cc
View File

@ -463,13 +463,13 @@ int do_main(int argc, char* argv[]) {
fprintf(settings_file, " Reverse sample rate: %d\n",
reverse_sample_rate);
num_input_channels = msg.num_input_channels();
fprintf(settings_file, " Input channels: %" PRIuS "\n",
fprintf(settings_file, " Input channels: %" RTC_PRIuS "\n",
num_input_channels);
num_output_channels = msg.num_output_channels();
fprintf(settings_file, " Output channels: %" PRIuS "\n",
fprintf(settings_file, " Output channels: %" RTC_PRIuS "\n",
num_output_channels);
num_reverse_channels = msg.num_reverse_channels();
fprintf(settings_file, " Reverse channels: %" PRIuS "\n",
fprintf(settings_file, " Reverse channels: %" RTC_PRIuS "\n",
num_reverse_channels);
if (msg.has_timestamp_ms()) {
const int64_t timestamp = msg.timestamp_ms();

16
sdk/android/native_unittests/audio_device/audio_device_unittest.cc
View File

@ -183,7 +183,7 @@ class FifoAudioStream : public AudioStreamInterface {
const size_t size = fifo_->size();
if (size > largest_size_) {
largest_size_ = size;
PRINTD("(%" PRIuS ")", largest_size_);
PRINTD("(%" RTC_PRIuS ")", largest_size_);
}
total_written_elements_ += size;
}
@ -546,12 +546,12 @@ 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: %" PRIuS " [bytes]\n", bytes);
PRINT("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
PRINT("file size: %" RTC_PRIuS " [bytes]\n", bytes);
PRINT("file size: %" RTC_PRIuS " [samples]\n", bytes / kBytesPerSample);
const int seconds =
static_cast<int>(bytes / (sample_rate * kBytesPerSample));
PRINT("file size: %d [secs]\n", seconds);
PRINT("file size: %" PRIuS " [callbacks]\n",
PRINT("file size: %" RTC_PRIuS " [callbacks]\n",
seconds * kNumCallbacksPerSecond);
#endif
return file_name;
@ -971,16 +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: %" PRIuS "\n", kTag, output_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
PRINT("%schannels: %" RTC_PRIuS "\n", kTag, output_parameters_.channels());
PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.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: %" PRIuS "\n", kTag, input_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
PRINT("%schannels: %" RTC_PRIuS "\n", kTag, input_parameters_.channels());
PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.2f ms\n", kTag,
input_parameters_.frames_per_buffer(),
input_parameters_.GetBufferSizeInMilliseconds());
}

4
sdk/android/src/jni/audio_device/opensles_player.cc
View File

@ -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(%" PRIuS ")", channels);
ALOGD("SetPlayoutChannels(%" RTC_PRIuS ")", 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: %" PRIuS, buffer_size_in_samples);
ALOGD("native buffer size: %" RTC_PRIuS, buffer_size_in_samples);
ALOGD("native buffer size in ms: %.2f",
audio_parameters_.GetBufferSizeInMilliseconds());
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);

8
sdk/android/src/jni/audio_device/opensles_recorder.cc
View File

@ -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(%" PRIuS ")", channels);
ALOGD("SetRecordingChannels(%" RTC_PRIuS ")", channels);
audio_device_buffer_->SetRecordingChannels(channels);
// Allocated memory for internal data buffers given existing audio parameters.
AllocateDataBuffers();
@ -345,11 +345,11 @@ 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: %" PRIuS,
ALOGD("frames per native buffer: %" RTC_PRIuS,
audio_parameters_.frames_per_buffer());
ALOGD("frames per 10ms buffer: %" PRIuS,
ALOGD("frames per 10ms buffer: %" RTC_PRIuS,
audio_parameters_.frames_per_10ms_buffer());
ALOGD("bytes per native buffer: %" PRIuS,
ALOGD("bytes per native buffer: %" RTC_PRIuS,
audio_parameters_.GetBytesPerBuffer());
ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
RTC_DCHECK(audio_device_buffer_);

4
test/rtp_file_reader.cc
View File

@ -287,14 +287,14 @@ class PcapReader : public RtpFileReaderImpl {
}
printf("Total packets in file: %d\n", total_packet_count);
printf("Total RTP/RTCP packets: %" PRIuS "\n", packets_.size());
printf("Total RTP/RTCP packets: %" RTC_PRIuS "\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;
uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType;
printf("SSRC: %08x, %" PRIuS " packets, pt=%d\n", ssrc,
printf("SSRC: %08x, %" RTC_PRIuS " packets, pt=%d\n", ssrc,
packet_indices.size(), pt);
}

4
video/video_analyzer.cc
View File

@ -844,7 +844,7 @@ void VideoAnalyzer::PrintSamplesToFile() {
});
fprintf(out, "%s\n", graph_title_.c_str());
fprintf(out, "%" PRIuS "\n", samples_.size());
fprintf(out, "%" RTC_PRIuS "\n", samples_.size());
fprintf(out,
"dropped "
"input_time_ms "
@ -857,7 +857,7 @@ void VideoAnalyzer::PrintSamplesToFile() {
"encode_time_ms\n");
for (const Sample& sample : samples_) {
fprintf(out,
"%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS
"%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" RTC_PRIuS
" %lf %lf\n",
sample.dropped, sample.input_time_ms, sample.send_time_ms,
sample.recv_time_ms, sample.render_time_ms,