From b1686786e896b55fb3cb21aa5cf578781d1c4039 Mon Sep 17 00:00:00 2001 From: Oleh Prypin Date: Fri, 2 Aug 2019 09:36:47 +0200 Subject: [PATCH] Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 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 Reviewed-by: Karl Wiberg Reviewed-by: Niels Moller Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#28794} --- audio/remix_resample_unittest.cc | 2 +- common_audio/audio_converter_unittest.cc | 4 ++-- .../test/ReleaseTest-API/ReleaseTest-API.cc | 12 +++++++----- .../codecs/isac/main/test/simpleKenny.c | 2 +- .../audio_coding/codecs/opus/opus_fec_test.cc | 3 ++- .../codecs/tools/audio_codec_speed_test.cc | 2 +- .../android/audio_device_unittest.cc | 8 ++++---- .../android/audio_manager_unittest.cc | 8 ++++---- modules/audio_device/android/opensles_player.cc | 4 ++-- .../audio_device/android/opensles_recorder.cc | 8 ++++---- .../tools/rtp_to_text.cc | 2 +- rtc_base/format_macros.h | 17 ++++++++++------- rtc_tools/unpack_aecdump/unpack.cc | 6 +++--- .../audio_device/audio_device_unittest.cc | 16 ++++++++-------- .../src/jni/audio_device/opensles_player.cc | 4 ++-- .../src/jni/audio_device/opensles_recorder.cc | 8 ++++---- test/rtp_file_reader.cc | 4 ++-- video/video_analyzer.cc | 4 ++-- 18 files changed, 60 insertions(+), 54 deletions(-) diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc index 55f811be35..d2155a64f0 100644 --- a/audio/remix_resample_unittest.cc +++ b/audio/remix_resample_unittest.cc @@ -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; } diff --git a/common_audio/audio_converter_unittest.cc b/common_audio/audio_converter_unittest.cc index 9f49497bc3..84d8f5568e 100644 --- a/common_audio/audio_converter_unittest.cc +++ b/common_audio/audio_converter_unittest.cc @@ -79,7 +79,7 @@ float ComputeSNR(const ChannelBuffer& 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 converter = AudioConverter::Create( 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 934794d334..de97d22a8d 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 @@ -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) { diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c index 4b48b5033a..f5d8e4f3a8 100644 --- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c +++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c @@ -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); diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc index 7f133803ed..1ab4d8650b 100644 --- a/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -70,7 +70,8 @@ class OpusFecTest : public TestWithParam { 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())); 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 1e6b4f0a13..3d5ba0b7c8 100644 --- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc +++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc @@ -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) { diff --git a/modules/audio_device/android/audio_device_unittest.cc b/modules/audio_device/android/audio_device_unittest.cc index 9449015acb..e2c6800f38 100644 --- a/modules/audio_device/android/audio_device_unittest.cc +++ b/modules/audio_device/android/audio_device_unittest.cc @@ -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(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; diff --git a/modules/audio_device/android/audio_manager_unittest.cc b/modules/audio_device/android/audio_manager_unittest.cc index 4abba51591..1b81904c34 100644 --- a/modules/audio_device/android/audio_manager_unittest.cc +++ b/modules/audio_device/android/audio_manager_unittest.cc @@ -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()); } diff --git a/modules/audio_device/android/opensles_player.cc b/modules/audio_device/android/opensles_player.cc index 509e51a1d5..61365703b2 100644 --- a/modules/audio_device/android/opensles_player.cc +++ b/modules/audio_device/android/opensles_player.cc @@ -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(audio_device_buffer_); diff --git a/modules/audio_device/android/opensles_recorder.cc b/modules/audio_device/android/opensles_recorder.cc index ed81561c68..05b5581912 100644 --- a/modules/audio_device/android/opensles_recorder.cc +++ b/modules/audio_device/android/opensles_recorder.cc @@ -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_); diff --git a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc index 57ad1375c3..c362623067 100644 --- a/modules/remote_bitrate_estimator/tools/rtp_to_text.cc +++ b/modules/remote_bitrate_estimator/tools/rtp_to_text.cc @@ -47,7 +47,7 @@ int main(int argc, char* argv[]) { ss << static_cast(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, diff --git a/rtc_base/format_macros.h b/rtc_base/format_macros.h index 0466770de5..998f5fe845 100644 --- a/rtc_base/format_macros.h +++ b/rtc_base/format_macros.h @@ -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 +// 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 diff --git a/rtc_tools/unpack_aecdump/unpack.cc b/rtc_tools/unpack_aecdump/unpack.cc index c9da63ad6a..ba3af129bf 100644 --- a/rtc_tools/unpack_aecdump/unpack.cc +++ b/rtc_tools/unpack_aecdump/unpack.cc @@ -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(); 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 c1353d2d79..da7790c0a0 100644 --- a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc +++ b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc @@ -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(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()); } diff --git a/sdk/android/src/jni/audio_device/opensles_player.cc b/sdk/android/src/jni/audio_device/opensles_player.cc index ce43d1ab9e..55030fcd28 100644 --- a/sdk/android/src/jni/audio_device/opensles_player.cc +++ b/sdk/android/src/jni/audio_device/opensles_player.cc @@ -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(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 f244690258..ac0d71a0dd 100644 --- a/sdk/android/src/jni/audio_device/opensles_recorder.cc +++ b/sdk/android/src/jni/audio_device/opensles_recorder.cc @@ -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_); diff --git a/test/rtp_file_reader.cc b/test/rtp_file_reader.cc index 40a5cff96e..9454d6fc66 100644 --- a/test/rtp_file_reader.cc +++ b/test/rtp_file_reader.cc @@ -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& 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); } diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 018d7ac5d0..2a2e1a41cb 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -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,