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,