diff options
Diffstat (limited to 'webrtc/tools')
-rw-r--r-- | webrtc/tools/agc/activity_metric.cc | 6 | ||||
-rw-r--r-- | webrtc/tools/agc/agc_harness.cc | 7 | ||||
-rw-r--r-- | webrtc/tools/agc/test_utils.cc | 2 | ||||
-rw-r--r-- | webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc | 2 | ||||
-rw-r--r-- | webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc | 2 | ||||
-rw-r--r-- | webrtc/tools/frame_analyzer/video_quality_analysis.cc | 2 | ||||
-rw-r--r-- | webrtc/tools/frame_editing/frame_editing_lib.cc | 1 | ||||
-rw-r--r-- | webrtc/tools/frame_editing/frame_editing_lib.h | 6 | ||||
-rw-r--r-- | webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc | 6 | ||||
-rw-r--r-- | webrtc/tools/rtcbot/OWNERS | 1 |
10 files changed, 18 insertions, 17 deletions
diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc index 18e7c6dad8..2cb0a1b2df 100644 --- a/webrtc/tools/agc/activity_metric.cc +++ b/webrtc/tools/agc/activity_metric.cc @@ -24,7 +24,7 @@ #include "webrtc/modules/audio_processing/vad/common.h" #include "webrtc/modules/audio_processing/vad/pitch_based_vad.h" #include "webrtc/modules/audio_processing/vad/standalone_vad.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" static const int kAgcAnalWindowSamples = 100; static const double kDefaultActivityThreshold = 0.3; @@ -56,7 +56,7 @@ namespace webrtc { // silence frame. Otherwise true VAD would drift with respect to the audio. // We only consider mono inputs. static void DitherSilence(AudioFrame* frame) { - ASSERT_EQ(1, frame->num_channels_); + ASSERT_EQ(1u, frame->num_channels_); const double kRmsSilence = 5; const double sum_squared_silence = kRmsSilence * kRmsSilence * frame->samples_per_channel_; @@ -65,7 +65,7 @@ static void DitherSilence(AudioFrame* frame) { sum_squared += frame->data_[n] * frame->data_[n]; if (sum_squared <= sum_squared_silence) { for (size_t n = 0; n < frame->samples_per_channel_; n++) - frame->data_[n] = (rand() & 0xF) - 8; + frame->data_[n] = (rand() & 0xF) - 8; // NOLINT: ignore non-threadsafe. } } diff --git a/webrtc/tools/agc/agc_harness.cc b/webrtc/tools/agc/agc_harness.cc index 73e1e09935..0d35d4b56a 100644 --- a/webrtc/tools/agc/agc_harness.cc +++ b/webrtc/tools/agc/agc_harness.cc @@ -12,10 +12,11 @@ #include "gflags/gflags.h" #include "webrtc/base/checks.h" +#include "webrtc/base/format_macros.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/system_wrappers/include/trace.h" -#include "webrtc/test/channel_transport/include/channel_transport.h" +#include "webrtc/test/channel_transport/channel_transport.h" #include "webrtc/test/testsupport/trace_to_stderr.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" @@ -176,8 +177,8 @@ class AgcVoiceEngine { printf("Codecs:\n"); for (int i = 0; i < codec_->NumOfCodecs(); i++) { RTC_CHECK_EQ(0, codec_->GetCodec(i, params)); - printf("%d %s/%d/%d\n", params.pltype, params.plname, params.plfreq, - params.channels); + printf("%d %s/%d/%" PRIuS "\n", params.pltype, params.plname, + params.plfreq, params.channels); } } diff --git a/webrtc/tools/agc/test_utils.cc b/webrtc/tools/agc/test_utils.cc index 81819c598e..a0ed74732d 100644 --- a/webrtc/tools/agc/test_utils.cc +++ b/webrtc/tools/agc/test_utils.cc @@ -14,7 +14,7 @@ #include <algorithm> -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc b/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc index 1eb8925537..2594fd1317 100644 --- a/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc +++ b/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc @@ -16,7 +16,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/test/channel_transport/include/channel_transport.h" +#include "webrtc/test/channel_transport/channel_transport.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_codec.h" diff --git a/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc b/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc index 570fa0ad24..b6b1596866 100644 --- a/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc +++ b/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc @@ -13,7 +13,7 @@ #include <stdio.h> #include "webrtc/base/scoped_ptr.h" -#include "webrtc/test/channel_transport/include/channel_transport.h" +#include "webrtc/test/channel_transport/channel_transport.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_volume_control.h" diff --git a/webrtc/tools/frame_analyzer/video_quality_analysis.cc b/webrtc/tools/frame_analyzer/video_quality_analysis.cc index 172baa72b8..dfd57f1961 100644 --- a/webrtc/tools/frame_analyzer/video_quality_analysis.cc +++ b/webrtc/tools/frame_analyzer/video_quality_analysis.cc @@ -227,7 +227,7 @@ void RunAnalysis(const char* reference_file_name, const char* test_file_name, ResultsContainer* results) { // Check if the reference_file_name ends with "y4m". bool y4m_mode = false; - if (std::string(reference_file_name).find("y4m") != std::string::npos){ + if (std::string(reference_file_name).find("y4m") != std::string::npos) { y4m_mode = true; } diff --git a/webrtc/tools/frame_editing/frame_editing_lib.cc b/webrtc/tools/frame_editing/frame_editing_lib.cc index 79c6033a30..90855a354c 100644 --- a/webrtc/tools/frame_editing/frame_editing_lib.cc +++ b/webrtc/tools/frame_editing/frame_editing_lib.cc @@ -15,6 +15,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" +#include "webrtc/tools/frame_editing/frame_editing_lib.h" #include "webrtc/typedefs.h" using std::string; diff --git a/webrtc/tools/frame_editing/frame_editing_lib.h b/webrtc/tools/frame_editing/frame_editing_lib.h index 245d60f376..94595c43bb 100644 --- a/webrtc/tools/frame_editing/frame_editing_lib.h +++ b/webrtc/tools/frame_editing/frame_editing_lib.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_ -#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_ +#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_ +#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_ #include <string> @@ -36,4 +36,4 @@ int EditFrames(const std::string& in_path, int width, int height, int last_frame_to_process, const std::string& out_path); } // namespace webrtc -#endif // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_ +#endif // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_ diff --git a/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc b/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc index bae145a78f..737661c3df 100644 --- a/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc +++ b/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc @@ -25,7 +25,7 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name, const char* results_file_name, int width, int height) { // Check if the reference_file_name ends with "y4m". bool y4m_mode = false; - if (std::string(reference_file_name).find("y4m") != std::string::npos){ + if (std::string(reference_file_name).find("y4m") != std::string::npos) { y4m_mode = true; } @@ -38,8 +38,8 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name, uint8_t* ref_frame = new uint8_t[size]; bool read_result = true; - for(int frame_counter = 0; frame_counter < MAX_NUM_FRAMES_PER_FILE; - ++frame_counter){ + for (int frame_counter = 0; frame_counter < MAX_NUM_FRAMES_PER_FILE; + ++frame_counter) { read_result &= (y4m_mode) ? webrtc::test::ExtractFrameFromY4mFile( reference_file_name, width, height, frame_counter, ref_frame): webrtc::test::ExtractFrameFromYuvFile(reference_file_name, width, diff --git a/webrtc/tools/rtcbot/OWNERS b/webrtc/tools/rtcbot/OWNERS index efdce51ca6..296f71fffc 100644 --- a/webrtc/tools/rtcbot/OWNERS +++ b/webrtc/tools/rtcbot/OWNERS @@ -1,2 +1 @@ andresp@webrtc.org -houssainy@google.com |