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-rw-r--r--webrtc/tools/agc/activity_metric.cc6
-rw-r--r--webrtc/tools/agc/agc_harness.cc7
-rw-r--r--webrtc/tools/agc/test_utils.cc2
-rw-r--r--webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc2
-rw-r--r--webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc2
-rw-r--r--webrtc/tools/frame_analyzer/video_quality_analysis.cc2
-rw-r--r--webrtc/tools/frame_editing/frame_editing_lib.cc1
-rw-r--r--webrtc/tools/frame_editing/frame_editing_lib.h6
-rw-r--r--webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc6
-rw-r--r--webrtc/tools/rtcbot/OWNERS1
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