Lint clean video/ and add lint presubmit check.
BUG=webrtc:5316
Review URL: https://codereview.webrtc.org/1507643004
Cr-Commit-Position: refs/heads/master@{#10953}
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index d7c5ea2..be26269 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -19,6 +19,7 @@
'webrtc/audio',
'webrtc/call',
'webrtc/modules/video_processing',
+ 'webrtc/video',
]
# List of directories of "supported" native APIs. That means changes to headers
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 0a5e5df..26818f0 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -48,7 +48,7 @@
namespace webrtc {
-static const unsigned long kSilenceTimeoutMs = 2000;
+static const uint32_t kSilenceTimeoutMs = 2000;
class EndToEndTest : public test::CallTest {
public:
@@ -558,7 +558,7 @@
TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {
class FecNackObserver : public test::EndToEndTest {
public:
- explicit FecNackObserver()
+ FecNackObserver()
: EndToEndTest(kDefaultTimeoutMs),
state_(kFirstPacket),
fec_sequence_number_(0),
@@ -3047,7 +3047,7 @@
}
bool sender_done = false;
bool receiver_done = false;
- while(!sender_done || !receiver_done) {
+ while (!sender_done || !receiver_done) {
packet_event_->Wait(kSilenceTimeoutMs);
int64_t time_now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&test_crit_);
@@ -3135,7 +3135,7 @@
TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
class UnusedEncoder : public test::FakeEncoder {
- public:
+ public:
UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 21b4423..0e9bd30 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -322,9 +322,9 @@
std::string RampUpDownUpTester::GetModifierString() const {
std::string str("_");
- char temp_str[5];
- sprintf(temp_str, "%i", static_cast<int>(num_streams_));
- str += std::string(temp_str);
+ std::ostringstream s;
+ s << num_streams_;
+ str += s.str();
str += "stream";
str += (num_streams_ > 1 ? "s" : "");
str += "_";
diff --git a/webrtc/video/send_statistics_proxy.h b/webrtc/video/send_statistics_proxy.h
index c098e3a..e9e89f6 100644
--- a/webrtc/video/send_statistics_proxy.h
+++ b/webrtc/video/send_statistics_proxy.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_
#define WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_
+#include <map>
#include <string>
#include "webrtc/base/criticalsection.h"
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 09c1037..d262689 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -13,6 +13,7 @@
#include <deque>
#include <map>
#include <sstream>
+#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
@@ -97,7 +98,6 @@
thread->Start();
comparison_thread_pool_.push_back(thread);
}
-
}
~VideoAnalyzer() {
@@ -1017,14 +1017,16 @@
test::VideoRenderer::Create("Local Preview", params_.common.width,
params_.common.height));
size_t stream_id = params_.ss.selected_stream;
- char title[32];
- if (params_.ss.streams.size() == 1) {
- sprintf(title, "Loopback Video");
- } else {
- sprintf(title, "Loopback Video - Stream #%" PRIuS, stream_id);
+ std::string title = "Loopback Video";
+ if (params_.ss.streams.size() > 1) {
+ std::ostringstream s;
+ s << stream_id;
+ title += " - Stream #" + s.str();
}
+
rtc::scoped_ptr<test::VideoRenderer> loopback_video(
- test::VideoRenderer::Create(title, params_.ss.streams[stream_id].width,
+ test::VideoRenderer::Create(title.c_str(),
+ params_.ss.streams[stream_id].width,
params_.ss.streams[stream_id].height));
// TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
diff --git a/webrtc/video/video_quality_test.h b/webrtc/video/video_quality_test.h
index b88c513..dd2b011 100644
--- a/webrtc/video/video_quality_test.h
+++ b/webrtc/video/video_quality_test.h
@@ -11,6 +11,7 @@
#define WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
#include <string>
+#include <vector>
#include "webrtc/test/call_test.h"
#include "webrtc/test/frame_generator.h"
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 5bc79d6..9d1664e 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -12,6 +12,7 @@
#include <stdlib.h>
+#include <set>
#include <string>
#include "webrtc/base/checks.h"
@@ -111,7 +112,7 @@
memset(&codec, 0, sizeof(codec));
codec.plType = decoder.payload_type;
- strcpy(codec.plName, decoder.payload_name.c_str());
+ strncpy(codec.plName, decoder.payload_name.c_str(), sizeof(codec.plName));
if (decoder.payload_name == "VP8") {
codec.codecType = kVideoCodecVP8;
} else if (decoder.payload_name == "VP9") {
@@ -228,7 +229,7 @@
VideoCodec codec;
memset(&codec, 0, sizeof(codec));
codec.codecType = kVideoCodecULPFEC;
- strcpy(codec.plName, "ulpfec");
+ strncpy(codec.plName, "ulpfec", sizeof(codec.plName));
codec.plType = config_.rtp.fec.ulpfec_payload_type;
RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
}
@@ -236,7 +237,7 @@
VideoCodec codec;
memset(&codec, 0, sizeof(codec));
codec.codecType = kVideoCodecRED;
- strcpy(codec.plName, "red");
+ strncpy(codec.plName, "red", sizeof(codec.plName));
codec.plType = config_.rtp.fec.red_payload_type;
RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
if (config_.rtp.fec.red_rtx_payload_type != -1) {
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 08b0f19..d73ef9c 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -400,8 +400,8 @@
RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
RTC_DCHECK_GE(streams[i].max_qp, 0);
- sim_stream->width = static_cast<unsigned short>(streams[i].width);
- sim_stream->height = static_cast<unsigned short>(streams[i].height);
+ sim_stream->width = static_cast<uint16_t>(streams[i].width);
+ sim_stream->height = static_cast<uint16_t>(streams[i].height);
sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
@@ -410,12 +410,12 @@
streams[i].temporal_layer_thresholds_bps.size() + 1);
video_codec.width = std::max(video_codec.width,
- static_cast<unsigned short>(streams[i].width));
+ static_cast<uint16_t>(streams[i].width));
video_codec.height = std::max(
- video_codec.height, static_cast<unsigned short>(streams[i].height));
+ video_codec.height, static_cast<uint16_t>(streams[i].height));
video_codec.minBitrate =
- std::min(video_codec.minBitrate,
- static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
+ std::min(static_cast<uint16_t>(video_codec.minBitrate),
+ static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
video_codec.qpMax = std::max(video_codec.qpMax,
static_cast<unsigned int>(streams[i].max_qp));
@@ -500,7 +500,7 @@
std::map<uint32_t, RtpState> rtp_states;
for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_.rtp.ssrcs[i];
- rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc( ssrc);
+ rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
}
for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 0032230..6cdb66e 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1122,7 +1122,7 @@
}
void WaitOutputFrame() {
- const unsigned long kWaitFrameTimeoutMs = 3000;
+ const uint32_t kWaitFrameTimeoutMs = 3000;
EXPECT_EQ(kEventSignaled, output_frame_event_->Wait(kWaitFrameTimeoutMs))
<< "Timeout while waiting for output frames.";
}
@@ -1423,7 +1423,6 @@
template <typename T>
class VideoCodecConfigObserver : public test::SendTest,
public test::FakeEncoder {
-
public:
VideoCodecConfigObserver(VideoCodecType video_codec_type,
const char* codec_name)
