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Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
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28
webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
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@ -20,6 +20,7 @@
#endif
#include "gflags/gflags.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
#include "webrtc/modules/video_coding/codecs/test/stats.h"
@ -204,7 +205,8 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
FLAGS_packet_size);
return 7;
}
config->networking_config.packet_size_in_bytes = FLAGS_packet_size;
config->networking_config.packet_size_in_bytes =
static_cast<size_t>(FLAGS_packet_size);
if (FLAGS_max_payload_size <= 0) {
fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
@ -212,7 +214,7 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
return 8;
}
config->networking_config.max_payload_size_in_bytes =
FLAGS_max_payload_size;
static_cast<size_t>(FLAGS_max_payload_size);
// Check the width and height
if (FLAGS_width <= 0 || FLAGS_height <= 0) {
@ -290,10 +292,10 @@ void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
Log(" Input filename : %s\n", config.input_filename.c_str());
Log(" Output directory : %s\n", config.output_dir.c_str());
Log(" Output filename : %s\n", config.output_filename.c_str());
Log(" Frame length : %d bytes\n", config.frame_length_in_bytes);
Log(" Packet size : %d bytes\n",
Log(" Frame length : %" PRIuS " bytes\n", config.frame_length_in_bytes);
Log(" Packet size : %" PRIuS " bytes\n",
config.networking_config.packet_size_in_bytes);
Log(" Max payload size : %d bytes\n",
Log(" Max payload size : %" PRIuS " bytes\n",
config.networking_config.max_payload_size_in_bytes);
Log(" Packet loss:\n");
Log(" Mode : %s\n",
@ -320,8 +322,8 @@ void PrintCsvOutput(const webrtc::test::Stats& stats,
const webrtc::test::FrameStatistic& f = stats.stats_[i];
const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7d, %d, %2d, %2d, "
"%5.3f, %5.2f\n",
printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7" PRIuS ", %d, %2d, %2"
PRIuS ", %5.3f, %5.2f\n",
f.frame_number,
f.encoding_successful,
f.decoding_successful,
@ -352,13 +354,13 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
"{'name': 'input_filename', 'value': '%s'},\n"
"{'name': 'output_filename', 'value': '%s'},\n"
"{'name': 'output_dir', 'value': '%s'},\n"
"{'name': 'packet_size_in_bytes', 'value': '%d'},\n"
"{'name': 'max_payload_size_in_bytes', 'value': '%d'},\n"
"{'name': 'packet_size_in_bytes', 'value': '%" PRIuS "'},\n"
"{'name': 'max_payload_size_in_bytes', 'value': '%" PRIuS "'},\n"
"{'name': 'packet_loss_mode', 'value': '%s'},\n"
"{'name': 'packet_loss_probability', 'value': '%f'},\n"
"{'name': 'packet_loss_burst_length', 'value': '%d'},\n"
"{'name': 'exclude_frame_types', 'value': '%s'},\n"
"{'name': 'frame_length_in_bytes', 'value': '%d'},\n"
"{'name': 'frame_length_in_bytes', 'value': '%" PRIuS "'},\n"
"{'name': 'use_single_core', 'value': '%s'},\n"
"{'name': 'keyframe_interval;', 'value': '%d'},\n"
"{'name': 'video_codec_type', 'value': '%s'},\n"
@ -411,9 +413,9 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
"'encoding_successful': %s, 'decoding_successful': %s, "
"'encode_time': %d, 'decode_time': %d, "
"'encode_return_code': %d, 'decode_return_code': %d, "
"'bit_rate': %d, 'encoded_frame_length': %d, 'frame_type': %s, "
"'packets_dropped': %d, 'total_packets': %d, "
"'ssim': %f, 'psnr': %f},\n",
"'bit_rate': %d, 'encoded_frame_length': %" PRIuS ", "
"'frame_type': %s, 'packets_dropped': %d, "
"'total_packets': %" PRIuS ", 'ssim': %f, 'psnr': %f},\n",
f.frame_number,
f.encoding_successful ? "True " : "False",
f.decoding_successful ? "True " : "False",