Lint fix for webrtc/modules/video_coding PART 3!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)
BUG=webrtc:5309
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1540243002
Cr-Commit-Position: refs/heads/master@{#11105}
diff --git a/webrtc/modules/video_coding/test/receiver_tests.h b/webrtc/modules/video_coding/test/receiver_tests.h
index 9b9b377..d6bac07 100644
--- a/webrtc/modules/video_coding/test/receiver_tests.h
+++ b/webrtc/modules/video_coding/test/receiver_tests.h
@@ -11,6 +11,9 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+#include <stdio.h>
+#include <string>
+
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -19,12 +22,9 @@
#include "webrtc/modules/video_coding/test/video_source.h"
#include "webrtc/typedefs.h"
-#include <stdio.h>
-#include <string>
-
class RtpDataCallback : public webrtc::NullRtpData {
public:
- RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
+ explicit RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
virtual ~RtpDataCallback() {}
int32_t OnReceivedPayloadData(
@@ -40,4 +40,4 @@
int RtpPlay(const CmdArgs& args);
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
diff --git a/webrtc/modules/video_coding/test/rtp_player.cc b/webrtc/modules/video_coding/test/rtp_player.cc
index c9af450..9b64906 100644
--- a/webrtc/modules/video_coding/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/test/rtp_player.cc
@@ -26,9 +26,9 @@
#include "webrtc/test/rtp_file_reader.h"
#if 1
-# define DEBUG_LOG1(text, arg)
+#define DEBUG_LOG1(text, arg)
#else
-# define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
+#define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
#endif
namespace webrtc {
@@ -41,7 +41,9 @@
class RawRtpPacket {
public:
- RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
+ RawRtpPacket(const uint8_t* data,
+ size_t length,
+ uint32_t ssrc,
uint16_t seq_num)
: data_(new uint8_t[length]),
length_(length),
@@ -140,7 +142,7 @@
CriticalSectionScoped cs(crit_sect_.get());
int count = 0;
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
- ++it) {
+ ++it) {
if ((*it)->resend_time_ms() >= 0) {
count++;
}
@@ -164,7 +166,7 @@
printf("Packets still lost: %zd\n", packets_.size());
printf("Sequence numbers:\n");
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
- ++it) {
+ ++it) {
printf("%u, ", (*it)->seq_num());
}
printf("\n");
@@ -231,17 +233,14 @@
kDefaultTransmissionTimeOffsetExtensionId);
for (PayloadTypesIterator it = payload_types_.begin();
- it != payload_types_.end(); ++it) {
+ it != payload_types_.end(); ++it) {
VideoCodec codec;
memset(&codec, 0, sizeof(codec));
- strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName)-1);
+ strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName) - 1);
codec.plType = it->payload_type();
codec.codecType = it->codec_type();
- if (handler->rtp_module_->RegisterReceivePayload(codec.plName,
- codec.plType,
- 90000,
- 0,
- codec.maxBitrate) < 0) {
+ if (handler->rtp_module_->RegisterReceivePayload(
+ codec.plName, codec.plType, 90000, 0, codec.maxBitrate) < 0) {
return -1;
}
}
@@ -267,7 +266,8 @@
private:
class Handler : public RtpStreamInterface {
public:
- Handler(uint32_t ssrc, const PayloadTypes& payload_types,
+ Handler(uint32_t ssrc,
+ const PayloadTypes& payload_types,
LostPackets* lost_packets)
: rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(new RTPPayloadRegistry(
@@ -290,9 +290,7 @@
}
virtual uint32_t ssrc() const { return ssrc_; }
- virtual const PayloadTypes& payload_types() const {
- return payload_types_;
- }
+ virtual const PayloadTypes& payload_types() const { return payload_types_; }
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
@@ -351,8 +349,7 @@
virtual int NextPacket(int64_t time_now) {
// Send any packets ready to be resent.
for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
- packet != NULL;
- packet = lost_packets_.NextPacketToResend(time_now)) {
+ packet != NULL; packet = lost_packets_.NextPacketToResend(time_now)) {
int ret = SendPacket(packet->data(), packet->length());
if (ret > 0) {
printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());
@@ -392,8 +389,7 @@
if (!packet_source_->NextPacket(&next_packet_)) {
end_of_file_ = true;
return 0;
- }
- else if (next_packet_.length == 0) {
+ } else if (next_packet_.length == 0) {
return 0;
}
}
@@ -406,7 +402,7 @@
virtual uint32_t TimeUntilNextPacket() const {
int64_t time_left = (next_rtp_time_ - first_packet_rtp_time_) -
- (clock_->TimeInMilliseconds() - first_packet_time_ms_);
+ (clock_->TimeInMilliseconds() - first_packet_time_ms_);
if (time_left < 0) {
return 0;
}
@@ -438,7 +434,7 @@
if (no_loss_startup_ > 0) {
no_loss_startup_--;
- } else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
+ } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) { // NOLINT
uint16_t seq_num = header.sequenceNumber;
lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
@@ -470,9 +466,12 @@
};
RtpPlayerInterface* Create(const std::string& input_filename,
- PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
- const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
- bool reordering) {
+ PayloadSinkFactoryInterface* payload_sink_factory,
+ Clock* clock,
+ const PayloadTypes& payload_types,
+ float loss_rate,
+ int64_t rtt_ms,
+ bool reordering) {
rtc::scoped_ptr<test::RtpFileReader> packet_source(
test::RtpFileReader::Create(test::RtpFileReader::kRtpDump,
input_filename));
diff --git a/webrtc/modules/video_coding/test/rtp_player.h b/webrtc/modules/video_coding/test/rtp_player.h
index c227f1c..e50fb9a 100644
--- a/webrtc/modules/video_coding/test/rtp_player.h
+++ b/webrtc/modules/video_coding/test/rtp_player.h
@@ -24,12 +24,12 @@
class PayloadCodecTuple {
public:
- PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
+ PayloadCodecTuple(uint8_t payload_type,
+ const std::string& codec_name,
VideoCodecType codec_type)
: name_(codec_name),
payload_type_(payload_type),
- codec_type_(codec_type) {
- }
+ codec_type_(codec_type) {}
const std::string& name() const { return name_; }
uint8_t payload_type() const { return payload_type_; }
@@ -87,11 +87,14 @@
};
RtpPlayerInterface* Create(const std::string& inputFilename,
- PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
- const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
- bool reordering);
+ PayloadSinkFactoryInterface* payloadSinkFactory,
+ Clock* clock,
+ const PayloadTypes& payload_types,
+ float lossRate,
+ int64_t rttMs,
+ bool reordering);
} // namespace rtpplayer
} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
diff --git a/webrtc/modules/video_coding/test/stream_generator.cc b/webrtc/modules/video_coding/test/stream_generator.cc
index 304fe2e..167d55f 100644
--- a/webrtc/modules/video_coding/test/stream_generator.cc
+++ b/webrtc/modules/video_coding/test/stream_generator.cc
@@ -22,8 +22,7 @@
namespace webrtc {
StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
- : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {
-}
+ : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
packets_.clear();
@@ -41,8 +40,8 @@
const int packet_size =
(kFrameSize + num_media_packets / 2) / num_media_packets;
bool marker_bit = (i == num_media_packets - 1);
- packets_.push_back(GeneratePacket(
- sequence_number_, timestamp, packet_size, (i == 0), marker_bit, type));
+ packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
+ (i == 0), marker_bit, type));
++sequence_number_;
}
for (int i = 0; i < num_empty_packets; ++i) {
@@ -104,7 +103,9 @@
return true;
}
-void StreamGenerator::DropLastPacket() { packets_.pop_back(); }
+void StreamGenerator::DropLastPacket() {
+ packets_.pop_back();
+}
uint16_t StreamGenerator::NextSequenceNumber() const {
if (packets_.empty())
@@ -112,7 +113,9 @@
return packets_.front().seqNum;
}
-int StreamGenerator::PacketsRemaining() const { return packets_.size(); }
+int StreamGenerator::PacketsRemaining() const {
+ return packets_.size();
+}
std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
std::list<VCMPacket>::iterator it = packets_.begin();
diff --git a/webrtc/modules/video_coding/test/test_util.cc b/webrtc/modules/video_coding/test/test_util.cc
index fc670ad..7ff663e 100644
--- a/webrtc/modules/video_coding/test/test_util.cc
+++ b/webrtc/modules/video_coding/test/test_util.cc
@@ -28,12 +28,12 @@
rtt(0),
inputFile(webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv"),
outputFile(webrtc::test::OutputPath() +
- "video_coding_test_output_352x288.yuv") {
-}
+ "video_coding_test_output_352x288.yuv") {}
namespace {
-void SplitFilename(const std::string& filename, std::string* basename,
+void SplitFilename(const std::string& filename,
+ std::string* basename,
std::string* extension) {
assert(basename);
assert(extension);
@@ -41,7 +41,7 @@
std::string::size_type idx;
idx = filename.rfind('.');
- if(idx != std::string::npos) {
+ if (idx != std::string::npos) {
*basename = filename.substr(0, idx);
*extension = filename.substr(idx + 1);
} else {
@@ -50,21 +50,24 @@
}
}
-std::string AppendWidthHeightCount(const std::string& filename, int width,
- int height, int count) {
+std::string AppendWidthHeightCount(const std::string& filename,
+ int width,
+ int height,
+ int count) {
std::string basename;
std::string extension;
SplitFilename(filename, &basename, &extension);
std::stringstream ss;
- ss << basename << "_" << count << "." << width << "_" << height << "." <<
- extension;
+ ss << basename << "_" << count << "." << width << "_" << height << "."
+ << extension;
return ss.str();
}
} // namespace
FileOutputFrameReceiver::FileOutputFrameReceiver(
- const std::string& base_out_filename, uint32_t ssrc)
+ const std::string& base_out_filename,
+ uint32_t ssrc)
: out_filename_(),
out_file_(NULL),
timing_file_(NULL),
@@ -80,8 +83,8 @@
SplitFilename(base_out_filename, &basename, &extension);
}
std::stringstream ss;
- ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') <<
- ssrc << "." << extension;
+ ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') << ssrc
+ << "." << extension;
out_filename_ = ss.str();
}
@@ -113,8 +116,8 @@
printf("New size: %dx%d\n", video_frame.width(), video_frame.height());
width_ = video_frame.width();
height_ = video_frame.height();
- std::string filename_with_width_height = AppendWidthHeightCount(
- out_filename_, width_, height_, count_);
+ std::string filename_with_width_height =
+ AppendWidthHeightCount(out_filename_, width_, height_, count_);
++count_;
out_file_ = fopen(filename_with_width_height.c_str(), "wb");
if (out_file_ == NULL) {
@@ -122,7 +125,7 @@
}
}
fprintf(timing_file_, "%u, %u\n", video_frame.timestamp(),
- webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
+ webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
if (PrintVideoFrame(video_frame, out_file_) < 0) {
return -1;
}
@@ -130,7 +133,7 @@
}
webrtc::RtpVideoCodecTypes ConvertCodecType(const char* plname) {
- if (strncmp(plname,"VP8" , 3) == 0) {
+ if (strncmp(plname, "VP8", 3) == 0) {
return webrtc::kRtpVideoVp8;
} else {
// Default value.
diff --git a/webrtc/modules/video_coding/test/test_util.h b/webrtc/modules/video_coding/test/test_util.h
index 30f337d..45b88b9 100644
--- a/webrtc/modules/video_coding/test/test_util.h
+++ b/webrtc/modules/video_coding/test/test_util.h
@@ -33,11 +33,13 @@
virtual bool Reset() { return true; }
- virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
+ virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) { // NOLINT
return webrtc::kEventTimeout;
}
- virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
+ virtual bool StartTimer(bool periodic, unsigned long time) { // NOLINT
+ return true;
+ }
virtual bool StopTimer() { return true; }
};
@@ -46,9 +48,7 @@
public:
virtual ~NullEventFactory() {}
- virtual webrtc::EventWrapper* CreateEvent() {
- return new NullEvent;
- }
+ virtual webrtc::EventWrapper* CreateEvent() { return new NullEvent; }
};
class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
@@ -57,7 +57,7 @@
virtual ~FileOutputFrameReceiver();
// VCMReceiveCallback
- virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);
+ virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame); // NOLINT
private:
std::string out_filename_;
diff --git a/webrtc/modules/video_coding/test/tester_main.cc b/webrtc/modules/video_coding/test/tester_main.cc
index 0184ff1..33ca820 100644
--- a/webrtc/modules/video_coding/test/tester_main.cc
+++ b/webrtc/modules/video_coding/test/tester_main.cc
@@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-
#include <stdlib.h>
#include <string.h>
@@ -21,12 +20,15 @@
DEFINE_int32(width, 352, "Width in pixels of the frames in the input file.");
DEFINE_int32(height, 288, "Height in pixels of the frames in the input file.");
DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
-DEFINE_string(input_filename, webrtc::test::ProjectRootPath() +
- "/resources/foreman_cif.yuv", "Input file.");
-DEFINE_string(output_filename, webrtc::test::OutputPath() +
- "video_coding_test_output_352x288.yuv", "Output file.");
+DEFINE_string(input_filename,
+ webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv",
+ "Input file.");
+DEFINE_string(output_filename,
+ webrtc::test::OutputPath() +
+ "video_coding_test_output_352x288.yuv",
+ "Output file.");
-using namespace webrtc;
+namespace webrtc {
/*
* Build with EVENT_DEBUG defined
@@ -36,36 +38,37 @@
int vcmMacrosTests = 0;
int vcmMacrosErrors = 0;
-int ParseArguments(CmdArgs& args) {
- args.width = FLAGS_width;
- args.height = FLAGS_height;
- if (args.width < 1 || args.height < 1) {
+int ParseArguments(CmdArgs* args) {
+ args->width = FLAGS_width;
+ args->height = FLAGS_height;
+ if (args->width < 1 || args->height < 1) {
return -1;
}
- args.codecName = FLAGS_codec;
- if (args.codecName == "VP8") {
- args.codecType = kVideoCodecVP8;
- } else if (args.codecName == "VP9") {
- args.codecType = kVideoCodecVP9;
- } else if (args.codecName == "I420") {
- args.codecType = kVideoCodecI420;
+ args->codecName = FLAGS_codec;
+ if (args->codecName == "VP8") {
+ args->codecType = kVideoCodecVP8;
+ } else if (args->codecName == "VP9") {
+ args->codecType = kVideoCodecVP9;
+ } else if (args->codecName == "I420") {
+ args->codecType = kVideoCodecI420;
} else {
- printf("Invalid codec: %s\n", args.codecName.c_str());
+ printf("Invalid codec: %s\n", args->codecName.c_str());
return -1;
}
- args.inputFile = FLAGS_input_filename;
- args.outputFile = FLAGS_output_filename;
- args.rtt = FLAGS_rtt;
+ args->inputFile = FLAGS_input_filename;
+ args->outputFile = FLAGS_output_filename;
+ args->rtt = FLAGS_rtt;
return 0;
}
+} // namespace webrtc
-int main(int argc, char **argv) {
+int main(int argc, char** argv) {
// Initialize WebRTC fileutils.h so paths to resources can be resolved.
webrtc::test::SetExecutablePath(argv[0]);
google::ParseCommandLineFlags(&argc, &argv, true);
CmdArgs args;
- if (ParseArguments(args) != 0) {
+ if (webrtc::ParseArguments(&args) != 0) {
printf("Unable to parse input arguments\n");
return -1;
}
diff --git a/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
index 26a0d01..d4ee93f 100644
--- a/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
+++ b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
@@ -22,9 +22,8 @@
namespace webrtc {
namespace rtpplayer {
-class VcmPayloadSinkFactory::VcmPayloadSink
- : public PayloadSinkInterface,
- public VCMPacketRequestCallback {
+class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
+ public VCMPacketRequestCallback {
public:
VcmPayloadSink(VcmPayloadSinkFactory* factory,
RtpStreamInterface* stream,
@@ -43,9 +42,7 @@
vcm_->RegisterReceiveCallback(frame_receiver_.get());
}
- virtual ~VcmPayloadSink() {
- factory_->Remove(this);
- }
+ virtual ~VcmPayloadSink() { factory_->Remove(this); }
// PayloadSinkInterface
int32_t OnReceivedPayloadData(const uint8_t* payload_data,
@@ -136,8 +133,7 @@
}
const PayloadTypes& plt = stream->payload_types();
- for (PayloadTypesIterator it = plt.begin(); it != plt.end();
- ++it) {
+ for (PayloadTypesIterator it = plt.begin(); it != plt.end(); ++it) {
if (it->codec_type() != kVideoCodecULPFEC &&
it->codec_type() != kVideoCodecRED) {
VideoCodec codec;
diff --git a/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
index 1de1cd6..dae53b0 100644
--- a/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
+++ b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
@@ -29,9 +29,11 @@
class VcmPayloadSinkFactory : public PayloadSinkFactoryInterface {
public:
VcmPayloadSinkFactory(const std::string& base_out_filename,
- Clock* clock, bool protection_enabled,
+ Clock* clock,
+ bool protection_enabled,
VCMVideoProtection protection_method,
- int64_t rtt_ms, uint32_t render_delay_ms,
+ int64_t rtt_ms,
+ uint32_t render_delay_ms,
uint32_t min_playout_delay_ms);
virtual ~VcmPayloadSinkFactory();
diff --git a/webrtc/modules/video_coding/test/video_rtp_play.cc b/webrtc/modules/video_coding/test/video_rtp_play.cc
index 0a6b7d1..cb092e3 100644
--- a/webrtc/modules/video_coding/test/video_rtp_play.cc
+++ b/webrtc/modules/video_coding/test/video_rtp_play.cc
@@ -48,9 +48,9 @@
output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv";
webrtc::SimulatedClock clock(0);
- webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock,
- kConfigProtectionEnabled, kConfigProtectionMethod, kConfigRttMs,
- kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
+ webrtc::rtpplayer::VcmPayloadSinkFactory factory(
+ output_file, &clock, kConfigProtectionEnabled, kConfigProtectionMethod,
+ kConfigRttMs, kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
rtc::scoped_ptr<webrtc::rtpplayer::RtpPlayerInterface> rtp_player(
webrtc::rtpplayer::Create(args.inputFile, &factory, &clock, payload_types,
kConfigLossRate, kConfigRttMs,
@@ -63,7 +63,7 @@
while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
ret = factory.DecodeAndProcessAll(true);
if (ret < 0 || (kConfigMaxRuntimeMs > -1 &&
- clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
+ clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
break;
}
clock.AdvanceTimeMilliseconds(1);
diff --git a/webrtc/modules/video_coding/test/video_source.h b/webrtc/modules/video_coding/test/video_source.h
index 6b878e5..19d7f50 100644
--- a/webrtc/modules/video_coding/test/video_source.h
+++ b/webrtc/modules/video_coding/test/video_source.h
@@ -11,72 +11,75 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+#include <string>
+
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/typedefs.h"
-#include <string>
+enum VideoSize {
+ kUndefined,
+ kSQCIF, // 128*96 = 12 288
+ kQQVGA, // 160*120 = 19 200
+ kQCIF, // 176*144 = 25 344
+ kCGA, // 320*200 = 64 000
+ kQVGA, // 320*240 = 76 800
+ kSIF, // 352*240 = 84 480
+ kWQVGA, // 400*240 = 96 000
+ kCIF, // 352*288 = 101 376
+ kW288p, // 512*288 = 147 456 (WCIF)
+ k448p, // 576*448 = 281 088
+ kVGA, // 640*480 = 307 200
+ k432p, // 720*432 = 311 040
+ kW432p, // 768*432 = 331 776
+ k4SIF, // 704*480 = 337 920
+ kW448p, // 768*448 = 344 064
+ kNTSC, // 720*480 = 345 600
+ kFW448p, // 800*448 = 358 400
+ kWVGA, // 800*480 = 384 000
+ k4CIF, // 704*576 = 405 504
+ kSVGA, // 800*600 = 480 000
+ kW544p, // 960*544 = 522 240
+ kW576p, // 1024*576 = 589 824 (W4CIF)
+ kHD, // 960*720 = 691 200
+ kXGA, // 1024*768 = 786 432
+ kWHD, // 1280*720 = 921 600
+ kFullHD, // 1440*1080 = 1 555 200
+ kWFullHD, // 1920*1080 = 2 073 600
-enum VideoSize
- {
- kUndefined,
- kSQCIF, // 128*96 = 12 288
- kQQVGA, // 160*120 = 19 200
- kQCIF, // 176*144 = 25 344
- kCGA, // 320*200 = 64 000
- kQVGA, // 320*240 = 76 800
- kSIF, // 352*240 = 84 480
- kWQVGA, // 400*240 = 96 000
- kCIF, // 352*288 = 101 376
- kW288p, // 512*288 = 147 456 (WCIF)
- k448p, // 576*448 = 281 088
- kVGA, // 640*480 = 307 200
- k432p, // 720*432 = 311 040
- kW432p, // 768*432 = 331 776
- k4SIF, // 704*480 = 337 920
- kW448p, // 768*448 = 344 064
- kNTSC, // 720*480 = 345 600
- kFW448p, // 800*448 = 358 400
- kWVGA, // 800*480 = 384 000
- k4CIF, // 704*576 = 405 504
- kSVGA, // 800*600 = 480 000
- kW544p, // 960*544 = 522 240
- kW576p, // 1024*576 = 589 824 (W4CIF)
- kHD, // 960*720 = 691 200
- kXGA, // 1024*768 = 786 432
- kWHD, // 1280*720 = 921 600
- kFullHD, // 1440*1080 = 1 555 200
- kWFullHD, // 1920*1080 = 2 073 600
-
- kNumberOfVideoSizes
- };
-
-
-class VideoSource
-{
-public:
- VideoSource();
- VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
- VideoSource(std::string fileName, uint16_t width, uint16_t height,
- float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
-
- std::string GetFileName() const { return _fileName; }
- uint16_t GetWidth() const { return _width; }
- uint16_t GetHeight() const { return _height; }
- webrtc::VideoType GetType() const { return _type; }
- float GetFrameRate() const { return _frameRate; }
- int GetWidthHeight( VideoSize size);
-
- // Returns the filename with the path (including the leading slash) removed.
- std::string GetName() const;
-
- size_t GetFrameLength() const;
-
-private:
- std::string _fileName;
- uint16_t _width;
- uint16_t _height;
- webrtc::VideoType _type;
- float _frameRate;
+ kNumberOfVideoSizes
};
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+class VideoSource {
+ public:
+ VideoSource();
+ VideoSource(std::string fileName,
+ VideoSize size,
+ float frameRate,
+ webrtc::VideoType type = webrtc::kI420);
+ VideoSource(std::string fileName,
+ uint16_t width,
+ uint16_t height,
+ float frameRate = 30,
+ webrtc::VideoType type = webrtc::kI420);
+
+ std::string GetFileName() const { return _fileName; }
+ uint16_t GetWidth() const { return _width; }
+ uint16_t GetHeight() const { return _height; }
+ webrtc::VideoType GetType() const { return _type; }
+ float GetFrameRate() const { return _frameRate; }
+ int GetWidthHeight(VideoSize size);
+
+ // Returns the filename with the path (including the leading slash) removed.
+ std::string GetName() const;
+
+ size_t GetFrameLength() const;
+
+ private:
+ std::string _fileName;
+ uint16_t _width;
+ uint16_t _height;
+ webrtc::VideoType _type;
+ float _frameRate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
diff --git a/webrtc/modules/video_coding/timestamp_map.cc b/webrtc/modules/video_coding/timestamp_map.cc
index 14b16cd..97d2777 100644
--- a/webrtc/modules/video_coding/timestamp_map.cc
+++ b/webrtc/modules/video_coding/timestamp_map.cc
@@ -20,11 +20,9 @@
: ring_buffer_(new TimestampDataTuple[capacity]),
capacity_(capacity),
next_add_idx_(0),
- next_pop_idx_(0) {
-}
+ next_pop_idx_(0) {}
-VCMTimestampMap::~VCMTimestampMap() {
-}
+VCMTimestampMap::~VCMTimestampMap() {}
void VCMTimestampMap::Add(uint32_t timestamp, VCMFrameInformation* data) {
ring_buffer_[next_add_idx_].timestamp = timestamp;
@@ -62,4 +60,4 @@
bool VCMTimestampMap::IsEmpty() const {
return (next_add_idx_ == next_pop_idx_);
}
-}
+} // namespace webrtc
diff --git a/webrtc/modules/video_coding/timestamp_map.h b/webrtc/modules/video_coding/timestamp_map.h
index 3d6f1bc..435d058 100644
--- a/webrtc/modules/video_coding/timestamp_map.h
+++ b/webrtc/modules/video_coding/timestamp_map.h
@@ -44,4 +44,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/webrtc/modules/video_coding/timing.cc b/webrtc/modules/video_coding/timing.cc
index d2563a4..08dc307 100644
--- a/webrtc/modules/video_coding/timing.cc
+++ b/webrtc/modules/video_coding/timing.cc
@@ -10,17 +10,17 @@
#include "webrtc/modules/video_coding/timing.h"
+#include <algorithm>
+
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/modules/video_coding/jitter_buffer_common.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
-
namespace webrtc {
-VCMTiming::VCMTiming(Clock* clock,
- VCMTiming* master_timing)
+VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
clock_(clock),
master_(false),
@@ -120,8 +120,8 @@
// Not initialized, set current delay to target.
current_delay_ms_ = target_delay_ms;
} else if (target_delay_ms != current_delay_ms_) {
- int64_t delay_diff_ms = static_cast<int64_t>(target_delay_ms) -
- current_delay_ms_;
+ int64_t delay_diff_ms =
+ static_cast<int64_t>(target_delay_ms) - current_delay_ms_;
// Never change the delay with more than 100 ms every second. If we're
// changing the delay in too large steps we will get noticeable freezes. By
// limiting the change we can increase the delay in smaller steps, which
@@ -130,11 +130,13 @@
int64_t max_change_ms = 0;
if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
// wrap
- max_change_ms = kDelayMaxChangeMsPerS * (frame_timestamp +
- (static_cast<int64_t>(1) << 32) - prev_frame_timestamp_) / 90000;
+ max_change_ms = kDelayMaxChangeMsPerS *
+ (frame_timestamp + (static_cast<int64_t>(1) << 32) -
+ prev_frame_timestamp_) /
+ 90000;
} else {
max_change_ms = kDelayMaxChangeMsPerS *
- (frame_timestamp - prev_frame_timestamp_) / 90000;
+ (frame_timestamp - prev_frame_timestamp_) / 90000;
}
if (max_change_ms <= 0) {
// Any changes less than 1 ms are truncated and
@@ -155,7 +157,7 @@
CriticalSectionScoped cs(crit_sect_);
uint32_t target_delay_ms = TargetDelayInternal();
int64_t delayed_ms = actual_decode_time_ms -
- (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
+ (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
if (delayed_ms < 0) {
return;
}
@@ -193,8 +195,8 @@
ts_extrapolator_->Update(now_ms, time_stamp);
}
-int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms)
- const {
+int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp,
+ int64_t now_ms) const {
CriticalSectionScoped cs(crit_sect_);
const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms);
return render_time_ms;
@@ -203,7 +205,7 @@
int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
int64_t now_ms) const {
int64_t estimated_complete_time_ms =
- ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
+ ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
if (estimated_complete_time_ms == -1) {
estimated_complete_time_ms = now_ms;
}
@@ -214,19 +216,19 @@
}
// Must be called from inside a critical section.
-int32_t VCMTiming::MaxDecodeTimeMs(FrameType frame_type /*= kVideoFrameDelta*/)
- const {
+int32_t VCMTiming::MaxDecodeTimeMs(
+ FrameType frame_type /*= kVideoFrameDelta*/) const {
const int32_t decode_time_ms = codec_timer_.RequiredDecodeTimeMs(frame_type);
assert(decode_time_ms >= 0);
return decode_time_ms;
}
-uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
- const {
+uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
+ int64_t now_ms) const {
CriticalSectionScoped cs(crit_sect_);
- const int64_t max_wait_time_ms = render_time_ms - now_ms -
- MaxDecodeTimeMs() - render_delay_ms_;
+ const int64_t max_wait_time_ms =
+ render_time_ms - now_ms - MaxDecodeTimeMs() - render_delay_ms_;
if (max_wait_time_ms < 0) {
return 0;
@@ -234,8 +236,8 @@
return static_cast<uint32_t>(max_wait_time_ms);
}
-bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
- const {
+bool VCMTiming::EnoughTimeToDecode(
+ uint32_t available_processing_time_ms) const {
CriticalSectionScoped cs(crit_sect_);
int32_t max_decode_time_ms = MaxDecodeTimeMs();
if (max_decode_time_ms < 0) {
@@ -248,7 +250,8 @@
max_decode_time_ms = 1;
}
return static_cast<int32_t>(available_processing_time_ms) -
- max_decode_time_ms > 0;
+ max_decode_time_ms >
+ 0;
}
uint32_t VCMTiming::TargetVideoDelay() const {
@@ -258,7 +261,7 @@
uint32_t VCMTiming::TargetDelayInternal() const {
return std::max(min_playout_delay_ms_,
- jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
+ jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
}
void VCMTiming::GetTimings(int* decode_ms,
diff --git a/webrtc/modules/video_coding/timing.h b/webrtc/modules/video_coding/timing.h
index 46681a3..a4d0cf4 100644
--- a/webrtc/modules/video_coding/timing.h
+++ b/webrtc/modules/video_coding/timing.h
@@ -25,8 +25,7 @@
public:
// The primary timing component should be passed
// if this is the dual timing component.
- VCMTiming(Clock* clock,
- VCMTiming* master_timing = NULL);
+ explicit VCMTiming(Clock* clock, VCMTiming* master_timing = NULL);
~VCMTiming();
// Resets the timing to the initial state.
diff --git a/webrtc/modules/video_coding/timing_unittest.cc b/webrtc/modules/video_coding/timing_unittest.cc
index 02bd88d..2e8df83 100644
--- a/webrtc/modules/video_coding/timing_unittest.cc
+++ b/webrtc/modules/video_coding/timing_unittest.cc
@@ -55,8 +55,9 @@
clock.AdvanceTimeMilliseconds(1000);
timing.SetJitterDelay(jitterDelayMs);
timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(
- timeStamp, clock.TimeInMilliseconds()), clock.TimeInMilliseconds());
+ waitTime = timing.MaxWaitingTime(
+ timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
+ clock.TimeInMilliseconds());
// Since we gradually increase the delay we only get 100 ms every second.
EXPECT_EQ(jitterDelayMs - 10, waitTime);
@@ -85,11 +86,10 @@
for (int i = 0; i < 10; i++) {
int64_t startTimeMs = clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(10);
- timing.StopDecodeTimer(timeStamp,
- clock.TimeInMilliseconds() - startTimeMs,
- clock.TimeInMilliseconds(),
- timing.RenderTimeMs(
- timeStamp, clock.TimeInMilliseconds()));
+ timing.StopDecodeTimer(
+ timeStamp, clock.TimeInMilliseconds() - startTimeMs,
+ clock.TimeInMilliseconds(),
+ timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()));
timeStamp += 90000 / 25;
clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
@@ -107,7 +107,7 @@
uint32_t minTotalDelayMs = 200;
timing.set_min_playout_delay(minTotalDelayMs);
clock.AdvanceTimeMilliseconds(5000);
- timeStamp += 5*90000;
+ timeStamp += 5 * 90000;
timing.UpdateCurrentDelay(timeStamp);
const int kRenderDelayMs = 10;
timing.set_render_delay(kRenderDelayMs);
@@ -123,7 +123,7 @@
// Reset playout delay.
timing.set_min_playout_delay(0);
clock.AdvanceTimeMilliseconds(5000);
- timeStamp += 5*90000;
+ timeStamp += 5 * 90000;
timing.UpdateCurrentDelay(timeStamp);
}
@@ -137,8 +137,8 @@
timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
clock.AdvanceTimeMilliseconds(1000 / kFramerate);
timestamp += 90000 / kFramerate;
- int64_t render_time = timing.RenderTimeMs(0xFFFFFFFFu,
- clock.TimeInMilliseconds());
+ int64_t render_time =
+ timing.RenderTimeMs(0xFFFFFFFFu, clock.TimeInMilliseconds());
EXPECT_EQ(3 * 1000 / kFramerate, render_time);
render_time = timing.RenderTimeMs(89u, // One second later in 90 kHz.
clock.TimeInMilliseconds());
diff --git a/webrtc/modules/video_coding/utility/frame_dropper.cc b/webrtc/modules/video_coding/utility/frame_dropper.cc
index dfa7df8..a0aa67b 100644
--- a/webrtc/modules/video_coding/utility/frame_dropper.cc
+++ b/webrtc/modules/video_coding/utility/frame_dropper.cc
@@ -12,8 +12,7 @@
#include "webrtc/system_wrappers/include/trace.h"
-namespace webrtc
-{
+namespace webrtc {
const float kDefaultKeyFrameSizeAvgKBits = 0.9f;
const float kDefaultKeyFrameRatio = 0.99f;
@@ -22,339 +21,266 @@
const float kDefaultMaxTimeToDropFrames = 4.0f; // In seconds.
FrameDropper::FrameDropper()
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(kDefaultMaxTimeToDropFrames)
-{
- Reset();
+ : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+ _keyFrameRatio(kDefaultKeyFrameRatio),
+ _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+ _enabled(true),
+ _max_time_drops(kDefaultMaxTimeToDropFrames) {
+ Reset();
}
FrameDropper::FrameDropper(float max_time_drops)
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(max_time_drops)
-{
- Reset();
+ : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+ _keyFrameRatio(kDefaultKeyFrameRatio),
+ _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+ _enabled(true),
+ _max_time_drops(max_time_drops) {
+ Reset();
}
-void
-FrameDropper::Reset()
-{
- _keyFrameRatio.Reset(0.99f);
- _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
- _keyFrameSizeAvgKbits.Reset(0.9f);
- _keyFrameCount = 0;
+void FrameDropper::Reset() {
+ _keyFrameRatio.Reset(0.99f);
+ _keyFrameRatio.Apply(
+ 1.0f, 1.0f / 300.0f); // 1 key frame every 10th second in 30 fps
+ _keyFrameSizeAvgKbits.Reset(0.9f);
+ _keyFrameCount = 0;
+ _accumulator = 0.0f;
+ _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
+ _targetBitRate = 300.0f;
+ _incoming_frame_rate = 30;
+ _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
+ _dropNext = false;
+ _dropRatio.Reset(0.9f);
+ _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
+ _dropCount = 0;
+ _windowSize = 0.5f;
+ _wasBelowMax = true;
+ _fastMode = false; // start with normal (non-aggressive) mode
+ // Cap for the encoder buffer level/accumulator, in secs.
+ _cap_buffer_size = 3.0f;
+ // Cap on maximum amount of dropped frames between kept frames, in secs.
+ _max_time_drops = 4.0f;
+}
+
+void FrameDropper::Enable(bool enable) {
+ _enabled = enable;
+}
+
+void FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame) {
+ if (!_enabled) {
+ return;
+ }
+ float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
+ if (!deltaFrame &&
+ !_fastMode) { // fast mode does not treat key-frames any different
+ _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
+ _keyFrameRatio.Apply(1.0, 1.0);
+ if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered()) {
+ // Remove the average key frame size since we
+ // compensate for key frames when adding delta
+ // frames.
+ frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
+ } else {
+ // Shouldn't be negative, so zero is the lower bound.
+ frameSizeKbits = 0;
+ }
+ if (_keyFrameRatio.filtered() > 1e-5 &&
+ 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+ // We are sending key frames more often than our upper bound for
+ // how much we allow the key frame compensation to be spread
+ // out in time. Therefor we must use the key frame ratio rather
+ // than keyFrameSpreadFrames.
+ _keyFrameCount =
+ static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
+ } else {
+ // Compensate for the key frame the following frames
+ _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
+ }
+ } else {
+ // Decrease the keyFrameRatio
+ _keyFrameRatio.Apply(1.0, 0.0);
+ }
+ // Change the level of the accumulator (bucket)
+ _accumulator += frameSizeKbits;
+ CapAccumulator();
+}
+
+void FrameDropper::Leak(uint32_t inputFrameRate) {
+ if (!_enabled) {
+ return;
+ }
+ if (inputFrameRate < 1) {
+ return;
+ }
+ if (_targetBitRate < 0.0f) {
+ return;
+ }
+ _keyFrameSpreadFrames = 0.5f * inputFrameRate;
+ // T is the expected bits per frame (target). If all frames were the same
+ // size,
+ // we would get T bits per frame. Notice that T is also weighted to be able to
+ // force a lower frame rate if wanted.
+ float T = _targetBitRate / inputFrameRate;
+ if (_keyFrameCount > 0) {
+ // Perform the key frame compensation
+ if (_keyFrameRatio.filtered() > 0 &&
+ 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+ T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
+ } else {
+ T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
+ }
+ _keyFrameCount--;
+ }
+ _accumulator -= T;
+ if (_accumulator < 0.0f) {
_accumulator = 0.0f;
- _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
- _targetBitRate = 300.0f;
- _incoming_frame_rate = 30;
- _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
- _dropNext = false;
- _dropRatio.Reset(0.9f);
- _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
- _dropCount = 0;
- _windowSize = 0.5f;
- _wasBelowMax = true;
- _fastMode = false; // start with normal (non-aggressive) mode
- // Cap for the encoder buffer level/accumulator, in secs.
- _cap_buffer_size = 3.0f;
- // Cap on maximum amount of dropped frames between kept frames, in secs.
- _max_time_drops = 4.0f;
+ }
+ UpdateRatio();
}
-void
-FrameDropper::Enable(bool enable)
-{
- _enabled = enable;
+void FrameDropper::UpdateNack(uint32_t nackBytes) {
+ if (!_enabled) {
+ return;
+ }
+ _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
}
-void
-FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame)
-{
- if (!_enabled)
- {
- return;
- }
- float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
- if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
- {
- _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
- _keyFrameRatio.Apply(1.0, 1.0);
- if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered())
- {
- // Remove the average key frame size since we
- // compensate for key frames when adding delta
- // frames.
- frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
- }
- else
- {
- // Shouldn't be negative, so zero is the lower bound.
- frameSizeKbits = 0;
- }
- if (_keyFrameRatio.filtered() > 1e-5 &&
- 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
- {
- // We are sending key frames more often than our upper bound for
- // how much we allow the key frame compensation to be spread
- // out in time. Therefor we must use the key frame ratio rather
- // than keyFrameSpreadFrames.
- _keyFrameCount =
- static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
- }
- else
- {
- // Compensate for the key frame the following frames
- _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
- }
- }
- else
- {
- // Decrease the keyFrameRatio
- _keyFrameRatio.Apply(1.0, 0.0);
- }
- // Change the level of the accumulator (bucket)
- _accumulator += frameSizeKbits;
- CapAccumulator();
+void FrameDropper::FillBucket(float inKbits, float outKbits) {
+ _accumulator += (inKbits - outKbits);
}
-void
-FrameDropper::Leak(uint32_t inputFrameRate)
-{
- if (!_enabled)
- {
- return;
+void FrameDropper::UpdateRatio() {
+ if (_accumulator > 1.3f * _accumulatorMax) {
+ // Too far above accumulator max, react faster
+ _dropRatio.UpdateBase(0.8f);
+ } else {
+ // Go back to normal reaction
+ _dropRatio.UpdateBase(0.9f);
+ }
+ if (_accumulator > _accumulatorMax) {
+ // We are above accumulator max, and should ideally
+ // drop a frame. Increase the dropRatio and drop
+ // the frame later.
+ if (_wasBelowMax) {
+ _dropNext = true;
}
- if (inputFrameRate < 1)
- {
- return;
+ if (_fastMode) {
+ // always drop in aggressive mode
+ _dropNext = true;
}
- if (_targetBitRate < 0.0f)
- {
- return;
- }
- _keyFrameSpreadFrames = 0.5f * inputFrameRate;
- // T is the expected bits per frame (target). If all frames were the same size,
- // we would get T bits per frame. Notice that T is also weighted to be able to
- // force a lower frame rate if wanted.
- float T = _targetBitRate / inputFrameRate;
- if (_keyFrameCount > 0)
- {
- // Perform the key frame compensation
- if (_keyFrameRatio.filtered() > 0 &&
- 1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
- {
- T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
- }
- else
- {
- T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
- }
- _keyFrameCount--;
- }
- _accumulator -= T;
- if (_accumulator < 0.0f)
- {
- _accumulator = 0.0f;
- }
- UpdateRatio();
+
+ _dropRatio.Apply(1.0f, 1.0f);
+ _dropRatio.UpdateBase(0.9f);
+ } else {
+ _dropRatio.Apply(1.0f, 0.0f);
+ }
+ _wasBelowMax = _accumulator < _accumulatorMax;
}
-void
-FrameDropper::UpdateNack(uint32_t nackBytes)
-{
- if (!_enabled)
- {
- return;
- }
- _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
-}
-
-void
-FrameDropper::FillBucket(float inKbits, float outKbits)
-{
- _accumulator += (inKbits - outKbits);
-}
-
-void
-FrameDropper::UpdateRatio()
-{
- if (_accumulator > 1.3f * _accumulatorMax)
- {
- // Too far above accumulator max, react faster
- _dropRatio.UpdateBase(0.8f);
- }
- else
- {
- // Go back to normal reaction
- _dropRatio.UpdateBase(0.9f);
- }
- if (_accumulator > _accumulatorMax)
- {
- // We are above accumulator max, and should ideally
- // drop a frame. Increase the dropRatio and drop
- // the frame later.
- if (_wasBelowMax)
- {
- _dropNext = true;
- }
- if (_fastMode)
- {
- // always drop in aggressive mode
- _dropNext = true;
- }
-
- _dropRatio.Apply(1.0f, 1.0f);
- _dropRatio.UpdateBase(0.9f);
- }
- else
- {
- _dropRatio.Apply(1.0f, 0.0f);
- }
- _wasBelowMax = _accumulator < _accumulatorMax;
-}
-
-// This function signals when to drop frames to the caller. It makes use of the dropRatio
+// This function signals when to drop frames to the caller. It makes use of the
+// dropRatio
// to smooth out the drops over time.
-bool
-FrameDropper::DropFrame()
-{
- if (!_enabled)
- {
- return false;
- }
- if (_dropNext)
- {
- _dropNext = false;
- _dropCount = 0;
- }
-
- if (_dropRatio.filtered() >= 0.5f) // Drops per keep
- {
- // limit is the number of frames we should drop between each kept frame
- // to keep our drop ratio. limit is positive in this case.
- float denom = 1.0f - _dropRatio.filtered();
- if (denom < 1e-5)
- {
- denom = (float)1e-5;
- }
- int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
- // Put a bound on the max amount of dropped frames between each kept
- // frame, in terms of frame rate and window size (secs).
- int max_limit = static_cast<int>(_incoming_frame_rate *
- _max_time_drops);
- if (limit > max_limit) {
- limit = max_limit;
- }
- if (_dropCount < 0)
- {
- // Reset the _dropCount since it was negative and should be positive.
- if (_dropRatio.filtered() > 0.4f)
- {
- _dropCount = -_dropCount;
- }
- else
- {
- _dropCount = 0;
- }
- }
- if (_dropCount < limit)
- {
- // As long we are below the limit we should drop frames.
- _dropCount++;
- return true;
- }
- else
- {
- // Only when we reset _dropCount a frame should be kept.
- _dropCount = 0;
- return false;
- }
- }
- else if (_dropRatio.filtered() > 0.0f &&
- _dropRatio.filtered() < 0.5f) // Keeps per drop
- {
- // limit is the number of frames we should keep between each drop
- // in order to keep the drop ratio. limit is negative in this case,
- // and the _dropCount is also negative.
- float denom = _dropRatio.filtered();
- if (denom < 1e-5)
- {
- denom = (float)1e-5;
- }
- int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
- if (_dropCount > 0)
- {
- // Reset the _dropCount since we have a positive
- // _dropCount, and it should be negative.
- if (_dropRatio.filtered() < 0.6f)
- {
- _dropCount = -_dropCount;
- }
- else
- {
- _dropCount = 0;
- }
- }
- if (_dropCount > limit)
- {
- if (_dropCount == 0)
- {
- // Drop frames when we reset _dropCount.
- _dropCount--;
- return true;
- }
- else
- {
- // Keep frames as long as we haven't reached limit.
- _dropCount--;
- return false;
- }
- }
- else
- {
- _dropCount = 0;
- return false;
- }
- }
- _dropCount = 0;
+bool FrameDropper::DropFrame() {
+ if (!_enabled) {
return false;
+ }
+ if (_dropNext) {
+ _dropNext = false;
+ _dropCount = 0;
+ }
- // A simpler version, unfiltered and quicker
- //bool dropNext = _dropNext;
- //_dropNext = false;
- //return dropNext;
+ if (_dropRatio.filtered() >= 0.5f) { // Drops per keep
+ // limit is the number of frames we should drop between each kept frame
+ // to keep our drop ratio. limit is positive in this case.
+ float denom = 1.0f - _dropRatio.filtered();
+ if (denom < 1e-5) {
+ denom = 1e-5f;
+ }
+ int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+ // Put a bound on the max amount of dropped frames between each kept
+ // frame, in terms of frame rate and window size (secs).
+ int max_limit = static_cast<int>(_incoming_frame_rate * _max_time_drops);
+ if (limit > max_limit) {
+ limit = max_limit;
+ }
+ if (_dropCount < 0) {
+ // Reset the _dropCount since it was negative and should be positive.
+ if (_dropRatio.filtered() > 0.4f) {
+ _dropCount = -_dropCount;
+ } else {
+ _dropCount = 0;
+ }
+ }
+ if (_dropCount < limit) {
+ // As long we are below the limit we should drop frames.
+ _dropCount++;
+ return true;
+ } else {
+ // Only when we reset _dropCount a frame should be kept.
+ _dropCount = 0;
+ return false;
+ }
+ } else if (_dropRatio.filtered() > 0.0f &&
+ _dropRatio.filtered() < 0.5f) { // Keeps per drop
+ // limit is the number of frames we should keep between each drop
+ // in order to keep the drop ratio. limit is negative in this case,
+ // and the _dropCount is also negative.
+ float denom = _dropRatio.filtered();
+ if (denom < 1e-5) {
+ denom = 1e-5f;
+ }
+ int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+ if (_dropCount > 0) {
+ // Reset the _dropCount since we have a positive
+ // _dropCount, and it should be negative.
+ if (_dropRatio.filtered() < 0.6f) {
+ _dropCount = -_dropCount;
+ } else {
+ _dropCount = 0;
+ }
+ }
+ if (_dropCount > limit) {
+ if (_dropCount == 0) {
+ // Drop frames when we reset _dropCount.
+ _dropCount--;
+ return true;
+ } else {
+ // Keep frames as long as we haven't reached limit.
+ _dropCount--;
+ return false;
+ }
+ } else {
+ _dropCount = 0;
+ return false;
+ }
+ }
+ _dropCount = 0;
+ return false;
+
+ // A simpler version, unfiltered and quicker
+ // bool dropNext = _dropNext;
+ // _dropNext = false;
+ // return dropNext;
}
-void
-FrameDropper::SetRates(float bitRate, float incoming_frame_rate)
-{
- // Bit rate of -1 means infinite bandwidth.
- _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
- if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
- {
- // Rescale the accumulator level if the accumulator max decreases
- _accumulator = bitRate / _targetBitRate * _accumulator;
- }
- _targetBitRate = bitRate;
- CapAccumulator();
- _incoming_frame_rate = incoming_frame_rate;
+void FrameDropper::SetRates(float bitRate, float incoming_frame_rate) {
+ // Bit rate of -1 means infinite bandwidth.
+ _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
+ if (_targetBitRate > 0.0f && bitRate < _targetBitRate &&
+ _accumulator > _accumulatorMax) {
+ // Rescale the accumulator level if the accumulator max decreases
+ _accumulator = bitRate / _targetBitRate * _accumulator;
+ }
+ _targetBitRate = bitRate;
+ CapAccumulator();
+ _incoming_frame_rate = incoming_frame_rate;
}
-float
-FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const
-{
- if (!_enabled)
- {
- return static_cast<float>(inputFrameRate);
- }
- return inputFrameRate * (1.0f - _dropRatio.filtered());
+float FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const {
+ if (!_enabled) {
+ return static_cast<float>(inputFrameRate);
+ }
+ return inputFrameRate * (1.0f - _dropRatio.filtered());
}
// Put a cap on the accumulator, i.e., don't let it grow beyond some level.
@@ -366,5 +292,4 @@
_accumulator = max_accumulator;
}
}
-
-}
+} // namespace webrtc
diff --git a/webrtc/modules/video_coding/utility/frame_dropper.h b/webrtc/modules/video_coding/utility/frame_dropper.h
index 2293cef..7ec85ea 100644
--- a/webrtc/modules/video_coding/utility/frame_dropper.h
+++ b/webrtc/modules/video_coding/utility/frame_dropper.h
@@ -23,72 +23,72 @@
// over use when the encoder can't keep its bit rate.
class FrameDropper {
public:
- FrameDropper();
- explicit FrameDropper(float max_time_drops);
- virtual ~FrameDropper() {}
+ FrameDropper();
+ explicit FrameDropper(float max_time_drops);
+ virtual ~FrameDropper() {}
- // Resets the FrameDropper to its initial state.
- // This means that the frameRateWeight is set to its
- // default value as well.
- virtual void Reset();
+ // Resets the FrameDropper to its initial state.
+ // This means that the frameRateWeight is set to its
+ // default value as well.
+ virtual void Reset();
- virtual void Enable(bool enable);
- // Answers the question if it's time to drop a frame
- // if we want to reach a given frame rate. Must be
- // called for every frame.
- //
- // Return value : True if we should drop the current frame
- virtual bool DropFrame();
- // Updates the FrameDropper with the size of the latest encoded
- // frame. The FrameDropper calculates a new drop ratio (can be
- // seen as the probability to drop a frame) and updates its
- // internal statistics.
- //
- // Input:
- // - frameSizeBytes : The size of the latest frame
- // returned from the encoder.
- // - deltaFrame : True if the encoder returned
- // a key frame.
- virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
+ virtual void Enable(bool enable);
+ // Answers the question if it's time to drop a frame
+ // if we want to reach a given frame rate. Must be
+ // called for every frame.
+ //
+ // Return value : True if we should drop the current frame
+ virtual bool DropFrame();
+ // Updates the FrameDropper with the size of the latest encoded
+ // frame. The FrameDropper calculates a new drop ratio (can be
+ // seen as the probability to drop a frame) and updates its
+ // internal statistics.
+ //
+ // Input:
+ // - frameSizeBytes : The size of the latest frame
+ // returned from the encoder.
+ // - deltaFrame : True if the encoder returned
+ // a key frame.
+ virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
- virtual void Leak(uint32_t inputFrameRate);
+ virtual void Leak(uint32_t inputFrameRate);
- void UpdateNack(uint32_t nackBytes);
+ void UpdateNack(uint32_t nackBytes);
- // Sets the target bit rate and the frame rate produced by
- // the camera.
- //
- // Input:
- // - bitRate : The target bit rate
- virtual void SetRates(float bitRate, float incoming_frame_rate);
+ // Sets the target bit rate and the frame rate produced by
+ // the camera.
+ //
+ // Input:
+ // - bitRate : The target bit rate
+ virtual void SetRates(float bitRate, float incoming_frame_rate);
- // Return value : The current average frame rate produced
- // if the DropFrame() function is used as
- // instruction of when to drop frames.
- virtual float ActualFrameRate(uint32_t inputFrameRate) const;
+ // Return value : The current average frame rate produced
+ // if the DropFrame() function is used as
+ // instruction of when to drop frames.
+ virtual float ActualFrameRate(uint32_t inputFrameRate) const;
private:
- void FillBucket(float inKbits, float outKbits);
- void UpdateRatio();
- void CapAccumulator();
+ void FillBucket(float inKbits, float outKbits);
+ void UpdateRatio();
+ void CapAccumulator();
- rtc::ExpFilter _keyFrameSizeAvgKbits;
- rtc::ExpFilter _keyFrameRatio;
- float _keyFrameSpreadFrames;
- int32_t _keyFrameCount;
- float _accumulator;
- float _accumulatorMax;
- float _targetBitRate;
- bool _dropNext;
- rtc::ExpFilter _dropRatio;
- int32_t _dropCount;
- float _windowSize;
- float _incoming_frame_rate;
- bool _wasBelowMax;
- bool _enabled;
- bool _fastMode;
- float _cap_buffer_size;
- float _max_time_drops;
+ rtc::ExpFilter _keyFrameSizeAvgKbits;
+ rtc::ExpFilter _keyFrameRatio;
+ float _keyFrameSpreadFrames;
+ int32_t _keyFrameCount;
+ float _accumulator;
+ float _accumulatorMax;
+ float _targetBitRate;
+ bool _dropNext;
+ rtc::ExpFilter _dropRatio;
+ int32_t _dropCount;
+ float _windowSize;
+ float _incoming_frame_rate;
+ bool _wasBelowMax;
+ bool _enabled;
+ bool _fastMode;
+ float _cap_buffer_size;
+ float _max_time_drops;
}; // end of VCMFrameDropper class
} // namespace webrtc
diff --git a/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h b/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
index dee00ba..b68a4b8 100644
--- a/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
+++ b/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
@@ -20,20 +20,13 @@
class MockFrameDropper : public FrameDropper {
public:
- MOCK_METHOD0(Reset,
- void());
- MOCK_METHOD1(Enable,
- void(bool enable));
- MOCK_METHOD0(DropFrame,
- bool());
- MOCK_METHOD2(Fill,
- void(size_t frameSizeBytes, bool deltaFrame));
- MOCK_METHOD1(Leak,
- void(uint32_t inputFrameRate));
- MOCK_METHOD2(SetRates,
- void(float bitRate, float incoming_frame_rate));
- MOCK_CONST_METHOD1(ActualFrameRate,
- float(uint32_t inputFrameRate));
+ MOCK_METHOD0(Reset, void());
+ MOCK_METHOD1(Enable, void(bool enable));
+ MOCK_METHOD0(DropFrame, bool());
+ MOCK_METHOD2(Fill, void(size_t frameSizeBytes, bool deltaFrame));
+ MOCK_METHOD1(Leak, void(uint32_t inputFrameRate));
+ MOCK_METHOD2(SetRates, void(float bitRate, float incoming_frame_rate));
+ MOCK_CONST_METHOD1(ActualFrameRate, float(uint32_t inputFrameRate));
};
} // namespace webrtc
diff --git a/webrtc/modules/video_coding/utility/moving_average.h b/webrtc/modules/video_coding/utility/moving_average.h
index 8de1dd2..494bfd5 100644
--- a/webrtc/modules/video_coding/utility/moving_average.h
+++ b/webrtc/modules/video_coding/utility/moving_average.h
@@ -16,7 +16,7 @@
#include "webrtc/typedefs.h"
namespace webrtc {
-template<class T>
+template <class T>
class MovingAverage {
public:
MovingAverage();
@@ -30,17 +30,17 @@
std::list<T> samples_;
};
-template<class T>
-MovingAverage<T>::MovingAverage() : sum_(static_cast<T>(0)) {
-}
+template <class T>
+MovingAverage<T>::MovingAverage()
+ : sum_(static_cast<T>(0)) {}
-template<class T>
+template <class T>
void MovingAverage<T>::AddSample(T sample) {
samples_.push_back(sample);
sum_ += sample;
}
-template<class T>
+template <class T>
bool MovingAverage<T>::GetAverage(size_t num_samples, T* avg) {
if (num_samples > samples_.size())
return false;
@@ -55,13 +55,13 @@
return true;
}
-template<class T>
+template <class T>
void MovingAverage<T>::Reset() {
sum_ = static_cast<T>(0);
samples_.clear();
}
-template<class T>
+template <class T>
int MovingAverage<T>::size() {
return samples_.size();
}
diff --git a/webrtc/modules/video_coding/utility/quality_scaler.cc b/webrtc/modules/video_coding/utility/quality_scaler.cc
index 9aae17c..76bf9f5 100644
--- a/webrtc/modules/video_coding/utility/quality_scaler.cc
+++ b/webrtc/modules/video_coding/utility/quality_scaler.cc
@@ -26,8 +26,7 @@
downscale_shift_(0),
framerate_down_(false),
min_width_(kDefaultMinDownscaleDimension),
- min_height_(kDefaultMinDownscaleDimension) {
-}
+ min_height_(kDefaultMinDownscaleDimension) {}
void QualityScaler::Init(int low_qp_threshold,
int high_qp_threshold,
@@ -91,7 +90,7 @@
AdjustScale(false);
}
} else if (average_qp_.GetAverage(num_samples_, &avg_qp) &&
- avg_qp <= low_qp_threshold_) {
+ avg_qp <= low_qp_threshold_) {
if (use_framerate_reduction_ && framerate_down_) {
target_framerate_ = -1;
framerate_down_ = false;
@@ -104,7 +103,7 @@
assert(downscale_shift_ >= 0);
for (int shift = downscale_shift_;
shift > 0 && (res_.width / 2 >= min_width_) &&
- (res_.height / 2 >= min_height_);
+ (res_.height / 2 >= min_height_);
--shift) {
res_.width /= 2;
res_.height /= 2;
@@ -124,13 +123,8 @@
if (res.width == frame.width())
return frame;
- scaler_.Set(frame.width(),
- frame.height(),
- res.width,
- res.height,
- kI420,
- kI420,
- kScaleBox);
+ scaler_.Set(frame.width(), frame.height(), res.width, res.height, kI420,
+ kI420, kScaleBox);
if (scaler_.Scale(frame, &scaled_frame_) != 0)
return frame;
diff --git a/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
index 58f7bee..bad73a7 100644
--- a/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
+++ b/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
@@ -33,6 +33,7 @@
int width;
int height;
};
+
protected:
enum ScaleDirection {
kKeepScaleAtHighQp,
@@ -43,8 +44,8 @@
enum BadQualityMetric { kDropFrame, kReportLowQP };
QualityScalerTest() {
- input_frame_.CreateEmptyFrame(
- kWidth, kHeight, kWidth, kHalfWidth, kHalfWidth);
+ input_frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, kHalfWidth,
+ kHalfWidth);
qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false);
qs_.ReportFramerate(kFramerate);
qs_.OnEncodeFrame(input_frame_);
@@ -97,7 +98,8 @@
int num_second,
int initial_framerate);
- void VerifyQualityAdaptation(int initial_framerate, int seconds,
+ void VerifyQualityAdaptation(int initial_framerate,
+ int seconds,
bool expect_spatial_resize,
bool expect_framerate_reduction);
@@ -183,8 +185,8 @@
void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
const int initial_min_dimension = input_frame_.width() < input_frame_.height()
- ? input_frame_.width()
- : input_frame_.height();
+ ? input_frame_.width()
+ : input_frame_.height();
int min_dimension = initial_min_dimension;
int current_shift = 0;
// Drop all frames to force-trigger downscaling.
@@ -229,14 +231,14 @@
const int kOddWidth = 517;
const int kHalfOddWidth = (kOddWidth + 1) / 2;
const int kOddHeight = 1239;
- input_frame_.CreateEmptyFrame(
- kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth, kHalfOddWidth);
+ input_frame_.CreateEmptyFrame(kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth,
+ kHalfOddWidth);
ContinuouslyDownscalesByHalfDimensionsAndBackUp();
}
void QualityScalerTest::DoesNotDownscaleFrameDimensions(int width, int height) {
- input_frame_.CreateEmptyFrame(
- width, height, width, (width + 1) / 2, (width + 1) / 2);
+ input_frame_.CreateEmptyFrame(width, height, width, (width + 1) / 2,
+ (width + 1) / 2);
for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
qs_.ReportDroppedFrame();
@@ -259,7 +261,9 @@
}
QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
- BadQualityMetric dropframe_lowqp, int num_second, int initial_framerate) {
+ BadQualityMetric dropframe_lowqp,
+ int num_second,
+ int initial_framerate) {
QualityScalerTest::Resolution res;
res.framerate = initial_framerate;
qs_.OnEncodeFrame(input_frame_);
@@ -288,7 +292,9 @@
}
void QualityScalerTest::VerifyQualityAdaptation(
- int initial_framerate, int seconds, bool expect_spatial_resize,
+ int initial_framerate,
+ int seconds,
+ bool expect_spatial_resize,
bool expect_framerate_reduction) {
const int kDisabledBadQpThreshold = kMaxQp + 1;
qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator,
@@ -298,8 +304,8 @@
int init_height = qs_.GetScaledResolution().height;
// Test reducing framerate by dropping frame continuously.
- QualityScalerTest::Resolution res = TriggerResolutionChange(
- kDropFrame, seconds, initial_framerate);
+ QualityScalerTest::Resolution res =
+ TriggerResolutionChange(kDropFrame, seconds, initial_framerate);
if (expect_framerate_reduction) {
EXPECT_LT(res.framerate, initial_framerate);
diff --git a/webrtc/modules/video_coding/utility/vp8_header_parser.cc b/webrtc/modules/video_coding/utility/vp8_header_parser.cc
index 13d1616..631385d 100644
--- a/webrtc/modules/video_coding/utility/vp8_header_parser.cc
+++ b/webrtc/modules/video_coding/utility/vp8_header_parser.cc
@@ -43,12 +43,12 @@
const uint32_t in_bits = *(const uint32_t*)(br->buf_);
br->buf_ += BITS >> 3;
#if defined(WEBRTC_ARCH_BIG_ENDIAN)
- bits = static_cast<uint32_t>(in_bits);
- if (BITS != 8 * sizeof(uint32_t))
- bits >>= (8 * sizeof(uint32_t) - BITS);
+ bits = static_cast<uint32_t>(in_bits);
+ if (BITS != 8 * sizeof(uint32_t))
+ bits >>= (8 * sizeof(uint32_t) - BITS);
#else
- bits = BSwap32(in_bits);
- bits >>= 32 - BITS;
+ bits = BSwap32(in_bits);
+ bits >>= 32 - BITS;
#endif
br->value_ = bits | (br->value_ << BITS);
br->bits_ += BITS;
@@ -60,12 +60,12 @@
static void VP8InitBitReader(VP8BitReader* const br,
const uint8_t* const start,
const uint8_t* const end) {
- br->range_ = 255 - 1;
- br->buf_ = start;
+ br->range_ = 255 - 1;
+ br->buf_ = start;
br->buf_end_ = end;
- br->value_ = 0;
- br->bits_ = -8; // To load the very first 8bits.
- br->eof_ = 0;
+ br->value_ = 0;
+ br->bits_ = -8; // To load the very first 8bits.
+ br->eof_ = 0;
VP8LoadNewBytes(br);
}
@@ -122,7 +122,7 @@
int s;
VP8Get(br);
for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
- VP8Get(br) ? VP8GetSignedValue(br, 7) : 0;
+ VP8Get(br) ? VP8GetSignedValue(br, 7) : 0;
}
for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
VP8Get(br) ? VP8GetSignedValue(br, 6) : 0;
diff --git a/webrtc/modules/video_coding/utility/vp8_header_parser.h b/webrtc/modules/video_coding/utility/vp8_header_parser.h
index b4b1d33..b0c684c 100644
--- a/webrtc/modules/video_coding/utility/vp8_header_parser.h
+++ b/webrtc/modules/video_coding/utility/vp8_header_parser.h
@@ -28,46 +28,34 @@
typedef struct VP8BitReader VP8BitReader;
struct VP8BitReader {
// Boolean decoder.
- uint32_t value_; // Current value.
- uint32_t range_; // Current range minus 1. In [127, 254] interval.
- int bits_; // Number of valid bits left.
+ uint32_t value_; // Current value.
+ uint32_t range_; // Current range minus 1. In [127, 254] interval.
+ int bits_; // Number of valid bits left.
// Read buffer.
- const uint8_t* buf_; // Next byte to be read.
- const uint8_t* buf_end_; // End of read buffer.
- int eof_; // True if input is exhausted.
+ const uint8_t* buf_; // Next byte to be read.
+ const uint8_t* buf_end_; // End of read buffer.
+ int eof_; // True if input is exhausted.
};
const uint8_t kVP8Log2Range[128] = {
- 7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4,
- 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
- 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
- 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
- 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
- 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
- 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
- 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
- 0
-};
+ 7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, 3, 3,
+ 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+ 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0};
// range = ((range - 1) << kVP8Log2Range[range]) + 1
const uint8_t kVP8NewRange[128] = {
- 127, 127, 191, 127, 159, 191, 223, 127,
- 143, 159, 175, 191, 207, 223, 239, 127,
- 135, 143, 151, 159, 167, 175, 183, 191,
- 199, 207, 215, 223, 231, 239, 247, 127,
- 131, 135, 139, 143, 147, 151, 155, 159,
- 163, 167, 171, 175, 179, 183, 187, 191,
- 195, 199, 203, 207, 211, 215, 219, 223,
- 227, 231, 235, 239, 243, 247, 251, 127,
- 129, 131, 133, 135, 137, 139, 141, 143,
- 145, 147, 149, 151, 153, 155, 157, 159,
- 161, 163, 165, 167, 169, 171, 173, 175,
- 177, 179, 181, 183, 185, 187, 189, 191,
- 193, 195, 197, 199, 201, 203, 205, 207,
- 209, 211, 213, 215, 217, 219, 221, 223,
- 225, 227, 229, 231, 233, 235, 237, 239,
- 241, 243, 245, 247, 249, 251, 253, 127
-};
+ 127, 127, 191, 127, 159, 191, 223, 127, 143, 159, 175, 191, 207, 223, 239,
+ 127, 135, 143, 151, 159, 167, 175, 183, 191, 199, 207, 215, 223, 231, 239,
+ 247, 127, 131, 135, 139, 143, 147, 151, 155, 159, 163, 167, 171, 175, 179,
+ 183, 187, 191, 195, 199, 203, 207, 211, 215, 219, 223, 227, 231, 235, 239,
+ 243, 247, 251, 127, 129, 131, 133, 135, 137, 139, 141, 143, 145, 147, 149,
+ 151, 153, 155, 157, 159, 161, 163, 165, 167, 169, 171, 173, 175, 177, 179,
+ 181, 183, 185, 187, 189, 191, 193, 195, 197, 199, 201, 203, 205, 207, 209,
+ 211, 213, 215, 217, 219, 221, 223, 225, 227, 229, 231, 233, 235, 237, 239,
+ 241, 243, 245, 247, 249, 251, 253, 127};
// Gets the QP, QP range: [0, 127].
// Returns true on success, false otherwise.
diff --git a/webrtc/modules/video_coding/video_coding_impl.cc b/webrtc/modules/video_coding/video_coding_impl.cc
index 64cc090..c471dda 100644
--- a/webrtc/modules/video_coding/video_coding_impl.cc
+++ b/webrtc/modules/video_coding/video_coding_impl.cc
@@ -8,33 +8,33 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+
+#include <algorithm>
+
#include "webrtc/common_types.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/encoded_frame.h"
#include "webrtc/modules/video_coding/jitter_buffer.h"
#include "webrtc/modules/video_coding/packet.h"
-#include "webrtc/modules/video_coding/video_coding_impl.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
namespace vcm {
-int64_t
-VCMProcessTimer::Period() const {
- return _periodMs;
+int64_t VCMProcessTimer::Period() const {
+ return _periodMs;
}
-int64_t
-VCMProcessTimer::TimeUntilProcess() const {
- const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
- const int64_t time_until_process = _periodMs - time_since_process;
- return std::max<int64_t>(time_until_process, 0);
+int64_t VCMProcessTimer::TimeUntilProcess() const {
+ const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
+ const int64_t time_until_process = _periodMs - time_since_process;
+ return std::max<int64_t>(time_until_process, 0);
}
-void
-VCMProcessTimer::Processed() {
- _latestMs = _clock->TimeInMilliseconds();
+void VCMProcessTimer::Processed() {
+ _latestMs = _clock->TimeInMilliseconds();
}
} // namespace vcm
@@ -59,8 +59,8 @@
const RTPFragmentationHeader* fragmentation) {
CriticalSectionScoped cs(cs_.get());
if (callback_)
- return callback_->Encoded(
- encoded_image, codec_specific_info, fragmentation);
+ return callback_->Encoded(encoded_image, codec_specific_info,
+ fragmentation);
return 0;
}
@@ -84,9 +84,7 @@
receiver_(clock, event_factory),
own_event_factory_(owns_event_factory ? event_factory : NULL) {}
- virtual ~VideoCodingModuleImpl() {
- own_event_factory_.reset();
- }
+ virtual ~VideoCodingModuleImpl() { own_event_factory_.reset(); }
int64_t TimeUntilNextProcess() override {
int64_t sender_time = sender_.TimeUntilNextProcess();
@@ -321,9 +319,8 @@
encoder_rate_observer, qm_settings_callback);
}
-VideoCodingModule* VideoCodingModule::Create(
- Clock* clock,
- EventFactory* event_factory) {
+VideoCodingModule* VideoCodingModule::Create(Clock* clock,
+ EventFactory* event_factory) {
assert(clock);
assert(event_factory);
return new VideoCodingModuleImpl(clock, event_factory, false, nullptr,
diff --git a/webrtc/modules/video_coding/video_coding_robustness_unittest.cc b/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
index 4111109..94c62dc 100644
--- a/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
+++ b/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
@@ -47,9 +47,7 @@
vcm_->RegisterExternalDecoder(&decoder_, video_codec_.plType);
}
- virtual void TearDown() {
- VideoCodingModule::Destroy(vcm_);
- }
+ virtual void TearDown() { VideoCodingModule::Destroy(vcm_); }
void InsertPacket(uint32_t timestamp,
uint16_t seq_no,
@@ -87,19 +85,17 @@
.With(Args<0, 1>(ElementsAre(6, 7)))
.Times(1);
for (int ts = 0; ts <= 6000; ts += 3000) {
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
- Field(&EncodedImage::_length,
- kPayloadLen * 3),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
+ EXPECT_CALL(decoder_,
+ Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
+ Field(&EncodedImage::_length, kPayloadLen * 3),
+ Field(&EncodedImage::_completeFrame, true)),
+ false, _, _, _))
.Times(1)
.InSequence(s);
}
ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kHardNack,
- kNoErrors));
+ VideoCodingModule::kHardNack, kNoErrors));
InsertPacket(0, 0, true, false, kVideoFrameKey);
InsertPacket(0, 1, false, false, kVideoFrameKey);
@@ -136,14 +132,11 @@
}
TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
- EXPECT_CALL(request_callback_, ResendPackets(_, _))
- .Times(0);
- EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
- .Times(1);
+ EXPECT_CALL(request_callback_, ResendPackets(_, _)).Times(0);
+ EXPECT_CALL(frame_type_callback_, RequestKeyFrame()).Times(1);
ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kHardNack,
- kNoErrors));
+ VideoCodingModule::kHardNack, kNoErrors));
InsertPacket(3000, 3, true, false, kVideoFrameDelta);
InsertPacket(3000, 4, false, false, kVideoFrameDelta);
@@ -166,46 +159,43 @@
.With(Args<0, 1>(ElementsAre(4)))
.Times(0);
- EXPECT_CALL(decoder_, Copy())
- .Times(0);
- EXPECT_CALL(decoderCopy_, Copy())
- .Times(0);
+ EXPECT_CALL(decoder_, Copy()).Times(0);
+ EXPECT_CALL(decoderCopy_, Copy()).Times(0);
// Decode operations
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
- Field(&EncodedImage::_completeFrame,
- false)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
+ EXPECT_CALL(decoder_,
+ Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+ Field(&EncodedImage::_completeFrame, true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_,
+ Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+ Field(&EncodedImage::_completeFrame, false)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_,
+ Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+ Field(&EncodedImage::_completeFrame, true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_,
+ Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
+ Field(&EncodedImage::_completeFrame, true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kNone,
- kWithErrors));
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(VideoCodingModule::kNone,
+ kWithErrors));
InsertPacket(0, 0, true, false, kVideoFrameKey);
InsertPacket(0, 1, false, false, kVideoFrameKey);
InsertPacket(0, 2, false, true, kVideoFrameKey);
EXPECT_EQ(VCM_OK, vcm_->Decode(33)); // Decode timestamp 0.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
clock_->AdvanceTimeMilliseconds(33);
InsertPacket(3000, 3, true, false, kVideoFrameDelta);
@@ -223,7 +213,7 @@
clock_->AdvanceTimeMilliseconds(10);
EXPECT_EQ(VCM_OK, vcm_->Decode(23)); // Decode timestamp 6000 complete.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
clock_->AdvanceTimeMilliseconds(23);
InsertPacket(3000, 4, false, false, kVideoFrameDelta);
diff --git a/webrtc/modules/video_coding/video_sender.cc b/webrtc/modules/video_coding/video_sender.cc
index 7d8e97b..3bc3381 100644
--- a/webrtc/modules/video_coding/video_sender.cc
+++ b/webrtc/modules/video_coding/video_sender.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_types.h"
#include <algorithm> // std::max
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/common_types.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/encoded_frame.h"
@@ -126,14 +126,10 @@
_nextFrameTypes.resize(VCM_MAX(sendCodec->numberOfSimulcastStreams, 1),
kVideoFrameDelta);
- _mediaOpt.SetEncodingData(sendCodec->codecType,
- sendCodec->maxBitrate * 1000,
- sendCodec->startBitrate * 1000,
- sendCodec->width,
- sendCodec->height,
- sendCodec->maxFramerate,
- numLayers,
- maxPayloadSize);
+ _mediaOpt.SetEncodingData(sendCodec->codecType, sendCodec->maxBitrate * 1000,
+ sendCodec->startBitrate * 1000, sendCodec->width,
+ sendCodec->height, sendCodec->maxFramerate,
+ numLayers, maxPayloadSize);
return VCM_OK;
}
@@ -158,8 +154,8 @@
// Register an external decoder object.
// This can not be used together with external decoder callbacks.
void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
- uint8_t payloadType,
- bool internalSource /*= false*/) {
+ uint8_t payloadType,
+ bool internalSource /*= false*/) {
RTC_DCHECK(main_thread_.CalledOnValidThread());
rtc::CritScope lock(&send_crit_);
@@ -174,8 +170,8 @@
}
return;
}
- _codecDataBase.RegisterExternalEncoder(
- externalEncoder, payloadType, internalSource);
+ _codecDataBase.RegisterExternalEncoder(externalEncoder, payloadType,
+ internalSource);
}
// Get encode bitrate
diff --git a/webrtc/modules/video_coding/video_sender_unittest.cc b/webrtc/modules/video_coding/video_sender_unittest.cc
index 3f5c3cd..2daa9d7 100644
--- a/webrtc/modules/video_coding/video_sender_unittest.cc
+++ b/webrtc/modules/video_coding/video_sender_unittest.cc
@@ -41,9 +41,7 @@
namespace webrtc {
namespace vcm {
namespace {
-enum {
- kMaxNumberOfTemporalLayers = 3
-};
+enum { kMaxNumberOfTemporalLayers = 3 };
struct Vp8StreamInfo {
float framerate_fps[kMaxNumberOfTemporalLayers];
@@ -87,7 +85,7 @@
class PacketizationCallback : public VCMPacketizationCallback {
public:
- PacketizationCallback(Clock* clock)
+ explicit PacketizationCallback(Clock* clock)
: clock_(clock), start_time_ms_(clock_->TimeInMilliseconds()) {}
virtual ~PacketizationCallback() {}
@@ -211,16 +209,12 @@
memset(&settings_, 0, sizeof(settings_));
EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &settings_));
settings_.numberOfSimulcastStreams = kNumberOfStreams;
- ConfigureStream(kDefaultWidth / 4,
- kDefaultHeight / 4,
- 100,
+ ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, 100,
&settings_.simulcastStream[0]);
- ConfigureStream(kDefaultWidth / 2,
- kDefaultHeight / 2,
- 500,
+ ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, 500,
&settings_.simulcastStream[1]);
- ConfigureStream(
- kDefaultWidth, kDefaultHeight, 1200, &settings_.simulcastStream[2]);
+ ConfigureStream(kDefaultWidth, kDefaultHeight, 1200,
+ &settings_.simulcastStream[2]);
settings_.plType = kUnusedPayloadType; // Use the mocked encoder.
generator_.reset(
new EmptyFrameGenerator(settings_.width, settings_.height));
@@ -244,12 +238,11 @@
assert(stream < kNumberOfStreams);
std::vector<FrameType> frame_types(kNumberOfStreams, kVideoFrameDelta);
frame_types[stream] = kVideoFrameKey;
- EXPECT_CALL(
- encoder_,
- Encode(_,
- _,
- Pointee(ElementsAreArray(&frame_types[0], frame_types.size()))))
- .Times(1).WillRepeatedly(Return(0));
+ EXPECT_CALL(encoder_,
+ Encode(_, _, Pointee(ElementsAreArray(&frame_types[0],
+ frame_types.size()))))
+ .Times(1)
+ .WillRepeatedly(Return(0));
}
static void ConfigureStream(int width,