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Lint fix for webrtc/modules/video_coding PART 3!

Browse Source 
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}
This commit is contained in:
philipel 2015-12-21 08:23:20 -08:00 committed by Commit bot
parent f5b1abf5b0
commit 5908c71128
29 changed files with 662 additions and 767 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
PRESUBMIT.py
View File

@ -22,6 +22,7 @@ CPPLINT_DIRS = [
'webrtc/examples',
'webrtc/modules/remote_bitrate_estimator',
'webrtc/modules/rtp_rtcp',
'webrtc/modules/video_coding',
'webrtc/modules/video_processing',
'webrtc/sound',
'webrtc/tools',

8
webrtc/modules/video_coding/test/receiver_tests.h
View File

@ -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(

33
webrtc/modules/video_coding/test/rtp_player.cc
View File

@ -41,7 +41,9 @@ enum {
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),
@ -237,11 +239,8 @@ class SsrcHandlers {
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 @@ class SsrcHandlers {
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 @@ class SsrcHandlers {
}
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 @@ class RtpPlayerImpl : public RtpPlayerInterface {
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 @@ class RtpPlayerImpl : public RtpPlayerInterface {
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;
}
}
@ -438,7 +434,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
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);
@ -470,8 +466,11 @@ class RtpPlayerImpl : public RtpPlayerInterface {
};
RtpPlayerInterface* Create(const std::string& input_filename,
PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
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,

13
webrtc/modules/video_coding/test/rtp_player.h
View File

@ -24,12 +24,12 @@ namespace rtpplayer {
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_; }
@ -87,8 +87,11 @@ class RtpPlayerInterface {
};
RtpPlayerInterface* Create(const std::string& inputFilename,
PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
PayloadSinkFactoryInterface* payloadSinkFactory,
Clock* clock,
const PayloadTypes& payload_types,
float lossRate,
int64_t rttMs,
bool reordering);
} // namespace rtpplayer

15
webrtc/modules/video_coding/test/stream_generator.cc
View File

@ -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 @@ void StreamGenerator::GenerateFrame(FrameType type,
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 @@ bool StreamGenerator::NextPacket(VCMPacket* packet) {
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 @@ uint16_t StreamGenerator::NextSequenceNumber() const {
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();

27
webrtc/modules/video_coding/test/test_util.cc
View File

@ -28,12 +28,12 @@ CmdArgs::CmdArgs()
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);
@ -50,21 +50,24 @@ void SplitFilename(const std::string& filename, std::string* basename,
}
}
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 @@ FileOutputFrameReceiver::FileOutputFrameReceiver(
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 @@ int32_t FileOutputFrameReceiver::FrameToRender(
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) {

12
webrtc/modules/video_coding/test/test_util.h
View File

@ -33,11 +33,13 @@ class NullEvent : public webrtc::EventWrapper {
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 @@ class NullEventFactory : public webrtc::EventFactory {
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 @@ class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
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_;

47
webrtc/modules/video_coding/test/tester_main.cc
View File

@ -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_string(codec, "VP8", "Codec to use (VP8 or I420).");
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
@ -36,28 +38,29 @@ using namespace webrtc;
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) {
// Initialize WebRTC fileutils.h so paths to resources can be resolved.
@ -65,7 +68,7 @@ int main(int argc, char **argv) {
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;
}

10
webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
View File

@ -22,8 +22,7 @@
namespace webrtc {
namespace rtpplayer {
class VcmPayloadSinkFactory::VcmPayloadSink
: public PayloadSinkInterface,
class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
public VCMPacketRequestCallback {
public:
VcmPayloadSink(VcmPayloadSinkFactory* factory,
@ -43,9 +42,7 @@ class VcmPayloadSinkFactory::VcmPayloadSink
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 @@ PayloadSinkInterface* VcmPayloadSinkFactory::Create(
}
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;

6
webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
View File

@ -29,9 +29,11 @@ namespace rtpplayer {
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();

6
webrtc/modules/video_coding/test/video_rtp_play.cc
View File

@ -48,9 +48,9 @@ int RtpPlay(const CmdArgs& args) {
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,

23
webrtc/modules/video_coding/test/video_source.h
View File

@ -11,13 +11,12 @@
#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
{
enum VideoSize {
kUndefined,
kSQCIF, // 128*96 = 12 288
kQQVGA, // 160*120 = 19 200
@ -50,14 +49,18 @@ enum VideoSize
kNumberOfVideoSizes
};
class VideoSource
{
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);
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; }

8
webrtc/modules/video_coding/timestamp_map.cc
View File

@ -20,11 +20,9 @@ VCMTimestampMap::VCMTimestampMap(size_t capacity)
: 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 @@ VCMFrameInformation* VCMTimestampMap::Pop(uint32_t timestamp) {
bool VCMTimestampMap::IsEmpty() const {
return (next_add_idx_ == next_pop_idx_);
}
}
} // namespace webrtc

39
webrtc/modules/video_coding/timing.cc
View File

@ -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 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
// 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
@ -130,8 +130,10 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
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;
@ -193,8 +195,8 @@ void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
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;
@ -214,19 +216,19 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
}
// 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 @@ uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
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 @@ bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
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 {

3
webrtc/modules/video_coding/timing.h
View File

@ -25,8 +25,7 @@ class VCMTiming {
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.

16
webrtc/modules/video_coding/timing_unittest.cc
View File

@ -55,8 +55,9 @@ TEST(ReceiverTiming, Tests) {
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 @@ TEST(ReceiverTiming, Tests) {
for (int i = 0; i < 10; i++) {
int64_t startTimeMs = clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(10);
timing.StopDecodeTimer(timeStamp,
clock.TimeInMilliseconds() - startTimeMs,
timing.StopDecodeTimer(
timeStamp, clock.TimeInMilliseconds() - startTimeMs,
clock.TimeInMilliseconds(),
timing.RenderTimeMs(
timeStamp, clock.TimeInMilliseconds()));
timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()));
timeStamp += 90000 / 25;
clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
@ -137,8 +137,8 @@ TEST(ReceiverTiming, WrapAround) {
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());

213
webrtc/modules/video_coding/utility/frame_dropper.cc
View File

@ -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;
@ -22,32 +21,27 @@ const float kDefaultDropRatioMax = 0.96f;
const float kDefaultMaxTimeToDropFrames = 4.0f; // In seconds.
FrameDropper::FrameDropper()
:
_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
: _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
_keyFrameRatio(kDefaultKeyFrameRatio),
_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
_enabled(true),
_max_time_drops(kDefaultMaxTimeToDropFrames)
{
_max_time_drops(kDefaultMaxTimeToDropFrames) {
Reset();
}
FrameDropper::FrameDropper(float max_time_drops)
:
_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
: _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
_keyFrameRatio(kDefaultKeyFrameRatio),
_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
_enabled(true),
_max_time_drops(max_time_drops)
{
_max_time_drops(max_time_drops) {
Reset();
}
void
FrameDropper::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
_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;
@ -68,54 +62,41 @@ FrameDropper::Reset()
_max_time_drops = 4.0f;
}
void
FrameDropper::Enable(bool enable)
{
void FrameDropper::Enable(bool enable) {
_enabled = enable;
}
void
FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame)
{
if (!_enabled)
{
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
{
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())
{
if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered()) {
// Remove the average key frame size since we
// compensate for key frames when adding delta
// frames.
frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
}
else
{
} else {
// Shouldn't be negative, so zero is the lower bound.
frameSizeKbits = 0;
}
if (_keyFrameRatio.filtered() > 1e-5 &&
1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
{
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
{
} else {
// Compensate for the key frame the following frames
_keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
}
}
else
{
} else {
// Decrease the keyFrameRatio
_keyFrameRatio.Apply(1.0, 0.0);
}
@ -124,201 +105,151 @@ FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame)
CapAccumulator();
}
void
FrameDropper::Leak(uint32_t inputFrameRate)
{
if (!_enabled)
{
void FrameDropper::Leak(uint32_t inputFrameRate) {
if (!_enabled) {
return;
}
if (inputFrameRate < 1)
{
if (inputFrameRate < 1) {
return;
}
if (_targetBitRate < 0.0f)
{
if (_targetBitRate < 0.0f) {
return;
}
_keyFrameSpreadFrames = 0.5f * inputFrameRate;
// T is the expected bits per frame (target). If all frames were the same size,
// 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)
{
if (_keyFrameCount > 0) {
// Perform the key frame compensation
if (_keyFrameRatio.filtered() > 0 &&
1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
{
1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
}
else
{
} else {
T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
}
_keyFrameCount--;
}
_accumulator -= T;
if (_accumulator < 0.0f)
{
if (_accumulator < 0.0f) {
_accumulator = 0.0f;
}
UpdateRatio();
}
void
FrameDropper::UpdateNack(uint32_t nackBytes)
{
if (!_enabled)
{
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)
{
void FrameDropper::FillBucket(float inKbits, float outKbits) {
_accumulator += (inKbits - outKbits);
}
void
FrameDropper::UpdateRatio()
{
if (_accumulator > 1.3f * _accumulatorMax)
{
void FrameDropper::UpdateRatio() {
if (_accumulator > 1.3f * _accumulatorMax) {
// Too far above accumulator max, react faster
_dropRatio.UpdateBase(0.8f);
}
else
{
} else {
// Go back to normal reaction
_dropRatio.UpdateBase(0.9f);
}
if (_accumulator > _accumulatorMax)
{
if (_accumulator > _accumulatorMax) {
// We are above accumulator max, and should ideally
// drop a frame. Increase the dropRatio and drop
// the frame later.
if (_wasBelowMax)
{
if (_wasBelowMax) {
_dropNext = true;
}
if (_fastMode)
{
if (_fastMode) {
// always drop in aggressive mode
_dropNext = true;
}
_dropRatio.Apply(1.0f, 1.0f);
_dropRatio.UpdateBase(0.9f);
}
else
{
} 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)
{
bool FrameDropper::DropFrame() {
if (!_enabled) {
return false;
}
if (_dropNext)
{
if (_dropNext) {
_dropNext = false;
_dropCount = 0;
}
if (_dropRatio.filtered() >= 0.5f) // Drops per keep
{
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;
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);
int max_limit = static_cast<int>(_incoming_frame_rate * _max_time_drops);
if (limit > max_limit) {
limit = max_limit;
}
if (_dropCount < 0)
{
if (_dropCount < 0) {
// Reset the _dropCount since it was negative and should be positive.
if (_dropRatio.filtered() > 0.4f)
{
if (_dropRatio.filtered() > 0.4f) {
_dropCount = -_dropCount;
}
else
{
} else {
_dropCount = 0;
}
}
if (_dropCount < limit)
{
if (_dropCount < limit) {
// As long we are below the limit we should drop frames.
_dropCount++;
return true;
}
else
{
} 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
{
} 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;
if (denom < 1e-5) {
denom = 1e-5f;
}
int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
if (_dropCount > 0)
{
if (_dropCount > 0) {
// Reset the _dropCount since we have a positive
// _dropCount, and it should be negative.
if (_dropRatio.filtered() < 0.6f)
{
if (_dropRatio.filtered() < 0.6f) {
_dropCount = -_dropCount;
}
else
{
} else {
_dropCount = 0;
}
}
if (_dropCount > limit)
{
if (_dropCount == 0)
{
if (_dropCount > limit) {
if (_dropCount == 0) {
// Drop frames when we reset _dropCount.
_dropCount--;
return true;
}
else
{
} else {
// Keep frames as long as we haven't reached limit.
_dropCount--;
return false;
}
}
else
{
} else {
_dropCount = 0;
return false;
}
@ -332,13 +263,11 @@ FrameDropper::DropFrame()
// return dropNext;
}
void
FrameDropper::SetRates(float bitRate, float 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)
{
if (_targetBitRate > 0.0f && bitRate < _targetBitRate &&
_accumulator > _accumulatorMax) {
// Rescale the accumulator level if the accumulator max decreases
_accumulator = bitRate / _targetBitRate * _accumulator;
}
@ -347,11 +276,8 @@ FrameDropper::SetRates(float bitRate, float incoming_frame_rate)
_incoming_frame_rate = incoming_frame_rate;
}
float
FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const
{
if (!_enabled)
{
float FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const {
if (!_enabled) {
return static_cast<float>(inputFrameRate);
}
return inputFrameRate * (1.0f - _dropRatio.filtered());
@ -366,5 +292,4 @@ void FrameDropper::CapAccumulator() {
_accumulator = max_accumulator;
}
}
}
} // namespace webrtc

21
webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
View File

@ -20,20 +20,13 @@ namespace webrtc {
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

4
webrtc/modules/video_coding/utility/moving_average.h
View File

@ -31,8 +31,8 @@ class MovingAverage {
};
template <class T>
MovingAverage<T>::MovingAverage() : sum_(static_cast<T>(0)) {
}
MovingAverage<T>::MovingAverage()
: sum_(static_cast<T>(0)) {}
template <class T>
void MovingAverage<T>::AddSample(T sample) {

12
webrtc/modules/video_coding/utility/quality_scaler.cc
View File

@ -26,8 +26,7 @@ QualityScaler::QualityScaler()
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,
@ -124,13 +123,8 @@ const VideoFrame& QualityScaler::GetScaledFrame(const VideoFrame& frame) {
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;

28
webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
View File

@ -33,6 +33,7 @@ class QualityScalerTest : public ::testing::Test {
int width;
int height;
};
protected:
enum ScaleDirection {
kKeepScaleAtHighQp,
@ -43,8 +44,8 @@ class QualityScalerTest : public ::testing::Test {
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 @@ class QualityScalerTest : public ::testing::Test {
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);
@ -229,14 +231,14 @@ TEST_F(QualityScalerTest,
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 @@ TEST_F(QualityScalerTest, DoesNotDownscaleFrom1Px) {
}
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 @@ QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
}
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 @@ void QualityScalerTest::VerifyQualityAdaptation(
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);

42
webrtc/modules/video_coding/utility/vp8_header_parser.h
View File

@ -38,36 +38,24 @@ struct VP8BitReader {
};
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.

25
webrtc/modules/video_coding/video_coding_impl.cc
View File

@ -8,32 +8,32 @@
* 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 {
int64_t VCMProcessTimer::Period() const {
return _periodMs;
}
int64_t
VCMProcessTimer::TimeUntilProcess() const {
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() {
void VCMProcessTimer::Processed() {
_latestMs = _clock->TimeInMilliseconds();
}
} // namespace vcm
@ -59,8 +59,8 @@ class EncodedImageCallbackWrapper : public EncodedImageCallback {
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 @@ class VideoCodingModuleImpl : public VideoCodingModule {
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();
@ -321,8 +319,7 @@ VideoCodingModule* VideoCodingModule::Create(
encoder_rate_observer, qm_settings_callback);
}
VideoCodingModule* VideoCodingModule::Create(
Clock* clock,
VideoCodingModule* VideoCodingModule::Create(Clock* clock,
EventFactory* event_factory) {
assert(clock);
assert(event_factory);

58
webrtc/modules/video_coding/video_coding_robustness_unittest.cc
View File

@ -47,9 +47,7 @@ class VCMRobustnessTest : public ::testing::Test {
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 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
.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)),
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, TestHardNack) {
}
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);
@ -166,39 +159,36 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
.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)),
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)),
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)),
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)),
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,
ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(VideoCodingModule::kNone,
kWithErrors));
InsertPacket(0, 0, true, false, kVideoFrameKey);

18
webrtc/modules/video_coding/video_sender.cc
View File

@ -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 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
_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;
}
@ -174,8 +170,8 @@ void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
}
return;
}
_codecDataBase.RegisterExternalEncoder(
externalEncoder, payloadType, internalSource);
_codecDataBase.RegisterExternalEncoder(externalEncoder, payloadType,
internalSource);
}
// Get encode bitrate

29
webrtc/modules/video_coding/video_sender_unittest.cc
View File

@ -41,9 +41,7 @@ using webrtc::test::FrameGenerator;
namespace webrtc {
namespace vcm {
namespace {
enum {
kMaxNumberOfTemporalLayers = 3
};
enum { kMaxNumberOfTemporalLayers = 3 };
struct Vp8StreamInfo {
float framerate_fps[kMaxNumberOfTemporalLayers];
@ -87,7 +85,7 @@ class EmptyFrameGenerator : public FrameGenerator {
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 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
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 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
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,