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Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.

Browse Source 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
solenberg@webrtc.org 2013-12-13 23:57:54 +00:00
parent e1bc6c8d8b
commit 341e91441a
19 changed files with 412 additions and 105 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

37
webrtc/common_types.h
View File

@ -69,27 +69,28 @@ protected:
enum TraceModule
{
kTraceUndefined = 0,
kTraceUndefined = 0,
// not a module, triggered from the engine code
kTraceVoice = 0x0001,
kTraceVoice = 0x0001,
// not a module, triggered from the engine code
kTraceVideo = 0x0002,
kTraceVideo = 0x0002,
// not a module, triggered from the utility code
kTraceUtility = 0x0003,
kTraceRtpRtcp = 0x0004,
kTraceTransport = 0x0005,
kTraceSrtp = 0x0006,
kTraceAudioCoding = 0x0007,
kTraceAudioMixerServer = 0x0008,
kTraceAudioMixerClient = 0x0009,
kTraceFile = 0x000a,
kTraceAudioProcessing = 0x000b,
kTraceVideoCoding = 0x0010,
kTraceVideoMixer = 0x0011,
kTraceAudioDevice = 0x0012,
kTraceVideoRenderer = 0x0014,
kTraceVideoCapture = 0x0015,
kTraceVideoPreocessing = 0x0016
kTraceUtility = 0x0003,
kTraceRtpRtcp = 0x0004,
kTraceTransport = 0x0005,
kTraceSrtp = 0x0006,
kTraceAudioCoding = 0x0007,
kTraceAudioMixerServer = 0x0008,
kTraceAudioMixerClient = 0x0009,
kTraceFile = 0x000a,
kTraceAudioProcessing = 0x000b,
kTraceVideoCoding = 0x0010,
kTraceVideoMixer = 0x0011,
kTraceAudioDevice = 0x0012,
kTraceVideoRenderer = 0x0014,
kTraceVideoCapture = 0x0015,
kTraceVideoPreocessing = 0x0016,
kTraceRemoteBitrateEstimator = 0x0017,
};
enum TraceLevel

2
webrtc/modules/interface/module_common_types.h
View File

@ -28,7 +28,9 @@
namespace webrtc {
struct RTPHeaderExtension {
bool hasTransmissionTimeOffset;
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
uint32_t absoluteSendTime;
};

5
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
View File

@ -17,6 +17,7 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -225,6 +226,8 @@ RemoteBitrateEstimator* RemoteBitrateEstimatorFactory::Create(
RemoteBitrateObserver* observer,
Clock* clock,
uint32_t min_bitrate_bps) const {
WEBRTC_TRACE(kTraceStateInfo, kTraceRemoteBitrateEstimator, -1,
"RemoteBitrateEstimatorFactory: Instantiating.");
return new RemoteBitrateEstimatorSingleStream(observer, clock,
min_bitrate_bps);
}
@ -233,6 +236,8 @@ RemoteBitrateEstimator* AbsoluteSendTimeRemoteBitrateEstimatorFactory::Create(
RemoteBitrateObserver* observer,
Clock* clock,
uint32_t min_bitrate_bps) const {
WEBRTC_TRACE(kTraceStateInfo, kTraceRemoteBitrateEstimator, -1,
"AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
return new RemoteBitrateEstimatorSingleStream(observer, clock,
min_bitrate_bps);
}

4
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
View File

@ -391,9 +391,11 @@ bool RTPHeaderParser::Parse(RTPHeader& header,
// If in effect, MAY be omitted for those packets for which the offset
// is zero.
header.extension.hasTransmissionTimeOffset = false;
header.extension.transmissionTimeOffset = 0;
// May not be present in packet.
header.extension.hasAbsoluteSendTime = false;
header.extension.absoluteSendTime = 0;
if (X) {
@ -490,6 +492,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
// Negative offset, correct sign for Word24 to Word32.
header.extension.transmissionTimeOffset |= 0xFF000000;
}
header.extension.hasTransmissionTimeOffset = true;
break;
}
case kRtpExtensionAudioLevel: {
@ -524,6 +527,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
absoluteSendTime += *ptr++ << 8;
absoluteSendTime += *ptr++;
header.extension.absoluteSendTime = absoluteSendTime;
header.extension.hasAbsoluteSendTime = true;
break;
}
default: {

7
webrtc/system_wrappers/source/trace_impl.cc
View File

@ -273,6 +273,10 @@ int32_t TraceImpl::AddModuleAndId(char* trace_message,
sprintf(trace_message, " VIDEO PROC:%5ld %5ld;", id_engine,
id_channel);
break;
case kTraceRemoteBitrateEstimator:
sprintf(trace_message, " BWE RBE:%5ld %5ld;", id_engine,
id_channel);
break;
}
} else {
switch (module) {
@ -332,6 +336,9 @@ int32_t TraceImpl::AddModuleAndId(char* trace_message,
case kTraceVideoPreocessing:
sprintf(trace_message, " VIDEO PROC:%11ld;", idl);
break;
case kTraceRemoteBitrateEstimator:
sprintf(trace_message, " BWE RBE:%11ld;", idl);
break;
}
}
return kMessageLength;

285
webrtc/video/bitrate_estimator_tests.cc Normal file
View File

@ -0,0 +1,285 @@
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <functional>
#include <list>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator_capturer.h"
namespace webrtc {
static const int kTOFExtensionId = 4;
static const int kASTExtensionId = 5;
static unsigned int kDefaultTimeoutMs = 30 * 1000;
static const uint32_t kSendSsrc = 0x654321;
static const uint32_t kReceiverLocalSsrc = 0x123456;
static const uint8_t kSendPayloadType = 125;
class BitrateEstimatorTest : public ::testing::Test {
public:
BitrateEstimatorTest()
: receiver_trace_(),
send_transport_(),
receive_transport_(),
sender_call_(),
receiver_call_(),
fake_encoder_(Clock::GetRealTimeClock()),
fake_decoder_(),
send_config_(),
receive_config_(),
streams_() {
}
virtual ~BitrateEstimatorTest() {
EXPECT_TRUE(streams_.empty());
}
virtual void SetUp() {
// Create receiver call first so that we are guaranteed to have a trace
// callback when sender call is created.
Call::Config receiver_call_config(&receive_transport_);
receiver_call_config.trace_callback = &receiver_trace_;
receiver_call_.reset(Call::Create(receiver_call_config));
Call::Config sender_call_config(&send_transport_);
sender_call_.reset(Call::Create(sender_call_config));
send_transport_.SetReceiver(receiver_call_->Receiver());
receive_transport_.SetReceiver(sender_call_->Receiver());
send_config_ = sender_call_->GetDefaultSendConfig();
send_config_.rtp.ssrcs.push_back(kSendSsrc);
send_config_.encoder = &fake_encoder_;
send_config_.internal_source = false;
test::FakeEncoder::SetCodecSettings(&send_config_.codec, 1);
send_config_.codec.plType = kSendPayloadType;
receive_config_ = receiver_call_->GetDefaultReceiveConfig();
receive_config_.codecs.clear();
receive_config_.codecs.push_back(send_config_.codec);
ExternalVideoDecoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type = send_config_.codec.plType;
receive_config_.external_decoders.push_back(decoder);
receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
}
virtual void TearDown() {
std::for_each(streams_.begin(), streams_.end(),
std::mem_fun(&Stream::StopSending));
send_transport_.StopSending();
receive_transport_.StopSending();
while (!streams_.empty()) {
delete streams_.back();
streams_.pop_back();
}
// The TraceCallback instance MUST outlive Calls, destroy Calls explicitly.
receiver_call_.reset();
}
protected:
friend class Stream;
class TraceObserver : public TraceCallback {
public:
TraceObserver()
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
received_log_lines_(),
expected_log_lines_(),
done_(EventWrapper::Create()) {
}
void PushExpectedLogLine(const std::string& expected_log_line) {
CriticalSectionScoped cs(crit_sect_.get());
expected_log_lines_.push_back(expected_log_line);
}
virtual void Print(TraceLevel level,
const char* message,
int length) OVERRIDE {
CriticalSectionScoped cs(crit_sect_.get());
if (!(level & kTraceStateInfo)) {
return;
}
std::string msg(message);
if (msg.find("BitrateEstimator") != std::string::npos) {
received_log_lines_.push_back(msg);
}
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
std::string b = expected_log_lines_.front();
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
EXPECT_TRUE(a.find(b) != std::string::npos);
}
if (expected_log_lines_.size() <= 0) {
if (num_popped > 0) {
done_->Set();
}
return;
}
}
EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
private:
typedef std::list<std::string> Strings;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
Strings received_log_lines_;
Strings expected_log_lines_;
scoped_ptr<EventWrapper> done_;
};
class Stream {
public:
explicit Stream(BitrateEstimatorTest* test)
: test_(test),
is_sending_receiving_(false),
send_stream_(NULL),
receive_stream_(NULL),
frame_generator_capturer_() {
test_->send_config_.rtp.ssrcs[0]++;
send_stream_ =
test_->sender_call_->CreateVideoSendStream(test_->send_config_);
frame_generator_capturer_.reset(
test::FrameGeneratorCapturer::Create(send_stream_->Input(),
test_->send_config_.codec.width,
test_->send_config_.codec.height,
30,
Clock::GetRealTimeClock()));
send_stream_->StartSending();
frame_generator_capturer_->Start();
test_->receive_config_.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_);
receive_stream_->StartReceiving();
is_sending_receiving_ = true;
}
~Stream() {
frame_generator_capturer_.reset(NULL);
test_->sender_call_->DestroyVideoSendStream(send_stream_);
send_stream_ = NULL;
test_->receiver_call_->DestroyVideoReceiveStream(receive_stream_);
receive_stream_ = NULL;
}
void StopSending() {
if (is_sending_receiving_) {
frame_generator_capturer_->Stop();
send_stream_->StopSending();
receive_stream_->StopReceiving();
is_sending_receiving_ = false;
}
}
private:
BitrateEstimatorTest* test_;
bool is_sending_receiving_;
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
};
TraceObserver receiver_trace_;
test::DirectTransport send_transport_;
test::DirectTransport receive_transport_;
scoped_ptr<Call> sender_call_;
scoped_ptr<Call> receiver_call_;
test::FakeEncoder fake_encoder_;
test::FakeDecoder fake_decoder_;
VideoSendStream::Config send_config_;
VideoReceiveStream::Config receive_config_;
std::vector<Stream*> streams_;
};
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefault) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToAST) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_trace_.PushExpectedLogLine(
"AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOF) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_trace_.PushExpectedLogLine(
"AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
receiver_trace_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
receiver_trace_.PushExpectedLogLine(
"RemoteBitrateEstimatorFactory: Instantiating.");
streams_.push_back(new Stream(this));
streams_[0]->StopSending();
streams_[1]->StopSending();
EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
}
} // namespace webrtc

4
webrtc/video/call_tests.cc
View File

@ -49,7 +49,7 @@ class CallTest : public ::testing::Test {
receive_stream_(NULL),
fake_encoder_(Clock::GetRealTimeClock()) {}
~CallTest() {
virtual ~CallTest() {
EXPECT_EQ(NULL, send_stream_);
EXPECT_EQ(NULL, receive_stream_);
}
@ -861,7 +861,7 @@ TEST_F(CallTest, SendsAndReceivesMultipleStreams) {
sender_transport.StopSending();
receiver_transport.StopSending();
}
};
TEST_F(CallTest, ObserversEncodedFrames) {
class EncodedFrameTestObserver : public EncodedFrameObserver {

8
webrtc/video/rampup_tests.cc
View File

@ -36,7 +36,7 @@
namespace webrtc {
namespace {
static const int kTOffsetExtensionId = 7;
static const int kAbsoluteSendTimeExtensionId = 7;
static const int kMaxPacketSize = 1500;
}
@ -74,8 +74,8 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTOffsetExtensionId);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
remote_bitrate_estimator_.reset(
@ -220,7 +220,7 @@ class RampUpTest : public ::testing::TestWithParam<bool> {
kRtxSsrcs + kNumberOfStreams);
}
send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
RtpExtension(RtpExtension::kAbsSendTime, kAbsoluteSendTimeExtensionId));
VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);

14
webrtc/video/video_receive_stream.cc
View File

@ -63,6 +63,20 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
rtp_rtcp_->SetRembStatus(channel_, false, config_.rtp.remb);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
int id = config_.rtp.extensions[i].id;
if (extension == RtpExtension::kTOffset) {
if (rtp_rtcp_->SetReceiveTimestampOffsetStatus(channel_, true, id) != 0)
abort();
} else if (extension == RtpExtension::kAbsSendTime) {
if (rtp_rtcp_->SetReceiveAbsoluteSendTimeStatus(channel_, true, id) != 0)
abort();
} else {
abort(); // Unsupported extension.
}
}
network_ = ViENetwork::GetInterface(video_engine);
assert(network_ != NULL);

2
webrtc/video/video_send_stream.cc
View File

@ -10,8 +10,6 @@
#include "webrtc/video/video_send_stream.h"
#include <string.h>
#include <string>
#include <vector>

12
webrtc/video/video_send_stream_tests.cc
View File

@ -190,7 +190,7 @@ void VideoSendStreamTest::SendsSetSsrcs(size_t num_ssrcs,
frame_generator_capturer->Stop();
send_stream_->StopSending();
call->DestroyVideoSendStream(send_stream_);
}
};
TEST_F(VideoSendStreamTest, SendsSetSsrc) { SendsSetSsrcs(1, false); }
@ -249,8 +249,11 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
EXPECT_TRUE(
parser_->Parse(packet, static_cast<int>(length), &header));
if (header.extension.absoluteSendTime > 0)
observation_complete_->Set();
EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
EXPECT_EQ(header.extension.transmissionTimeOffset, 0);
EXPECT_GT(header.extension.absoluteSendTime, 0u);
observation_complete_->Set();
return SEND_PACKET;
}
@ -294,7 +297,10 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
EXPECT_TRUE(
parser_->Parse(packet, static_cast<int>(length), &header));
EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
EXPECT_GT(header.extension.transmissionTimeOffset, 0);
EXPECT_EQ(header.extension.absoluteSendTime, 0u);
observation_complete_->Set();
return SEND_PACKET;

6
webrtc/video_engine/vie_channel.cc
View File

@ -92,7 +92,6 @@ ViEChannel::ViEChannel(int32_t channel_id,
bandwidth_observer_(bandwidth_observer),
send_timestamp_extension_id_(kInvalidRtpExtensionId),
absolute_send_time_extension_id_(kInvalidRtpExtensionId),
receive_absolute_send_time_enabled_(false),
external_transport_(NULL),
decoder_reset_(true),
wait_for_key_frame_(false),
@ -934,14 +933,9 @@ int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) {
}
int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
receive_absolute_send_time_enabled_ = enable;
return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1;
}
bool ViEChannel::GetReceiveAbsoluteSendTimeStatus() const {
return receive_absolute_send_time_enabled_;
}
void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
rtp_rtcp_->SetRtcpXrRrtrStatus(enable);

1
webrtc/video_engine/vie_channel.h
View File

@ -396,7 +396,6 @@ class ViEChannel
scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
int send_timestamp_extension_id_;
int absolute_send_time_extension_id_;
bool receive_absolute_send_time_enabled_;
bool using_packet_spread_;
Transport* external_transport_;

79
webrtc/video_engine/vie_channel_group.cc
View File

@ -16,6 +16,7 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/encoder_state_feedback.h"
#include "webrtc/video_engine/vie_channel.h"
@ -25,18 +26,22 @@
namespace webrtc {
namespace {
static const uint32_t kTimeOffsetSwitchThreshold = 30;
class WrappingBitrateEstimator : public RemoteBitrateEstimator {
public:
WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock,
ProcessThread* process_thread)
WrappingBitrateEstimator(int engine_id, RemoteBitrateObserver* observer,
Clock* clock, ProcessThread* process_thread)
: observer_(observer),
clock_(clock),
process_thread_(process_thread),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
engine_id_(engine_id),
min_bitrate_bps_(30000),
rbe_(RemoteBitrateEstimatorFactory().Create(observer_, clock_,
min_bitrate_bps_)),
receive_absolute_send_time_(false) {
using_absolute_send_time_(false),
packets_since_absolute_send_time_(0) {
assert(process_thread_ != NULL);
process_thread_->RegisterModule(rbe_.get());
}
@ -44,29 +49,11 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
process_thread_->DeRegisterModule(rbe_.get());
}
void SetReceiveAbsoluteSendTimeStatus(bool enable) {
CriticalSectionScoped cs(crit_sect_.get());
if (enable == receive_absolute_send_time_) {
return;
}
process_thread_->DeRegisterModule(rbe_.get());
if (enable) {
rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
observer_, clock_, min_bitrate_bps_));
} else {
rbe_.reset(RemoteBitrateEstimatorFactory().Create(observer_, clock_,
min_bitrate_bps_));
}
process_thread_->RegisterModule(rbe_.get());
receive_absolute_send_time_ = enable;
}
virtual void IncomingPacket(int64_t arrival_time_ms,
int payload_size,
const RTPHeader& header) {
CriticalSectionScoped cs(crit_sect_.get());
PickEstimator(header);
rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
}
@ -97,25 +84,60 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
}
private:
// Instantiate RBE for Time Offset or Absolute Send Time extensions.
void PickEstimator(const RTPHeader& header) {
if (header.extension.hasAbsoluteSendTime) {
// If we see AST in header, switch RBE strategy immediately.
if (!using_absolute_send_time_) {
process_thread_->DeRegisterModule(rbe_.get());
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_),
"WrappingBitrateEstimator: Switching to absolute send time RBE.");
rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
observer_, clock_, min_bitrate_bps_));
process_thread_->RegisterModule(rbe_.get());
using_absolute_send_time_ = true;
}
packets_since_absolute_send_time_ = 0;
} else {
// When we don't see AST, wait for a few packets before going back to TOF.
if (using_absolute_send_time_) {
++packets_since_absolute_send_time_;
if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
process_thread_->DeRegisterModule(rbe_.get());
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_),
"WrappingBitrateEstimator: Switching to transmission time offset "
"RBE.");
rbe_.reset(RemoteBitrateEstimatorFactory().Create(observer_, clock_,
min_bitrate_bps_));
process_thread_->RegisterModule(rbe_.get());
using_absolute_send_time_ = false;
}
}
}
}
RemoteBitrateObserver* observer_;
Clock* clock_;
ProcessThread* process_thread_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
const int engine_id_;
const uint32_t min_bitrate_bps_;
scoped_ptr<RemoteBitrateEstimator> rbe_;
bool receive_absolute_send_time_;
bool using_absolute_send_time_;
uint32_t packets_since_absolute_send_time_;
DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
};
} // namespace
ChannelGroup::ChannelGroup(ProcessThread* process_thread,
ChannelGroup::ChannelGroup(int engine_id, ProcessThread* process_thread,
const Config& config)
: remb_(new VieRemb()),
bitrate_controller_(BitrateController::CreateBitrateController(true)),
call_stats_(new CallStats()),
remote_bitrate_estimator_(new WrappingBitrateEstimator(remb_.get(),
Clock::GetRealTimeClock(), process_thread)),
remote_bitrate_estimator_(new WrappingBitrateEstimator(engine_id,
remb_.get(), Clock::GetRealTimeClock(),
process_thread)),
encoder_state_feedback_(new EncoderStateFeedback()),
process_thread_(process_thread) {
call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
@ -186,9 +208,4 @@ bool ChannelGroup::SetChannelRembStatus(int channel_id, bool sender,
}
return true;
}
void ChannelGroup::SetReceiveAbsoluteSendTimeStatus(bool enable) {
static_cast<WrappingBitrateEstimator*>(remote_bitrate_estimator_.get())->
SetReceiveAbsoluteSendTimeStatus(enable);
}
} // namespace webrtc

3
webrtc/video_engine/vie_channel_group.h
View File

@ -31,7 +31,7 @@ class VieRemb;
// group are assumed to send/receive data to the same end-point.
class ChannelGroup {
public:
ChannelGroup(ProcessThread* process_thread,
ChannelGroup(int engine_id, ProcessThread* process_thread,
const Config& config);
~ChannelGroup();
@ -42,7 +42,6 @@ class ChannelGroup {
bool SetChannelRembStatus(int channel_id, bool sender, bool receiver,
ViEChannel* channel);
void SetReceiveAbsoluteSendTimeStatus(bool enable);
BitrateController* GetBitrateController();
CallStats* GetCallStats();

31
webrtc/video_engine/vie_channel_manager.cc
View File

@ -89,7 +89,7 @@ int ViEChannelManager::CreateChannel(int* channel_id) {
}
// Create a new channel group and add this channel.
ChannelGroup* group = new ChannelGroup(module_process_thread_,
ChannelGroup* group = new ChannelGroup(engine_id_, module_process_thread_,
config_);
BitrateController* bitrate_controller = group->GetBitrateController();
ViEEncoder* vie_encoder = new ViEEncoder(engine_id_, new_channel_id,
@ -366,35 +366,6 @@ bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
return group->SetChannelRembStatus(channel_id, sender, receiver, channel);
}
bool ViEChannelManager::SetReceiveAbsoluteSendTimeStatus(int channel_id,
bool enable,
int id) {
CriticalSectionScoped cs(channel_id_critsect_);
ViEChannel* channel = ViEChannelPtr(channel_id);
if (!channel) {
return false;
}
if (channel->SetReceiveAbsoluteSendTimeStatus(enable, id) != 0) {
return false;
}
// Enable absolute send time extension on the group if at least one of the
// channels use it.
ChannelGroup* group = FindGroup(channel_id);
assert(group);
bool any_enabled = false;
for (ChannelMap::const_iterator c_it = channel_map_.begin();
c_it != channel_map_.end(); ++c_it) {
if (group->HasChannel(c_it->first) &&
c_it->second->GetReceiveAbsoluteSendTimeStatus()) {
any_enabled = true;
break;
}
}
group->SetReceiveAbsoluteSendTimeStatus(any_enabled);
return true;
}
void ViEChannelManager::UpdateSsrcs(int channel_id,
const std::list<unsigned int>& ssrcs) {
CriticalSectionScoped cs(channel_id_critsect_);

4
webrtc/video_engine/vie_channel_manager.h
View File

@ -74,10 +74,6 @@ class ViEChannelManager: private ViEManagerBase {
// Adds a channel to include when sending REMB.
bool SetRembStatus(int channel_id, bool sender, bool receiver);
// Switches a channel and its associated group to use (or not) the absolute
// send time header extension with |id|.
bool SetReceiveAbsoluteSendTimeStatus(int channel_id, bool enable, int id);
// Updates the SSRCs for a channel. If one of the SSRCs already is registered,
// it will simply be ignored and no error is returned.
void UpdateSsrcs(int channel_id, const std::list<unsigned int>& ssrcs);

12
webrtc/video_engine/vie_rtp_rtcp_impl.cc
View File

@ -796,8 +796,16 @@ int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
ViEId(shared_data_->instance_id(), video_channel),
"ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(%d, %d, %d)",
video_channel, enable, id);
if (!shared_data_->channel_manager()->SetReceiveAbsoluteSendTimeStatus(
video_channel, enable, id)) {
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
ViEChannel* vie_channel = cs.Channel(video_channel);
if (!vie_channel) {
WEBRTC_TRACE(kTraceError, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
return -1;
}
if (vie_channel->SetReceiveAbsoluteSendTimeStatus(enable, id) != 0) {
shared_data_->SetLastError(kViERtpRtcpUnknownError);
return -1;
}

1
webrtc/webrtc_tests.gypi
View File

@ -33,6 +33,7 @@
'target_name': 'video_engine_tests',
'type': '<(gtest_target_type)',
'sources': [
'video/bitrate_estimator_tests.cc',
'video/call_tests.cc',
'video/video_send_stream_tests.cc',
'test/common_unittest.cc',