Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate

This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)

Original reverted cl is in patch set #1.
Changes in following patch sets.

The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()

It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.

Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)

BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True  // Due to bug  in android_x86 cq builder....

Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
This commit is contained in:
perkj
2016-05-11 06:01:13 -07:00
committed by Commit bot
parent 2f5ae66471
commit ec81bcd519
29 changed files with 575 additions and 372 deletions

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@ -83,6 +83,10 @@ BitrateController* BitrateController::CreateBitrateController(
return new BitrateControllerImpl(clock, observer);
}
BitrateController* BitrateController::CreateBitrateController(Clock* clock) {
return new BitrateControllerImpl(clock, nullptr);
}
BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
BitrateObserver* observer)
: clock_(clock),
@ -94,8 +98,8 @@ BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
last_fraction_loss_(0),
last_rtt_ms_(0),
last_reserved_bitrate_bps_(0) {
// This calls the observer_, which means that the observer provided by the
// user must be ready to accept a bitrate update when it constructs the
// This calls the observer_ if set, which means that the observer provided by
// the user must be ready to accept a bitrate update when it constructs the
// controller. We do this to avoid having to keep synchronized initial values
// in both the controller and the allocator.
MaybeTriggerOnNetworkChanged();
@ -199,11 +203,15 @@ void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
}
void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
uint32_t bitrate;
if (!observer_)
return;
uint32_t bitrate_bps;
uint8_t fraction_loss;
int64_t rtt;
if (GetNetworkParameters(&bitrate, &fraction_loss, &rtt))
observer_->OnNetworkChanged(bitrate, fraction_loss, rtt);
if (GetNetworkParameters(&bitrate_bps, &fraction_loss, &rtt))
observer_->OnNetworkChanged(bitrate_bps, fraction_loss, rtt);
}
bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,

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@ -28,6 +28,8 @@ namespace webrtc {
class BitrateControllerImpl : public BitrateController {
public:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// |observer| is left for project that is not yet updated.
BitrateControllerImpl(Clock* clock, BitrateObserver* observer);
virtual ~BitrateControllerImpl() {}
@ -50,6 +52,11 @@ class BitrateControllerImpl : public BitrateController {
void SetEventLog(RtcEventLog* event_log) override;
// Returns true if the parameters have changed since the last call.
bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt) override;
int64_t TimeUntilNextProcess() override;
void Process() override;
@ -64,20 +71,16 @@ class BitrateControllerImpl : public BitrateController {
int number_of_packets,
int64_t now_ms);
// Deprecated
void MaybeTriggerOnNetworkChanged();
// Returns true if the parameters have changed since the last call.
bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt);
void OnNetworkChanged(uint32_t bitrate,
uint8_t fraction_loss, // 0 - 255.
int64_t rtt) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
// Used by process thread.
Clock* clock_;
BitrateObserver* observer_;
Clock* const clock_;
BitrateObserver* const observer_;
int64_t last_bitrate_update_ms_;
rtc::CriticalSection critsect_;

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@ -14,11 +14,16 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
using webrtc::RtcpBandwidthObserver;
using webrtc::BitrateObserver;
using ::testing::Exactly;
using ::testing::Return;
using webrtc::BitrateController;
using webrtc::BitrateObserver;
using webrtc::PacedSender;
using webrtc::RtcpBandwidthObserver;
uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1,
int num_packets2, uint8_t fraction_loss2) {

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@ -18,6 +18,7 @@
#include <map>
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
@ -26,6 +27,8 @@ class CriticalSectionWrapper;
class RtcEventLog;
struct PacketInfo;
// Deprecated
// TODO(perkj): Remove BitrateObserver when no implementations use it.
class BitrateObserver {
// Observer class for bitrate changes announced due to change in bandwidth
// estimate or due to bitrate allocation changes. Fraction loss and rtt is
@ -46,10 +49,15 @@ class BitrateController : public Module {
// estimation and divide the available bitrate between all its registered
// BitrateObservers.
public:
static const int kDefaultStartBitrateKbps = 300;
static const int kDefaultStartBitratebps = 300000;
// Deprecated:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// Remove this method once other other projects does not use it.
static BitrateController* CreateBitrateController(Clock* clock,
BitrateObserver* observer);
static BitrateController* CreateBitrateController(Clock* clock);
virtual ~BitrateController() {}
virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
@ -71,6 +79,10 @@ class BitrateController : public Module {
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
virtual bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_

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@ -39,6 +39,8 @@ class MockBitrateController : public BitrateController {
MOCK_METHOD1(SetEventLog, void(RtcEventLog* event_log));
MOCK_CONST_METHOD1(AvailableBandwidth, bool(uint32_t* bandwidth));
MOCK_METHOD1(SetReservedBitrate, void(uint32_t reserved_bitrate_bps));
MOCK_METHOD3(GetNetworkParameters,
bool(uint32_t* bitrate, uint8_t* fraction_loss, int64_t* rtt));
MOCK_METHOD0(Process, void());
MOCK_METHOD0(TimeUntilNextProcess, int64_t());