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Porting auto mute to new ViE API

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This CL also includes tests for the auto mute function. A few minor lint
warnings were fixed too. Note that the auto mute function is still work
in progress.

The callback ViEEncoderObserver::VideoAutoMuted was not ported from the
old API. This is TBD; see issue 2457.

BUG=2436
R=holmer@google.com, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5021 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2013-10-23 11:04:57 +00:00
parent 886aef09a8
commit ba975e2078
4 changed files with 149 additions and 10 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/video_engine/internal/video_send_stream.cc
View File

@ -12,6 +12,7 @@
#include <string.h>
#include <string>
#include <vector>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@ -195,6 +196,13 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
image_process_ = ViEImageProcess::GetInterface(video_engine);
image_process_->RegisterPreEncodeCallback(channel_,
config_.pre_encode_callback);
if (config.auto_muter.threshold_bps > 0) {
assert(config.auto_muter.window_bps >= 0);
codec_->EnableAutoMuting(channel_,
config.auto_muter.threshold_bps,
config.auto_muter.window_bps);
}
}
VideoSendStream::~VideoSendStream() {

10
webrtc/video_engine/new_include/video_send_stream.h
View File

@ -146,6 +146,16 @@ class VideoSendStream {
// Set to resume a previously destroyed send stream.
SendStreamState* start_state;
// Parameters for auto muter. If threshold_bps > 0, video will be muted when
// the bandwidth estimate drops below this limit, and enabled again when the
// bandwidth estimate goes above threshold_bps + window_bps. Setting the
// threshold to zero disables the auto muter.
struct AutoMuter {
AutoMuter() : threshold_bps(0), window_bps(0) {}
int threshold_bps;
int window_bps;
} auto_muter;
};
// Gets interface used to insert captured frames. Valid as long as the

3
webrtc/video_engine/test/common/fake_encoder.cc
View File

@ -100,9 +100,8 @@ int32_t FakeEncoder::Encode(
int stream_bits = (bits_available > max_stream_bits) ? max_stream_bits :
bits_available;
int stream_bytes = (stream_bits + 7) / 8;
EXPECT_LT(static_cast<size_t>(stream_bytes), sizeof(encoded_buffer_));
if (static_cast<size_t>(stream_bytes) > sizeof(encoded_buffer_))
return -1;
stream_bytes = sizeof(encoded_buffer_);
EncodedImage encoded(
encoded_buffer_, stream_bytes, sizeof(encoded_buffer_));

138
webrtc/video_engine/test/send_stream_tests.cc
View File

@ -8,15 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_video/interface/i420_video_frame.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.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/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/video_engine/internal/transport_adapter.h"
#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/new_include/frame_callback.h"
#include "webrtc/video_engine/new_include/video_send_stream.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/fake_encoder.h"
@ -178,7 +181,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
static const uint8_t kTOffsetExtensionId = 13;
class DelayedEncoder : public test::FakeEncoder {
public:
DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
explicit DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
virtual int32_t Encode(
const I420VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
@ -220,12 +223,12 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
RunSendTest(call.get(), send_config, &observer);
}
class LossyReceiveStatistics : public NullReceiveStatistics {
class FakeReceiveStatistics : public NullReceiveStatistics {
public:
LossyReceiveStatistics(uint32_t send_ssrc,
uint32_t last_sequence_number,
uint32_t cumulative_lost,
uint8_t fraction_lost)
FakeReceiveStatistics(uint32_t send_ssrc,
uint32_t last_sequence_number,
uint32_t cumulative_lost,
uint8_t fraction_lost)
: lossy_stats_(new LossyStatistician(last_sequence_number,
cumulative_lost,
fraction_lost)) {
@ -300,7 +303,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
// Send lossy receive reports to trigger FEC enabling.
if (send_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
LossyReceiveStatistics lossy_receive_stats(
FakeReceiveStatistics lossy_receive_stats(
kSendSsrc, header.sequenceNumber, send_count_ / 2, 127);
RTCPSender rtcp_sender(
0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
@ -353,7 +356,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
void VideoSendStreamTest::TestNackRetransmission(uint32_t retransmit_ssrc) {
class NackObserver : public SendTransportObserver {
public:
NackObserver(uint32_t retransmit_ssrc)
explicit NackObserver(uint32_t retransmit_ssrc)
: SendTransportObserver(30 * 1000),
transport_adapter_(&transport_),
send_count_(0),
@ -402,6 +405,7 @@ void VideoSendStreamTest::TestNackRetransmission(uint32_t retransmit_ssrc) {
return true;
}
private:
internal::TransportAdapter transport_adapter_;
test::DirectTransport transport_;
@ -477,4 +481,122 @@ TEST_F(VideoSendStreamTest, MaxPacketSize) {
RunSendTest(call.get(), send_config, &observer);
}
// The test will go through a number of phases.
// 1. Start sending packets.
// 2. As soon as the RTP stream has been detected, signal a low REMB value to
// activate the auto muter.
// 3. Wait until |kMuteTimeFrames| have been captured without seeing any RTP
// packets.
// 4. Signal a high REMB and the wait for the RTP stream to start again.
// When the stream is detected again, the test ends.
TEST_F(VideoSendStreamTest, AutoMute) {
static const int kMuteTimeFrames = 60; // Mute for 2 seconds @ 30 fps.
static const int kMuteThresholdBps = 70000;
static const int kMuteWindowBps = 10000;
// Let the low REMB value be 10 kbps lower than the muter threshold, and the
// high REMB value be 5 kbps higher than the re-enabling threshold.
static const int kLowRembBps = kMuteThresholdBps - 10000;
static const int kHighRembBps = kMuteThresholdBps + kMuteWindowBps + 5000;
class RembObserver : public SendTransportObserver, public I420FrameCallback {
public:
RembObserver()
: SendTransportObserver(30 * 1000), // Timeout after 30 seconds.
transport_adapter_(&transport_),
clock_(Clock::GetRealTimeClock()),
test_state_(kBeforeMute),
rtp_count_(0),
last_sequence_number_(0),
mute_frame_count_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
void SetReceiver(PacketReceiver* receiver) {
transport_.SetReceiver(receiver);
}
virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
// Receive statistics reporting having lost 0% of the packets.
// This is needed for the send-side bitrate controller to work properly.
CriticalSectionScoped lock(crit_sect_.get());
SendRtcpFeedback(0); // REMB is only sent if value is > 0.
return true;
}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++rtp_count_;
RTPHeader header;
EXPECT_TRUE(
rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
last_sequence_number_ = header.sequenceNumber;
if (test_state_ == kBeforeMute) {
// The stream has started. Try to mute it.
SendRtcpFeedback(kLowRembBps);
test_state_ = kDuringMute;
} else if (test_state_ == kDuringMute) {
mute_frame_count_ = 0;
} else if (test_state_ == kWaitingForPacket) {
send_test_complete_->Set();
}
return true;
}
// This method implements the I420FrameCallback.
void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
if (test_state_ == kDuringMute && ++mute_frame_count_ > kMuteTimeFrames) {
SendRtcpFeedback(kHighRembBps);
test_state_ = kWaitingForPacket;
}
}
private:
enum TestState {
kBeforeMute,
kDuringMute,
kWaitingForPacket,
kAfterMute
};
virtual void SendRtcpFeedback(int remb_value) {
FakeReceiveStatistics receive_stats(
kSendSsrc, last_sequence_number_, rtp_count_, 0);
RTCPSender rtcp_sender(0, false, clock_, &receive_stats);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrc);
if (remb_value > 0) {
rtcp_sender.SetREMBStatus(true);
rtcp_sender.SetREMBData(remb_value, 0, NULL);
}
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
}
internal::TransportAdapter transport_adapter_;
test::DirectTransport transport_;
Clock* clock_;
TestState test_state_;
int rtp_count_;
int last_sequence_number_;
int mute_frame_count_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
} observer;
Call::Config call_config(&observer);
scoped_ptr<Call> call(Call::Create(call_config));
observer.SetReceiver(call->Receiver());
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.nack.rtp_history_ms = 1000;
send_config.auto_muter.threshold_bps = kMuteThresholdBps;
send_config.auto_muter.window_bps = kMuteWindowBps;
send_config.pre_encode_callback = &observer;
RunSendTest(call.get(), send_config, &observer);
}
} // namespace webrtc