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Reland of Periodically update codec bit/frame rate settings.

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Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.

> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6

BUG=webrtc:7664

Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
This commit is contained in:
sprang
2017-06-27 07:06:52 -07:00
committed by Commit Bot
parent f0a6fb19c6
commit 4847ae6b51
9 changed files with 462 additions and 298 deletions
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1
webrtc/call/rampup_tests.cc
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@ -54,6 +54,7 @@ RampUpTester::RampUpTester(size_t num_video_streams,
report_perf_stats_(report_perf_stats),
sender_call_(nullptr),
send_stream_(nullptr),
send_transport_(nullptr),
start_bitrate_bps_(start_bitrate_bps),
min_run_time_ms_(min_run_time_ms),
expected_bitrate_bps_(0),

100
webrtc/media/engine/simulcast.cc
View File

@ -49,7 +49,7 @@ const SimulcastFormat kSimulcastFormats[] = {
{0, 0, 1, 200, 150, 30}
};
const int kMaxScreenshareSimulcastStreams = 2;
const int kDefaultScreenshareSimulcastStreams = 2;
// Multiway: Number of temporal layers for each simulcast stream, for maximum
// possible number of simulcast streams |kMaxSimulcastStreams|. The array
@ -176,12 +176,8 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
bool is_screencast) {
size_t num_simulcast_layers;
if (is_screencast) {
if (UseSimulcastScreenshare()) {
num_simulcast_layers =
std::min<int>(max_streams, kMaxScreenshareSimulcastStreams);
} else {
num_simulcast_layers = 1;
}
num_simulcast_layers =
UseSimulcastScreenshare() ? kDefaultScreenshareSimulcastStreams : 1;
} else {
num_simulcast_layers = FindSimulcastMaxLayers(width, height);
}
@ -198,60 +194,33 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
std::vector<webrtc::VideoStream> streams;
streams.resize(num_simulcast_layers);
if (is_screencast) {
ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
// For legacy screenshare in conference mode, tl0 and tl1 bitrates are
// piggybacked on the VideoCodec struct as target and max bitrates,
// respectively. See eg. webrtc::VP8EncoderImpl::SetRates().
streams[0].width = width;
streams[0].height = height;
streams[0].max_qp = max_qp;
streams[0].max_framerate = 5;
streams[0].min_bitrate_bps = kMinVideoBitrateKbps * 1000;
streams[0].target_bitrate_bps = config.tl0_bitrate_kbps * 1000;
streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
streams[0].temporal_layer_thresholds_bps.clear();
streams[0].temporal_layer_thresholds_bps.push_back(config.tl0_bitrate_kbps *
1000);
// With simulcast enabled, add another spatial layer. This one will have a
// more normal layout, with the regular 3 temporal layer pattern and no fps
// restrictions. The base simulcast stream will still use legacy setup.
if (num_simulcast_layers == kMaxScreenshareSimulcastStreams) {
// Add optional upper simulcast layer.
// Lowest temporal layers of a 3 layer setup will have 40% of the total
// bitrate allocation for that stream. Make sure the gap between the
// target of the lower stream and first temporal layer of the higher one
// is at most 2x the bitrate, so that upswitching is not hampered by
// stalled bitrate estimates.
int max_bitrate_bps = 2 * ((streams[0].target_bitrate_bps * 10) / 4);
// Cap max bitrate so it isn't overly high for the given resolution.
max_bitrate_bps = std::min<int>(
max_bitrate_bps, FindSimulcastMaxBitrateBps(width, height));
streams[1].width = width;
streams[1].height = height;
streams[1].max_qp = max_qp;
streams[1].max_framerate = max_framerate;
// Three temporal layers means two thresholds.
streams[1].temporal_layer_thresholds_bps.resize(2);
streams[1].min_bitrate_bps = streams[0].target_bitrate_bps * 2;
streams[1].target_bitrate_bps = max_bitrate_bps;
streams[1].max_bitrate_bps = max_bitrate_bps;
}
} else {
if (!is_screencast) {
// Format width and height has to be divisible by |2 ^ number_streams - 1|.
width = NormalizeSimulcastSize(width, num_simulcast_layers);
height = NormalizeSimulcastSize(height, num_simulcast_layers);
}
// Add simulcast sub-streams from lower resolution to higher resolutions.
// Add simulcast streams, from highest resolution (|s| = number_streams -1)
// to lowest resolution at |s| = 0.
for (size_t s = num_simulcast_layers - 1;; --s) {
streams[s].width = width;
streams[s].height = height;
// TODO(pbos): Fill actual temporal-layer bitrate thresholds.
streams[s].max_qp = max_qp;
// Add simulcast sub-streams from lower resolution to higher resolutions.
// Add simulcast streams, from highest resolution (|s| = number_streams -1)
// to lowest resolution at |s| = 0.
for (size_t s = num_simulcast_layers - 1;; --s) {
streams[s].width = width;
streams[s].height = height;
// TODO(pbos): Fill actual temporal-layer bitrate thresholds.
streams[s].max_qp = max_qp;
if (is_screencast && s == 0) {
ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
// For legacy screenshare in conference mode, tl0 and tl1 bitrates are
// piggybacked on the VideoCodec struct as target and max bitrates,
// respectively. See eg. webrtc::VP8EncoderImpl::SetRates().
streams[s].min_bitrate_bps = kMinVideoBitrateKbps * 1000;
streams[s].target_bitrate_bps = config.tl0_bitrate_kbps * 1000;
streams[s].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
streams[s].temporal_layer_thresholds_bps.clear();
streams[s].temporal_layer_thresholds_bps.push_back(
config.tl0_bitrate_kbps * 1000);
streams[s].max_framerate = 5;
} else {
streams[s].temporal_layer_thresholds_bps.resize(
kDefaultConferenceNumberOfTemporalLayers[s] - 1);
streams[s].max_bitrate_bps = FindSimulcastMaxBitrateBps(width, height);
@ -259,19 +228,20 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
FindSimulcastTargetBitrateBps(width, height);
streams[s].min_bitrate_bps = FindSimulcastMinBitrateBps(width, height);
streams[s].max_framerate = max_framerate;
}
if (!is_screencast) {
width /= 2;
height /= 2;
if (s == 0)
break;
}
if (s == 0)
break;
}
// Spend additional bits to boost the max stream.
int bitrate_left_bps = max_bitrate_bps - GetTotalMaxBitrateBps(streams);
if (bitrate_left_bps > 0) {
streams.back().max_bitrate_bps += bitrate_left_bps;
}
// Spend additional bits to boost the max stream.
int bitrate_left_bps = max_bitrate_bps - GetTotalMaxBitrateBps(streams);
if (bitrate_left_bps > 0) {
streams.back().max_bitrate_bps += bitrate_left_bps;
}
return streams;

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

@ -118,7 +118,13 @@ uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate) {
// Update encoding rates following protection settings.
float target_video_bitrate_kbps =
static_cast<float>(video_target_bitrate_) / 1000.0f;
frame_dropper_->SetRates(target_video_bitrate_kbps, incoming_frame_rate_);
float framerate = incoming_frame_rate_;
if (framerate == 0.0) {
// No framerate estimate available, use configured max framerate instead.
framerate = user_frame_rate_;
}
frame_dropper_->SetRates(target_video_bitrate_kbps, framerate);
return video_target_bitrate_;
}

27
webrtc/modules/video_coding/video_sender.cc
View File

@ -103,6 +103,11 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
numLayers = sendCodec->VP8().numberOfTemporalLayers;
} else if (sendCodec->codecType == kVideoCodecVP9) {
numLayers = sendCodec->VP9().numberOfTemporalLayers;
} else if (sendCodec->codecType == kVideoCodecGeneric &&
sendCodec->numberOfSimulcastStreams > 0) {
// This is mainly for unit testing, disabling frame dropping.
// TODO(sprang): Add a better way to disable frame dropping.
numLayers = sendCodec->simulcastStream[0].numberOfTemporalLayers;
} else {
numLayers = 1;
}
@ -197,13 +202,17 @@ EncoderParameters VideoSender::UpdateEncoderParameters(
input_frame_rate = current_codec_.maxFramerate;
BitrateAllocation bitrate_allocation;
if (bitrate_allocator) {
bitrate_allocation = bitrate_allocator->GetAllocation(video_target_rate_bps,
input_frame_rate);
} else {
DefaultVideoBitrateAllocator default_allocator(current_codec_);
bitrate_allocation = default_allocator.GetAllocation(video_target_rate_bps,
input_frame_rate);
// Only call allocators if bitrate > 0 (ie, not suspended), otherwise they
// might cap the bitrate to the min bitrate configured.
if (target_bitrate_bps > 0) {
if (bitrate_allocator) {
bitrate_allocation = bitrate_allocator->GetAllocation(
video_target_rate_bps, input_frame_rate);
} else {
DefaultVideoBitrateAllocator default_allocator(current_codec_);
bitrate_allocation = default_allocator.GetAllocation(
video_target_rate_bps, input_frame_rate);
}
}
EncoderParameters new_encoder_params = {bitrate_allocation, params.loss_rate,
params.rtt, input_frame_rate};
@ -221,7 +230,7 @@ void VideoSender::UpdateChannelParemeters(
encoder_params_.target_bitrate.get_sum_bps());
target_rate = encoder_params_.target_bitrate;
}
if (bitrate_updated_callback)
if (bitrate_updated_callback && target_rate.get_sum_bps() > 0)
bitrate_updated_callback->OnBitrateAllocationUpdated(target_rate);
}
@ -236,7 +245,7 @@ int32_t VideoSender::SetChannelParameters(
encoder_params.rtt = rtt;
encoder_params = UpdateEncoderParameters(encoder_params, bitrate_allocator,
target_bitrate_bps);
if (bitrate_updated_callback) {
if (bitrate_updated_callback && target_bitrate_bps > 0) {
bitrate_updated_callback->OnBitrateAllocationUpdated(
encoder_params.target_bitrate);
}

86
webrtc/test/fake_encoder.cc
View File

@ -24,11 +24,15 @@
namespace webrtc {
namespace test {
const int kKeyframeSizeFactor = 10;
FakeEncoder::FakeEncoder(Clock* clock)
: clock_(clock),
callback_(nullptr),
configured_input_framerate_(-1),
max_target_bitrate_kbps_(-1),
last_encode_time_ms_(0) {
pending_keyframe_(true),
debt_bytes_(0) {
// Generate some arbitrary not-all-zero data
for (size_t i = 0; i < sizeof(encoded_buffer_); ++i) {
encoded_buffer_[i] = static_cast<uint8_t>(i);
@ -47,6 +51,8 @@ int32_t FakeEncoder::InitEncode(const VideoCodec* config,
rtc::CritScope cs(&crit_sect_);
config_ = *config;
target_bitrate_.SetBitrate(0, 0, config_.startBitrate * 1000);
configured_input_framerate_ = config_.maxFramerate;
pending_keyframe_ = true;
return 0;
}
@ -59,9 +65,10 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
EncodedImageCallback* callback;
uint32_t target_bitrate_sum_kbps;
int max_target_bitrate_kbps;
int64_t last_encode_time_ms;
size_t num_encoded_bytes;
int framerate;
VideoCodecMode mode;
bool keyframe;
{
rtc::CritScope cs(&crit_sect_);
max_framerate = config_.maxFramerate;
@ -72,42 +79,32 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
callback = callback_;
target_bitrate_sum_kbps = target_bitrate_.get_sum_kbps();
max_target_bitrate_kbps = max_target_bitrate_kbps_;
last_encode_time_ms = last_encode_time_ms_;
num_encoded_bytes = sizeof(encoded_buffer_);
mode = config_.mode;
if (configured_input_framerate_ > 0) {
framerate = configured_input_framerate_;
} else {
framerate = max_framerate;
}
keyframe = pending_keyframe_;
pending_keyframe_ = false;
}
for (FrameType frame_type : *frame_types) {
if (frame_type == kVideoFrameKey) {
keyframe = true;
break;
}
}
int64_t time_now_ms = clock_->TimeInMilliseconds();
const bool first_encode = (last_encode_time_ms == 0);
RTC_DCHECK_GT(max_framerate, 0);
int64_t time_since_last_encode_ms = 1000 / max_framerate;
if (!first_encode) {
// For all frames but the first we can estimate the display time by looking
// at the display time of the previous frame.
time_since_last_encode_ms = time_now_ms - last_encode_time_ms;
}
if (time_since_last_encode_ms > 3 * 1000 / max_framerate) {
// Rudimentary check to make sure we don't widely overshoot bitrate target
// when resuming encoding after a suspension.
time_since_last_encode_ms = 3 * 1000 / max_framerate;
}
size_t bits_available =
static_cast<size_t>(target_bitrate_sum_kbps * time_since_last_encode_ms);
size_t min_bits = static_cast<size_t>(simulcast_streams[0].minBitrate *
time_since_last_encode_ms);
size_t bitrate =
std::max(target_bitrate_sum_kbps, simulcast_streams[0].minBitrate);
if (max_target_bitrate_kbps > 0)
bitrate = std::min(bitrate, static_cast<size_t>(max_target_bitrate_kbps));
if (bits_available < min_bits)
bits_available = min_bits;
size_t max_bits =
static_cast<size_t>(max_target_bitrate_kbps * time_since_last_encode_ms);
if (max_bits > 0 && max_bits < bits_available)
bits_available = max_bits;
{
rtc::CritScope cs(&crit_sect_);
last_encode_time_ms_ = time_now_ms;
}
size_t bits_available = bitrate * 1000 / framerate;
RTC_DCHECK_GT(num_simulcast_streams, 0);
for (unsigned char i = 0; i < num_simulcast_streams; ++i) {
@ -116,18 +113,27 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
specifics.codecType = kVideoCodecGeneric;
specifics.codecSpecific.generic.simulcast_idx = i;
size_t min_stream_bits = static_cast<size_t>(
simulcast_streams[i].minBitrate * time_since_last_encode_ms);
(simulcast_streams[i].minBitrate * 1000) / framerate);
size_t max_stream_bits = static_cast<size_t>(
simulcast_streams[i].maxBitrate * time_since_last_encode_ms);
(simulcast_streams[i].maxBitrate * 1000) / framerate);
size_t stream_bits = (bits_available > max_stream_bits) ? max_stream_bits :
bits_available;
size_t stream_bytes = (stream_bits + 7) / 8;
if (first_encode) {
if (keyframe) {
// The first frame is a key frame and should be larger.
// TODO(holmer): The FakeEncoder should store the bits_available between
// encodes so that it can compensate for oversized frames.
stream_bytes *= 10;
// Store the overshoot bytes and distribute them over the coming frames,
// so that we on average meet the bitrate target.
debt_bytes_ += (kKeyframeSizeFactor - 1) * stream_bytes;
stream_bytes *= kKeyframeSizeFactor;
} else {
if (debt_bytes_ > 0) {
// Pay at most half of the frame size for old debts.
size_t payment_size = std::min(stream_bytes / 2, debt_bytes_);
debt_bytes_ -= payment_size;
stream_bytes -= payment_size;
}
}
if (stream_bytes > num_encoded_bytes)
stream_bytes = num_encoded_bytes;
@ -176,6 +182,7 @@ int32_t FakeEncoder::SetRateAllocation(const BitrateAllocation& rate_allocation,
uint32_t framerate) {
rtc::CritScope cs(&crit_sect_);
target_bitrate_ = rate_allocation;
configured_input_framerate_ = framerate;
return 0;
}
@ -184,6 +191,11 @@ const char* FakeEncoder::ImplementationName() const {
return kImplementationName;
}
int FakeEncoder::GetConfiguredInputFramerate() const {
rtc::CritScope cs(&crit_sect_);
return configured_input_framerate_;
}
FakeH264Encoder::FakeH264Encoder(Clock* clock)
: FakeEncoder(clock), callback_(nullptr), idr_counter_(0) {
FakeEncoder::RegisterEncodeCompleteCallback(this);

8
webrtc/test/fake_encoder.h
View File

@ -45,6 +45,7 @@ class FakeEncoder : public VideoEncoder {
int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
uint32_t framerate) override;
const char* ImplementationName() const override;
int GetConfiguredInputFramerate() const;
static const char* kImplementationName;
@ -53,11 +54,16 @@ class FakeEncoder : public VideoEncoder {
VideoCodec config_ GUARDED_BY(crit_sect_);
EncodedImageCallback* callback_ GUARDED_BY(crit_sect_);
BitrateAllocation target_bitrate_ GUARDED_BY(crit_sect_);
int configured_input_framerate_ GUARDED_BY(crit_sect_);
int max_target_bitrate_kbps_ GUARDED_BY(crit_sect_);
int64_t last_encode_time_ms_ GUARDED_BY(crit_sect_);
bool pending_keyframe_ GUARDED_BY(crit_sect_);
rtc::CriticalSection crit_sect_;
uint8_t encoded_buffer_[100000];
// Current byte debt to be payed over a number of frames.
// The debt is acquired by keyframes overshooting the bitrate target.
size_t debt_bytes_;
};
class FakeH264Encoder : public FakeEncoder, public EncodedImageCallback {

37
webrtc/video/video_send_stream_tests.cc
View File

@ -975,10 +975,12 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
void TriggerLossReport(const RTPHeader& header) {
// Send lossy receive reports to trigger FEC enabling.
if (packet_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
const int kLossPercent = 5;
if (packet_count_++ % (100 / kLossPercent) != 0) {
FakeReceiveStatistics lossy_receive_stats(
kVideoSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
kVideoSendSsrcs[0], header.sequenceNumber,
(packet_count_ * (100 - kLossPercent)) / 100, // Cumulative lost.
static_cast<uint8_t>((255 * kLossPercent) / 100)); // Loss percent.
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr, nullptr,
transport_adapter_.get());
@ -1031,6 +1033,35 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
// Make sure there is at least one extension header, to make the RTP
// header larger than the base length of 12 bytes.
EXPECT_FALSE(send_config->rtp.extensions.empty());
// Setup screen content disables frame dropping which makes this easier.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
explicit VideoStreamFactory(size_t num_temporal_layers)
: num_temporal_layers_(num_temporal_layers) {
EXPECT_GT(num_temporal_layers, 0u);
}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
for (VideoStream& stream : streams) {
stream.temporal_layer_thresholds_bps.resize(num_temporal_layers_ -
1);
}
return streams;
}
const size_t num_temporal_layers_;
};
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>(2);
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
void PerformTest() override {

3
webrtc/video/vie_encoder.cc
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@ -778,13 +778,14 @@ void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
int64_t now_ms = clock_->TimeInMilliseconds();
if (pending_encoder_reconfiguration_) {
ReconfigureEncoder();
last_parameters_update_ms_.emplace(now_ms);
} else if (!last_parameters_update_ms_ ||
now_ms - *last_parameters_update_ms_ >=
vcm::VCMProcessTimer::kDefaultProcessIntervalMs) {
video_sender_.UpdateChannelParemeters(rate_allocator_.get(),
bitrate_observer_);
last_parameters_update_ms_.emplace(now_ms);
}
last_parameters_update_ms_.emplace(now_ms);
if (EncoderPaused()) {
TraceFrameDropStart();

490
webrtc/video/vie_encoder_unittest.cc
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