Nagram/TMessagesProj/jni/voip/webrtc/video/config/simulcast.cc
2023-02-19 01:24:25 +04:00

498 lines
20 KiB
C++

/*
* Copyright (c) 2014 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 "video/config/simulcast.h"
#include <stdint.h>
#include <stdio.h>
#include <algorithm>
#include <string>
#include <vector>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "media/base/media_constants.h"
#include "modules/video_coding/utility/simulcast_rate_allocator.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
#include "rtc_base/experiments/rate_control_settings.h"
#include "rtc_base/logging.h"
namespace cricket {
namespace {
constexpr char kUseLegacySimulcastLayerLimitFieldTrial[] =
"WebRTC-LegacySimulcastLayerLimit";
constexpr double kDefaultMaxRoundupRate = 0.1;
// Limits for legacy conference screensharing mode. Currently used for the
// lower of the two simulcast streams.
constexpr webrtc::DataRate kScreenshareDefaultTl0Bitrate =
webrtc::DataRate::KilobitsPerSec(200);
constexpr webrtc::DataRate kScreenshareDefaultTl1Bitrate =
webrtc::DataRate::KilobitsPerSec(1000);
// Min/max bitrate for the higher one of the two simulcast stream used for
// screen content.
constexpr webrtc::DataRate kScreenshareHighStreamMinBitrate =
webrtc::DataRate::KilobitsPerSec(600);
constexpr webrtc::DataRate kScreenshareHighStreamMaxBitrate =
webrtc::DataRate::KilobitsPerSec(1250);
constexpr int kDefaultNumTemporalLayers = 3;
constexpr int kScreenshareMaxSimulcastLayers = 2;
constexpr int kScreenshareTemporalLayers = 2;
struct SimulcastFormat {
int width;
int height;
// The maximum number of simulcast layers can be used for
// resolutions at `widthxheight` for legacy applications.
size_t max_layers;
// The maximum bitrate for encoding stream at `widthxheight`, when we are
// not sending the next higher spatial stream.
webrtc::DataRate max_bitrate;
// The target bitrate for encoding stream at `widthxheight`, when this layer
// is not the highest layer (i.e., when we are sending another higher spatial
// stream).
webrtc::DataRate target_bitrate;
// The minimum bitrate needed for encoding stream at `widthxheight`.
webrtc::DataRate min_bitrate;
};
// These tables describe from which resolution we can use how many
// simulcast layers at what bitrates (maximum, target, and minimum).
// Important!! Keep this table from high resolution to low resolution.
constexpr const SimulcastFormat kSimulcastFormats[] = {
{1920, 1080, 3, webrtc::DataRate::KilobitsPerSec(5000),
webrtc::DataRate::KilobitsPerSec(4000),
webrtc::DataRate::KilobitsPerSec(800)},
{1280, 720, 3, webrtc::DataRate::KilobitsPerSec(2500),
webrtc::DataRate::KilobitsPerSec(2500),
webrtc::DataRate::KilobitsPerSec(600)},
{960, 540, 3, webrtc::DataRate::KilobitsPerSec(1200),
webrtc::DataRate::KilobitsPerSec(1200),
webrtc::DataRate::KilobitsPerSec(350)},
{640, 360, 2, webrtc::DataRate::KilobitsPerSec(700),
webrtc::DataRate::KilobitsPerSec(500),
webrtc::DataRate::KilobitsPerSec(150)},
{480, 270, 2, webrtc::DataRate::KilobitsPerSec(450),
webrtc::DataRate::KilobitsPerSec(350),
webrtc::DataRate::KilobitsPerSec(150)},
{320, 180, 1, webrtc::DataRate::KilobitsPerSec(200),
webrtc::DataRate::KilobitsPerSec(150),
webrtc::DataRate::KilobitsPerSec(30)},
// As the resolution goes down, interpolate the target and max bitrates down
// towards zero. The min bitrate is still limited at 30 kbps and the target
// and the max will be capped from below accordingly.
{0, 0, 1, webrtc::DataRate::KilobitsPerSec(0),
webrtc::DataRate::KilobitsPerSec(0),
webrtc::DataRate::KilobitsPerSec(30)}};
constexpr webrtc::DataRate Interpolate(const webrtc::DataRate& a,
const webrtc::DataRate& b,
float rate) {
return a * (1.0 - rate) + b * rate;
}
// TODO(webrtc:12415): Flip this to a kill switch when this feature launches.
bool EnableLowresBitrateInterpolation(const webrtc::FieldTrialsView& trials) {
return absl::StartsWith(
trials.Lookup("WebRTC-LowresSimulcastBitrateInterpolation"), "Enabled");
}
std::vector<SimulcastFormat> GetSimulcastFormats(
bool enable_lowres_bitrate_interpolation) {
std::vector<SimulcastFormat> formats;
formats.insert(formats.begin(), std::begin(kSimulcastFormats),
std::end(kSimulcastFormats));
if (!enable_lowres_bitrate_interpolation) {
RTC_CHECK_GE(formats.size(), 2u);
SimulcastFormat& format0x0 = formats[formats.size() - 1];
const SimulcastFormat& format_prev = formats[formats.size() - 2];
format0x0.max_bitrate = format_prev.max_bitrate;
format0x0.target_bitrate = format_prev.target_bitrate;
format0x0.min_bitrate = format_prev.min_bitrate;
}
return formats;
}
// Multiway: Number of temporal layers for each simulcast stream.
int DefaultNumberOfTemporalLayers(const webrtc::FieldTrialsView& trials) {
const std::string group_name =
trials.Lookup("WebRTC-VP8ConferenceTemporalLayers");
if (group_name.empty())
return kDefaultNumTemporalLayers;
int num_temporal_layers = kDefaultNumTemporalLayers;
if (sscanf(group_name.c_str(), "%d", &num_temporal_layers) == 1 &&
num_temporal_layers > 0 &&
num_temporal_layers <= webrtc::kMaxTemporalStreams) {
return num_temporal_layers;
}
RTC_LOG(LS_WARNING) << "Attempt to set number of temporal layers to "
"incorrect value: "
<< group_name;
return kDefaultNumTemporalLayers;
}
int FindSimulcastFormatIndex(int width,
int height,
bool enable_lowres_bitrate_interpolation) {
RTC_DCHECK_GE(width, 0);
RTC_DCHECK_GE(height, 0);
const auto formats = GetSimulcastFormats(enable_lowres_bitrate_interpolation);
for (uint32_t i = 0; i < formats.size(); ++i) {
if (width * height >= formats[i].width * formats[i].height) {
return i;
}
}
RTC_DCHECK_NOTREACHED();
return -1;
}
} // namespace
// Round size to nearest simulcast-friendly size.
// Simulcast stream width and height must both be dividable by
// |2 ^ (simulcast_layers - 1)|.
int NormalizeSimulcastSize(int size, size_t simulcast_layers) {
int base2_exponent = static_cast<int>(simulcast_layers) - 1;
const absl::optional<int> experimental_base2_exponent =
webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent();
if (experimental_base2_exponent &&
(size > (1 << *experimental_base2_exponent))) {
base2_exponent = *experimental_base2_exponent;
}
return ((size >> base2_exponent) << base2_exponent);
}
SimulcastFormat InterpolateSimulcastFormat(
int width,
int height,
absl::optional<double> max_roundup_rate,
bool enable_lowres_bitrate_interpolation) {
const auto formats = GetSimulcastFormats(enable_lowres_bitrate_interpolation);
const int index = FindSimulcastFormatIndex(
width, height, enable_lowres_bitrate_interpolation);
if (index == 0)
return formats[index];
const int total_pixels_up =
formats[index - 1].width * formats[index - 1].height;
const int total_pixels_down = formats[index].width * formats[index].height;
const int total_pixels = width * height;
const float rate = (total_pixels_up - total_pixels) /
static_cast<float>(total_pixels_up - total_pixels_down);
// Use upper resolution if `rate` is below the configured threshold.
size_t max_layers = (rate < max_roundup_rate.value_or(kDefaultMaxRoundupRate))
? formats[index - 1].max_layers
: formats[index].max_layers;
webrtc::DataRate max_bitrate = Interpolate(formats[index - 1].max_bitrate,
formats[index].max_bitrate, rate);
webrtc::DataRate target_bitrate = Interpolate(
formats[index - 1].target_bitrate, formats[index].target_bitrate, rate);
webrtc::DataRate min_bitrate = Interpolate(formats[index - 1].min_bitrate,
formats[index].min_bitrate, rate);
return {width, height, max_layers, max_bitrate, target_bitrate, min_bitrate};
}
SimulcastFormat InterpolateSimulcastFormat(
int width,
int height,
bool enable_lowres_bitrate_interpolation) {
return InterpolateSimulcastFormat(width, height, absl::nullopt,
enable_lowres_bitrate_interpolation);
}
webrtc::DataRate FindSimulcastMaxBitrate(
int width,
int height,
bool enable_lowres_bitrate_interpolation) {
return InterpolateSimulcastFormat(width, height,
enable_lowres_bitrate_interpolation)
.max_bitrate;
}
webrtc::DataRate FindSimulcastTargetBitrate(
int width,
int height,
bool enable_lowres_bitrate_interpolation) {
return InterpolateSimulcastFormat(width, height,
enable_lowres_bitrate_interpolation)
.target_bitrate;
}
webrtc::DataRate FindSimulcastMinBitrate(
int width,
int height,
bool enable_lowres_bitrate_interpolation) {
return InterpolateSimulcastFormat(width, height,
enable_lowres_bitrate_interpolation)
.min_bitrate;
}
void BoostMaxSimulcastLayer(webrtc::DataRate max_bitrate,
std::vector<webrtc::VideoStream>* layers) {
if (layers->empty())
return;
const webrtc::DataRate total_bitrate = GetTotalMaxBitrate(*layers);
// We're still not using all available bits.
if (total_bitrate < max_bitrate) {
// Spend additional bits to boost the max layer.
const webrtc::DataRate bitrate_left = max_bitrate - total_bitrate;
layers->back().max_bitrate_bps += bitrate_left.bps();
}
}
webrtc::DataRate GetTotalMaxBitrate(
const std::vector<webrtc::VideoStream>& layers) {
if (layers.empty())
return webrtc::DataRate::Zero();
int total_max_bitrate_bps = 0;
for (size_t s = 0; s < layers.size() - 1; ++s) {
total_max_bitrate_bps += layers[s].target_bitrate_bps;
}
total_max_bitrate_bps += layers.back().max_bitrate_bps;
return webrtc::DataRate::BitsPerSec(total_max_bitrate_bps);
}
size_t LimitSimulcastLayerCount(int width,
int height,
size_t need_layers,
size_t layer_count,
const webrtc::FieldTrialsView& trials) {
if (!absl::StartsWith(trials.Lookup(kUseLegacySimulcastLayerLimitFieldTrial),
"Disabled")) {
// Max layers from one higher resolution in kSimulcastFormats will be used
// if the ratio (pixels_up - pixels) / (pixels_up - pixels_down) is less
// than configured `max_ratio`. pixels_down is the selected index in
// kSimulcastFormats based on pixels.
webrtc::FieldTrialOptional<double> max_ratio("max_ratio");
webrtc::ParseFieldTrial({&max_ratio},
trials.Lookup("WebRTC-SimulcastLayerLimitRoundUp"));
const bool enable_lowres_bitrate_interpolation =
EnableLowresBitrateInterpolation(trials);
size_t adaptive_layer_count = std::max(
need_layers,
InterpolateSimulcastFormat(width, height, max_ratio.GetOptional(),
enable_lowres_bitrate_interpolation)
.max_layers);
if (layer_count > adaptive_layer_count) {
RTC_LOG(LS_WARNING) << "Reducing simulcast layer count from "
<< layer_count << " to " << adaptive_layer_count;
layer_count = adaptive_layer_count;
}
}
return layer_count;
}
std::vector<webrtc::VideoStream> GetSimulcastConfig(
size_t min_layers,
size_t max_layers,
int width,
int height,
double bitrate_priority,
int max_qp,
bool is_screenshare_with_conference_mode,
bool temporal_layers_supported,
const webrtc::FieldTrialsView& trials) {
RTC_DCHECK_LE(min_layers, max_layers);
RTC_DCHECK(max_layers > 1 || is_screenshare_with_conference_mode);
const bool base_heavy_tl3_rate_alloc =
webrtc::RateControlSettings::ParseFromKeyValueConfig(&trials)
.Vp8BaseHeavyTl3RateAllocation();
if (is_screenshare_with_conference_mode) {
return GetScreenshareLayers(max_layers, width, height, bitrate_priority,
max_qp, temporal_layers_supported,
base_heavy_tl3_rate_alloc, trials);
} else {
// Some applications rely on the old behavior limiting the simulcast layer
// count based on the resolution automatically, which they can get through
// the WebRTC-LegacySimulcastLayerLimit field trial until they update.
max_layers =
LimitSimulcastLayerCount(width, height, min_layers, max_layers, trials);
return GetNormalSimulcastLayers(max_layers, width, height, bitrate_priority,
max_qp, temporal_layers_supported,
base_heavy_tl3_rate_alloc, trials);
}
}
std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
size_t layer_count,
int width,
int height,
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::FieldTrialsView& trials) {
std::vector<webrtc::VideoStream> layers(layer_count);
const bool enable_lowres_bitrate_interpolation =
EnableLowresBitrateInterpolation(trials);
// Format width and height has to be divisible by |2 ^ num_simulcast_layers -
// 1|.
width = NormalizeSimulcastSize(width, layer_count);
height = NormalizeSimulcastSize(height, layer_count);
// Add simulcast streams, from highest resolution (`s` = num_simulcast_layers
// -1) to lowest resolution at `s` = 0.
for (size_t s = layer_count - 1;; --s) {
layers[s].width = width;
layers[s].height = height;
// TODO(pbos): Fill actual temporal-layer bitrate thresholds.
layers[s].max_qp = max_qp;
layers[s].num_temporal_layers =
temporal_layers_supported ? DefaultNumberOfTemporalLayers(trials) : 1;
layers[s].max_bitrate_bps =
FindSimulcastMaxBitrate(width, height,
enable_lowres_bitrate_interpolation)
.bps();
layers[s].target_bitrate_bps =
FindSimulcastTargetBitrate(width, height,
enable_lowres_bitrate_interpolation)
.bps();
int num_temporal_layers = DefaultNumberOfTemporalLayers(trials);
if (s == 0) {
// If alternative temporal rate allocation is selected, adjust the
// bitrate of the lowest simulcast stream so that absolute bitrate for
// the base temporal layer matches the bitrate for the base temporal
// layer with the default 3 simulcast streams. Otherwise we risk a
// higher threshold for receiving a feed at all.
float rate_factor = 1.0;
if (num_temporal_layers == 3) {
if (base_heavy_tl3_rate_alloc) {
// Base heavy allocation increases TL0 bitrate from 40% to 60%.
rate_factor = 0.4 / 0.6;
}
} else {
rate_factor =
webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
3, 0, /*base_heavy_tl3_rate_alloc=*/false) /
webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
num_temporal_layers, 0, /*base_heavy_tl3_rate_alloc=*/false);
}
layers[s].max_bitrate_bps =
static_cast<int>(layers[s].max_bitrate_bps * rate_factor);
layers[s].target_bitrate_bps =
static_cast<int>(layers[s].target_bitrate_bps * rate_factor);
}
layers[s].min_bitrate_bps =
FindSimulcastMinBitrate(width, height,
enable_lowres_bitrate_interpolation)
.bps();
// Ensure consistency.
layers[s].max_bitrate_bps =
std::max(layers[s].min_bitrate_bps, layers[s].max_bitrate_bps);
layers[s].target_bitrate_bps =
std::max(layers[s].min_bitrate_bps, layers[s].target_bitrate_bps);
layers[s].max_framerate = kDefaultVideoMaxFramerate;
width /= 2;
height /= 2;
if (s == 0) {
break;
}
}
// Currently the relative bitrate priority of the sender is controlled by
// the value of the lowest VideoStream.
// TODO(bugs.webrtc.org/8630): The web specification describes being able to
// control relative bitrate for each individual simulcast layer, but this
// is currently just implemented per rtp sender.
layers[0].bitrate_priority = bitrate_priority;
return layers;
}
std::vector<webrtc::VideoStream> GetScreenshareLayers(
size_t max_layers,
int width,
int height,
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::FieldTrialsView& trials) {
size_t num_simulcast_layers =
std::min<int>(max_layers, kScreenshareMaxSimulcastLayers);
std::vector<webrtc::VideoStream> layers(num_simulcast_layers);
// 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::LibvpxVp8Encoder::SetRates().
layers[0].width = width;
layers[0].height = height;
layers[0].max_qp = max_qp;
layers[0].max_framerate = 5;
layers[0].min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps;
layers[0].target_bitrate_bps = kScreenshareDefaultTl0Bitrate.bps();
layers[0].max_bitrate_bps = kScreenshareDefaultTl1Bitrate.bps();
layers[0].num_temporal_layers = temporal_layers_supported ? 2 : 1;
// 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 layer will still use legacy setup.
if (num_simulcast_layers == kScreenshareMaxSimulcastLayers) {
// Add optional upper simulcast layer.
int max_bitrate_bps;
bool using_boosted_bitrate = false;
if (!temporal_layers_supported) {
// Set the max bitrate to where the base layer would have been if temporal
// layers were enabled.
max_bitrate_bps = static_cast<int>(
kScreenshareHighStreamMaxBitrate.bps() *
webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
kScreenshareTemporalLayers, 0, base_heavy_tl3_rate_alloc));
} else {
// Experimental temporal layer mode used, use increased max bitrate.
max_bitrate_bps = kScreenshareHighStreamMaxBitrate.bps();
using_boosted_bitrate = true;
}
layers[1].width = width;
layers[1].height = height;
layers[1].max_qp = max_qp;
layers[1].max_framerate = kDefaultVideoMaxFramerate;
layers[1].num_temporal_layers =
temporal_layers_supported ? kScreenshareTemporalLayers : 1;
layers[1].min_bitrate_bps = using_boosted_bitrate
? kScreenshareHighStreamMinBitrate.bps()
: layers[0].target_bitrate_bps * 2;
layers[1].target_bitrate_bps = max_bitrate_bps;
layers[1].max_bitrate_bps = max_bitrate_bps;
}
// The bitrate priority currently implemented on a per-sender level, so we
// just set it for the first simulcast layer.
layers[0].bitrate_priority = bitrate_priority;
return layers;
}
} // namespace cricket