/* * 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 "media/engine/simulcast.h" #include #include #include #include #include #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 webrtc::DataRate Interpolate(const webrtc::DataRate& a, const webrtc::DataRate& b, float rate) { return a * (1.0 - rate) + b * rate; } constexpr char kUseLegacySimulcastLayerLimitFieldTrial[] = "WebRTC-LegacySimulcastLayerLimit"; constexpr double kDefaultMaxRoundupRate = 0.1; // TODO(webrtc:12415): Flip this to a kill switch when this feature launches. bool EnableLowresBitrateInterpolation( const webrtc::WebRtcKeyValueConfig& trials) { return absl::StartsWith( trials.Lookup("WebRTC-LowresSimulcastBitrateInterpolation"), "Enabled"); } // 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); } // namespace 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)}}; std::vector GetSimulcastFormats( bool enable_lowres_bitrate_interpolation) { std::vector 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; } const int kMaxScreenshareSimulcastLayers = 2; // Multiway: Number of temporal layers for each simulcast stream. int DefaultNumberOfTemporalLayers(int simulcast_id, bool screenshare, const webrtc::WebRtcKeyValueConfig& trials) { RTC_CHECK_GE(simulcast_id, 0); RTC_CHECK_LT(simulcast_id, webrtc::kMaxSimulcastStreams); const int kDefaultNumTemporalLayers = 3; const int kDefaultNumScreenshareTemporalLayers = 2; int default_num_temporal_layers = screenshare ? kDefaultNumScreenshareTemporalLayers : kDefaultNumTemporalLayers; const std::string group_name = screenshare ? trials.Lookup("WebRTC-VP8ScreenshareTemporalLayers") : trials.Lookup("WebRTC-VP8ConferenceTemporalLayers"); if (group_name.empty()) return default_num_temporal_layers; int num_temporal_layers = default_num_temporal_layers; 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 default_num_temporal_layers; } 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_NOTREACHED(); return -1; } // 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(simulcast_layers) - 1; const absl::optional 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 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(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* 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& 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::WebRtcKeyValueConfig& 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 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 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::WebRtcKeyValueConfig& 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 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::WebRtcKeyValueConfig& trials) { std::vector 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(s, false, 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(s, false, 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(layers[s].max_bitrate_bps * rate_factor); layers[s].target_bitrate_bps = static_cast(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 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::WebRtcKeyValueConfig& trials) { auto max_screenshare_layers = kMaxScreenshareSimulcastLayers; size_t num_simulcast_layers = std::min(max_layers, max_screenshare_layers); std::vector 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 == kMaxScreenshareSimulcastLayers) { // Add optional upper simulcast layer. const int num_temporal_layers = DefaultNumberOfTemporalLayers(1, true, trials); 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( kScreenshareHighStreamMaxBitrate.bps() * webrtc::SimulcastRateAllocator::GetTemporalRateAllocation( num_temporal_layers, 0, base_heavy_tl3_rate_alloc)); } else if (DefaultNumberOfTemporalLayers(1, true, trials) != 3 || base_heavy_tl3_rate_alloc) { // Experimental temporal layer mode used, use increased max bitrate. max_bitrate_bps = kScreenshareHighStreamMaxBitrate.bps(); using_boosted_bitrate = true; } else { // Keep current bitrates with default 3tl/8 frame settings. // Lowest temporal layers of a 3 layer setup will have 40% of the total // bitrate allocation for that simulcast layer. Make sure the gap between // the target of the lower simulcast layer and first temporal layer of the // higher one is at most 2x the bitrate, so that upswitching is not // hampered by stalled bitrate estimates. max_bitrate_bps = 2 * ((layers[0].target_bitrate_bps * 10) / 4); } 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 ? DefaultNumberOfTemporalLayers(1, true, trials) : 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