670 lines
26 KiB
C++
670 lines
26 KiB
C++
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/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "video/video_send_stream_impl.h"
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#include <stdio.h>
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#include <algorithm>
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#include <cstdint>
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#include <string>
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "api/crypto/crypto_options.h"
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#include "api/rtp_parameters.h"
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#include "api/scoped_refptr.h"
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#include "api/video_codecs/video_codec.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "call/video_send_stream.h"
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#include "modules/pacing/paced_sender.h"
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#include "rtc_base/atomic_ops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/experiments/alr_experiment.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/experiments/min_video_bitrate_experiment.h"
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#include "rtc_base/experiments/rate_control_settings.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/synchronization/sequence_checker.h"
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#include "rtc_base/thread_checker.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace internal {
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namespace {
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// Max positive size difference to treat allocations as "similar".
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static constexpr int kMaxVbaSizeDifferencePercent = 10;
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// Max time we will throttle similar video bitrate allocations.
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static constexpr int64_t kMaxVbaThrottleTimeMs = 500;
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constexpr TimeDelta kEncoderTimeOut = TimeDelta::Seconds(2);
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bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
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const std::vector<RtpExtension>& extensions = config.rtp.extensions;
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return absl::c_any_of(extensions, [](const RtpExtension& ext) {
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return ext.uri == RtpExtension::kTransportSequenceNumberUri;
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});
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}
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// Calculate max padding bitrate for a multi layer codec.
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int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams,
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bool is_svc,
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VideoEncoderConfig::ContentType content_type,
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int min_transmit_bitrate_bps,
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bool pad_to_min_bitrate,
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bool alr_probing) {
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int pad_up_to_bitrate_bps = 0;
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RTC_DCHECK(!is_svc || streams.size() <= 1) << "Only one stream is allowed in "
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"SVC mode.";
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// Filter out only the active streams;
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std::vector<VideoStream> active_streams;
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for (const VideoStream& stream : streams) {
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if (stream.active)
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active_streams.emplace_back(stream);
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}
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if (active_streams.size() > 1 || (!active_streams.empty() && is_svc)) {
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// Simulcast or SVC is used.
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// if SVC is used, stream bitrates should already encode svc bitrates:
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// min_bitrate = min bitrate of a lowest svc layer.
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// target_bitrate = sum of target bitrates of lower layers + min bitrate
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// of the last one (as used in the calculations below).
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// max_bitrate = sum of all active layers' max_bitrate.
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if (alr_probing) {
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// With alr probing, just pad to the min bitrate of the lowest stream,
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// probing will handle the rest of the rampup.
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pad_up_to_bitrate_bps = active_streams[0].min_bitrate_bps;
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} else {
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// Without alr probing, pad up to start bitrate of the
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// highest active stream.
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const double hysteresis_factor =
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RateControlSettings::ParseFromFieldTrials()
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.GetSimulcastHysteresisFactor(content_type);
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if (is_svc) {
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// For SVC, since there is only one "stream", the padding bitrate
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// needed to enable the top spatial layer is stored in the
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// |target_bitrate_bps| field.
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// TODO(sprang): This behavior needs to die.
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pad_up_to_bitrate_bps = static_cast<int>(
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hysteresis_factor * active_streams[0].target_bitrate_bps + 0.5);
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} else {
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const size_t top_active_stream_idx = active_streams.size() - 1;
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pad_up_to_bitrate_bps = std::min(
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static_cast<int>(
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hysteresis_factor *
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active_streams[top_active_stream_idx].min_bitrate_bps +
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0.5),
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active_streams[top_active_stream_idx].target_bitrate_bps);
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// Add target_bitrate_bps of the lower active streams.
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for (size_t i = 0; i < top_active_stream_idx; ++i) {
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pad_up_to_bitrate_bps += active_streams[i].target_bitrate_bps;
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}
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}
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}
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} else if (!active_streams.empty() && pad_to_min_bitrate) {
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pad_up_to_bitrate_bps = active_streams[0].min_bitrate_bps;
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}
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pad_up_to_bitrate_bps =
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std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps);
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return pad_up_to_bitrate_bps;
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}
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RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
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const VideoSendStream::Config* config) {
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RtpSenderFrameEncryptionConfig frame_encryption_config;
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frame_encryption_config.frame_encryptor = config->frame_encryptor;
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frame_encryption_config.crypto_options = config->crypto_options;
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return frame_encryption_config;
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}
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RtpSenderObservers CreateObservers(RtcpRttStats* call_stats,
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EncoderRtcpFeedback* encoder_feedback,
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SendStatisticsProxy* stats_proxy,
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SendDelayStats* send_delay_stats) {
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RtpSenderObservers observers;
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observers.rtcp_rtt_stats = call_stats;
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observers.intra_frame_callback = encoder_feedback;
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observers.rtcp_loss_notification_observer = encoder_feedback;
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observers.rtcp_stats = stats_proxy;
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observers.report_block_data_observer = stats_proxy;
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observers.rtp_stats = stats_proxy;
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observers.bitrate_observer = stats_proxy;
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observers.frame_count_observer = stats_proxy;
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observers.rtcp_type_observer = stats_proxy;
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observers.send_delay_observer = stats_proxy;
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observers.send_packet_observer = send_delay_stats;
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return observers;
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}
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absl::optional<AlrExperimentSettings> GetAlrSettings(
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VideoEncoderConfig::ContentType content_type) {
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if (content_type == VideoEncoderConfig::ContentType::kScreen) {
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return AlrExperimentSettings::CreateFromFieldTrial(
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AlrExperimentSettings::kScreenshareProbingBweExperimentName);
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}
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return AlrExperimentSettings::CreateFromFieldTrial(
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AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
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}
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bool SameStreamsEnabled(const VideoBitrateAllocation& lhs,
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const VideoBitrateAllocation& rhs) {
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for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
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for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
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if (lhs.HasBitrate(si, ti) != rhs.HasBitrate(si, ti)) {
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return false;
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}
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}
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}
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return true;
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}
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} // namespace
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PacingConfig::PacingConfig()
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: pacing_factor("factor", PacedSender::kDefaultPaceMultiplier),
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max_pacing_delay("max_delay",
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TimeDelta::Millis(PacedSender::kMaxQueueLengthMs)) {
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ParseFieldTrial({&pacing_factor, &max_pacing_delay},
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field_trial::FindFullName("WebRTC-Video-Pacing"));
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}
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PacingConfig::PacingConfig(const PacingConfig&) = default;
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PacingConfig::~PacingConfig() = default;
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VideoSendStreamImpl::VideoSendStreamImpl(
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Clock* clock,
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SendStatisticsProxy* stats_proxy,
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rtc::TaskQueue* worker_queue,
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RtcpRttStats* call_stats,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocatorInterface* bitrate_allocator,
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SendDelayStats* send_delay_stats,
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VideoStreamEncoderInterface* video_stream_encoder,
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RtcEventLog* event_log,
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const VideoSendStream::Config* config,
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int initial_encoder_max_bitrate,
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double initial_encoder_bitrate_priority,
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std::map<uint32_t, RtpState> suspended_ssrcs,
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std::map<uint32_t, RtpPayloadState> suspended_payload_states,
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VideoEncoderConfig::ContentType content_type,
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std::unique_ptr<FecController> fec_controller)
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: clock_(clock),
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has_alr_probing_(config->periodic_alr_bandwidth_probing ||
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GetAlrSettings(content_type)),
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pacing_config_(PacingConfig()),
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stats_proxy_(stats_proxy),
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config_(config),
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worker_queue_(worker_queue),
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timed_out_(false),
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transport_(transport),
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bitrate_allocator_(bitrate_allocator),
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disable_padding_(true),
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max_padding_bitrate_(0),
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encoder_min_bitrate_bps_(0),
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encoder_target_rate_bps_(0),
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encoder_bitrate_priority_(initial_encoder_bitrate_priority),
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has_packet_feedback_(false),
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video_stream_encoder_(video_stream_encoder),
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encoder_feedback_(clock, config_->rtp.ssrcs, video_stream_encoder),
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bandwidth_observer_(transport->GetBandwidthObserver()),
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rtp_video_sender_(
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transport_->CreateRtpVideoSender(suspended_ssrcs,
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suspended_payload_states,
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config_->rtp,
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config_->rtcp_report_interval_ms,
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config_->send_transport,
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CreateObservers(call_stats,
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&encoder_feedback_,
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stats_proxy_,
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send_delay_stats),
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event_log,
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std::move(fec_controller),
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CreateFrameEncryptionConfig(config_),
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config->frame_transformer)),
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weak_ptr_factory_(this) {
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video_stream_encoder->SetFecControllerOverride(rtp_video_sender_);
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RTC_DCHECK_RUN_ON(worker_queue_);
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RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
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weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
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encoder_feedback_.SetRtpVideoSender(rtp_video_sender_);
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RTC_DCHECK(!config_->rtp.ssrcs.empty());
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RTC_DCHECK(transport_);
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RTC_DCHECK_NE(initial_encoder_max_bitrate, 0);
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if (initial_encoder_max_bitrate > 0) {
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encoder_max_bitrate_bps_ =
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rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
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} else {
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// TODO(srte): Make sure max bitrate is not set to negative values. We don't
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// have any way to handle unset values in downstream code, such as the
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// bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
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// behaviour that is not safe. Converting to 10 Mbps should be safe for
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// reasonable use cases as it allows adding the max of multiple streams
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// without wrappping around.
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const int kFallbackMaxBitrateBps = 10000000;
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RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
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<< initial_encoder_max_bitrate << " which is <= 0!";
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RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
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encoder_max_bitrate_bps_ = kFallbackMaxBitrateBps;
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}
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RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
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// If send-side BWE is enabled, check if we should apply updated probing and
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// pacing settings.
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if (TransportSeqNumExtensionConfigured(*config_)) {
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has_packet_feedback_ = true;
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absl::optional<AlrExperimentSettings> alr_settings =
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GetAlrSettings(content_type);
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if (alr_settings) {
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transport->EnablePeriodicAlrProbing(true);
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transport->SetPacingFactor(alr_settings->pacing_factor);
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configured_pacing_factor_ = alr_settings->pacing_factor;
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transport->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
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} else {
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RateControlSettings rate_control_settings =
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RateControlSettings::ParseFromFieldTrials();
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transport->EnablePeriodicAlrProbing(
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rate_control_settings.UseAlrProbing());
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const double pacing_factor =
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rate_control_settings.GetPacingFactor().value_or(
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pacing_config_.pacing_factor);
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transport->SetPacingFactor(pacing_factor);
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configured_pacing_factor_ = pacing_factor;
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transport->SetQueueTimeLimit(pacing_config_.max_pacing_delay.Get().ms());
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}
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}
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if (config_->periodic_alr_bandwidth_probing) {
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transport->EnablePeriodicAlrProbing(true);
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}
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RTC_DCHECK_GE(config_->rtp.payload_type, 0);
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RTC_DCHECK_LE(config_->rtp.payload_type, 127);
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video_stream_encoder_->SetStartBitrate(
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bitrate_allocator_->GetStartBitrate(this));
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// Only request rotation at the source when we positively know that the remote
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// side doesn't support the rotation extension. This allows us to prepare the
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// encoder in the expectation that rotation is supported - which is the common
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// case.
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bool rotation_applied = absl::c_none_of(
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config_->rtp.extensions, [](const RtpExtension& extension) {
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return extension.uri == RtpExtension::kVideoRotationUri;
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});
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video_stream_encoder_->SetSink(this, rotation_applied);
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}
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VideoSendStreamImpl::~VideoSendStreamImpl() {
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RTC_DCHECK_RUN_ON(worker_queue_);
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RTC_DCHECK(!rtp_video_sender_->IsActive())
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<< "VideoSendStreamImpl::Stop not called";
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RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
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transport_->DestroyRtpVideoSender(rtp_video_sender_);
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}
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void VideoSendStreamImpl::RegisterProcessThread(
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ProcessThread* module_process_thread) {
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rtp_video_sender_->RegisterProcessThread(module_process_thread);
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}
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void VideoSendStreamImpl::DeRegisterProcessThread() {
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rtp_video_sender_->DeRegisterProcessThread();
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}
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void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
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// Runs on a network thread.
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RTC_DCHECK(!worker_queue_->IsCurrent());
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rtp_video_sender_->DeliverRtcp(packet, length);
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}
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void VideoSendStreamImpl::UpdateActiveSimulcastLayers(
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const std::vector<bool> active_layers) {
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RTC_DCHECK_RUN_ON(worker_queue_);
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bool previously_active = rtp_video_sender_->IsActive();
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rtp_video_sender_->SetActiveModules(active_layers);
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if (!rtp_video_sender_->IsActive() && previously_active) {
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// Payload router switched from active to inactive.
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StopVideoSendStream();
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} else if (rtp_video_sender_->IsActive() && !previously_active) {
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// Payload router switched from inactive to active.
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StartupVideoSendStream();
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}
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}
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void VideoSendStreamImpl::Start() {
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RTC_DCHECK_RUN_ON(worker_queue_);
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RTC_LOG(LS_INFO) << "VideoSendStream::Start";
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if (rtp_video_sender_->IsActive())
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return;
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TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
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rtp_video_sender_->SetActive(true);
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StartupVideoSendStream();
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}
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void VideoSendStreamImpl::StartupVideoSendStream() {
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RTC_DCHECK_RUN_ON(worker_queue_);
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bitrate_allocator_->AddObserver(this, GetAllocationConfig());
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// Start monitoring encoder activity.
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{
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RTC_DCHECK(!check_encoder_activity_task_.Running());
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activity_ = false;
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timed_out_ = false;
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check_encoder_activity_task_ = RepeatingTaskHandle::DelayedStart(
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worker_queue_->Get(), kEncoderTimeOut, [this] {
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RTC_DCHECK_RUN_ON(worker_queue_);
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if (!activity_) {
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if (!timed_out_) {
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SignalEncoderTimedOut();
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}
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timed_out_ = true;
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disable_padding_ = true;
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} else if (timed_out_) {
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SignalEncoderActive();
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timed_out_ = false;
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}
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activity_ = false;
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return kEncoderTimeOut;
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|
});
|
||
|
}
|
||
|
|
||
|
video_stream_encoder_->SendKeyFrame();
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::Stop() {
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
RTC_LOG(LS_INFO) << "VideoSendStream::Stop";
|
||
|
if (!rtp_video_sender_->IsActive())
|
||
|
return;
|
||
|
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
|
||
|
rtp_video_sender_->SetActive(false);
|
||
|
StopVideoSendStream();
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::StopVideoSendStream() {
|
||
|
bitrate_allocator_->RemoveObserver(this);
|
||
|
check_encoder_activity_task_.Stop();
|
||
|
video_stream_encoder_->OnBitrateUpdated(DataRate::Zero(), DataRate::Zero(),
|
||
|
DataRate::Zero(), 0, 0, 0);
|
||
|
stats_proxy_->OnSetEncoderTargetRate(0);
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::SignalEncoderTimedOut() {
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
// If the encoder has not produced anything the last kEncoderTimeOut and it
|
||
|
// is supposed to, deregister as BitrateAllocatorObserver. This can happen
|
||
|
// if a camera stops producing frames.
|
||
|
if (encoder_target_rate_bps_ > 0) {
|
||
|
RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
|
||
|
bitrate_allocator_->RemoveObserver(this);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::OnBitrateAllocationUpdated(
|
||
|
const VideoBitrateAllocation& allocation) {
|
||
|
if (!worker_queue_->IsCurrent()) {
|
||
|
auto ptr = weak_ptr_;
|
||
|
worker_queue_->PostTask([=] {
|
||
|
if (!ptr.get())
|
||
|
return;
|
||
|
ptr->OnBitrateAllocationUpdated(allocation);
|
||
|
});
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
|
||
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
if (encoder_target_rate_bps_ != 0) {
|
||
|
if (video_bitrate_allocation_context_) {
|
||
|
// If new allocation is within kMaxVbaSizeDifferencePercent larger than
|
||
|
// the previously sent allocation and the same streams are still enabled,
|
||
|
// it is considered "similar". We do not want send similar allocations
|
||
|
// more once per kMaxVbaThrottleTimeMs.
|
||
|
const VideoBitrateAllocation& last =
|
||
|
video_bitrate_allocation_context_->last_sent_allocation;
|
||
|
const bool is_similar =
|
||
|
allocation.get_sum_bps() >= last.get_sum_bps() &&
|
||
|
allocation.get_sum_bps() <
|
||
|
(last.get_sum_bps() * (100 + kMaxVbaSizeDifferencePercent)) /
|
||
|
100 &&
|
||
|
SameStreamsEnabled(allocation, last);
|
||
|
if (is_similar &&
|
||
|
(now_ms - video_bitrate_allocation_context_->last_send_time_ms) <
|
||
|
kMaxVbaThrottleTimeMs) {
|
||
|
// This allocation is too similar, cache it and return.
|
||
|
video_bitrate_allocation_context_->throttled_allocation = allocation;
|
||
|
return;
|
||
|
}
|
||
|
} else {
|
||
|
video_bitrate_allocation_context_.emplace();
|
||
|
}
|
||
|
|
||
|
video_bitrate_allocation_context_->last_sent_allocation = allocation;
|
||
|
video_bitrate_allocation_context_->throttled_allocation.reset();
|
||
|
video_bitrate_allocation_context_->last_send_time_ms = now_ms;
|
||
|
|
||
|
// Send bitrate allocation metadata only if encoder is not paused.
|
||
|
rtp_video_sender_->OnBitrateAllocationUpdated(allocation);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::SignalEncoderActive() {
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
if (rtp_video_sender_->IsActive()) {
|
||
|
RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
|
||
|
bitrate_allocator_->AddObserver(this, GetAllocationConfig());
|
||
|
}
|
||
|
}
|
||
|
|
||
|
MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const {
|
||
|
return MediaStreamAllocationConfig{
|
||
|
static_cast<uint32_t>(encoder_min_bitrate_bps_),
|
||
|
encoder_max_bitrate_bps_,
|
||
|
static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_),
|
||
|
/* priority_bitrate */ 0,
|
||
|
!config_->suspend_below_min_bitrate,
|
||
|
encoder_bitrate_priority_};
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::OnEncoderConfigurationChanged(
|
||
|
std::vector<VideoStream> streams,
|
||
|
bool is_svc,
|
||
|
VideoEncoderConfig::ContentType content_type,
|
||
|
int min_transmit_bitrate_bps) {
|
||
|
if (!worker_queue_->IsCurrent()) {
|
||
|
rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
|
||
|
worker_queue_->PostTask([send_stream, streams, is_svc, content_type,
|
||
|
min_transmit_bitrate_bps]() mutable {
|
||
|
if (send_stream) {
|
||
|
send_stream->OnEncoderConfigurationChanged(
|
||
|
std::move(streams), is_svc, content_type, min_transmit_bitrate_bps);
|
||
|
}
|
||
|
});
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
|
||
|
TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
|
||
|
const VideoCodecType codec_type =
|
||
|
PayloadStringToCodecType(config_->rtp.payload_name);
|
||
|
|
||
|
const absl::optional<DataRate> experimental_min_bitrate =
|
||
|
GetExperimentalMinVideoBitrate(codec_type);
|
||
|
encoder_min_bitrate_bps_ =
|
||
|
experimental_min_bitrate
|
||
|
? experimental_min_bitrate->bps()
|
||
|
: std::max(streams[0].min_bitrate_bps, kDefaultMinVideoBitrateBps);
|
||
|
|
||
|
encoder_max_bitrate_bps_ = 0;
|
||
|
double stream_bitrate_priority_sum = 0;
|
||
|
for (const auto& stream : streams) {
|
||
|
// We don't want to allocate more bitrate than needed to inactive streams.
|
||
|
encoder_max_bitrate_bps_ += stream.active ? stream.max_bitrate_bps : 0;
|
||
|
if (stream.bitrate_priority) {
|
||
|
RTC_DCHECK_GT(*stream.bitrate_priority, 0);
|
||
|
stream_bitrate_priority_sum += *stream.bitrate_priority;
|
||
|
}
|
||
|
}
|
||
|
RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
|
||
|
encoder_bitrate_priority_ = stream_bitrate_priority_sum;
|
||
|
encoder_max_bitrate_bps_ =
|
||
|
std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
|
||
|
encoder_max_bitrate_bps_);
|
||
|
|
||
|
// TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead.
|
||
|
max_padding_bitrate_ = CalculateMaxPadBitrateBps(
|
||
|
streams, is_svc, content_type, min_transmit_bitrate_bps,
|
||
|
config_->suspend_below_min_bitrate, has_alr_probing_);
|
||
|
|
||
|
// Clear stats for disabled layers.
|
||
|
for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) {
|
||
|
stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]);
|
||
|
}
|
||
|
|
||
|
const size_t num_temporal_layers =
|
||
|
streams.back().num_temporal_layers.value_or(1);
|
||
|
|
||
|
rtp_video_sender_->SetEncodingData(streams[0].width, streams[0].height,
|
||
|
num_temporal_layers);
|
||
|
|
||
|
if (rtp_video_sender_->IsActive()) {
|
||
|
// The send stream is started already. Update the allocator with new bitrate
|
||
|
// limits.
|
||
|
bitrate_allocator_->AddObserver(this, GetAllocationConfig());
|
||
|
}
|
||
|
}
|
||
|
|
||
|
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
|
||
|
const EncodedImage& encoded_image,
|
||
|
const CodecSpecificInfo* codec_specific_info,
|
||
|
const RTPFragmentationHeader* fragmentation) {
|
||
|
// Encoded is called on whatever thread the real encoder implementation run
|
||
|
// on. In the case of hardware encoders, there might be several encoders
|
||
|
// running in parallel on different threads.
|
||
|
|
||
|
// Indicate that there still is activity going on.
|
||
|
activity_ = true;
|
||
|
|
||
|
auto enable_padding_task = [this]() {
|
||
|
if (disable_padding_) {
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
disable_padding_ = false;
|
||
|
// To ensure that padding bitrate is propagated to the bitrate allocator.
|
||
|
SignalEncoderActive();
|
||
|
}
|
||
|
};
|
||
|
if (!worker_queue_->IsCurrent()) {
|
||
|
worker_queue_->PostTask(enable_padding_task);
|
||
|
} else {
|
||
|
enable_padding_task();
|
||
|
}
|
||
|
|
||
|
EncodedImageCallback::Result result(EncodedImageCallback::Result::OK);
|
||
|
result = rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info,
|
||
|
fragmentation);
|
||
|
// Check if there's a throttled VideoBitrateAllocation that we should try
|
||
|
// sending.
|
||
|
rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
|
||
|
auto update_task = [send_stream]() {
|
||
|
if (send_stream) {
|
||
|
RTC_DCHECK_RUN_ON(send_stream->worker_queue_);
|
||
|
auto& context = send_stream->video_bitrate_allocation_context_;
|
||
|
if (context && context->throttled_allocation) {
|
||
|
send_stream->OnBitrateAllocationUpdated(*context->throttled_allocation);
|
||
|
}
|
||
|
}
|
||
|
};
|
||
|
if (!worker_queue_->IsCurrent()) {
|
||
|
worker_queue_->PostTask(update_task);
|
||
|
} else {
|
||
|
update_task();
|
||
|
}
|
||
|
|
||
|
return result;
|
||
|
}
|
||
|
|
||
|
void VideoSendStreamImpl::OnDroppedFrame(
|
||
|
EncodedImageCallback::DropReason reason) {
|
||
|
activity_ = true;
|
||
|
}
|
||
|
|
||
|
std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
|
||
|
return rtp_video_sender_->GetRtpStates();
|
||
|
}
|
||
|
|
||
|
std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
|
||
|
const {
|
||
|
return rtp_video_sender_->GetRtpPayloadStates();
|
||
|
}
|
||
|
|
||
|
uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
|
||
|
RTC_DCHECK_RUN_ON(worker_queue_);
|
||
|
RTC_DCHECK(rtp_video_sender_->IsActive())
|
||
|
<< "VideoSendStream::Start has not been called.";
|
||
|
|
||
|
// When the BWE algorithm doesn't pass a stable estimate, we'll use the
|
||
|
// unstable one instead.
|
||
|
if (update.stable_target_bitrate.IsZero()) {
|
||
|
update.stable_target_bitrate = update.target_bitrate;
|
||
|
}
|
||
|
|
||
|
rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate());
|
||
|
encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
|
||
|
const uint32_t protection_bitrate_bps =
|
||
|
rtp_video_sender_->GetProtectionBitrateBps();
|
||
|
DataRate link_allocation = DataRate::Zero();
|
||
|
if (encoder_target_rate_bps_ > protection_bitrate_bps) {
|
||
|
link_allocation =
|
||
|
DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
|
||
|
}
|
||
|
DataRate overhead =
|
||
|
update.target_bitrate - DataRate::BitsPerSec(encoder_target_rate_bps_);
|
||
|
DataRate encoder_stable_target_rate = update.stable_target_bitrate;
|
||
|
if (encoder_stable_target_rate > overhead) {
|
||
|
encoder_stable_target_rate = encoder_stable_target_rate - overhead;
|
||
|
} else {
|
||
|
encoder_stable_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
|
||
|
}
|
||
|
|
||
|
encoder_target_rate_bps_ =
|
||
|
std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
|
||
|
|
||
|
encoder_stable_target_rate =
|
||
|
std::min(DataRate::BitsPerSec(encoder_max_bitrate_bps_),
|
||
|
encoder_stable_target_rate);
|
||
|
|
||
|
DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
|
||
|
link_allocation = std::max(encoder_target_rate, link_allocation);
|
||
|
video_stream_encoder_->OnBitrateUpdated(
|
||
|
encoder_target_rate, encoder_stable_target_rate, link_allocation,
|
||
|
rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
|
||
|
update.round_trip_time.ms(), update.cwnd_reduce_ratio);
|
||
|
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
|
||
|
return protection_bitrate_bps;
|
||
|
}
|
||
|
|
||
|
} // namespace internal
|
||
|
} // namespace webrtc
|