| /* |
| * Copyright (c) 2012 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 "modules/rtp_rtcp/source/rtp_sender_video.h" |
| |
| #include <stdlib.h> |
| #include <string.h> |
| |
| #include <limits> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/byte_io.h" |
| #include "modules/rtp_rtcp/source/rtp_format_video_generic.h" |
| #include "modules/rtp_rtcp/source/rtp_format_vp8.h" |
| #include "modules/rtp_rtcp/source/rtp_format_vp9.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ptr_util.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| constexpr size_t kRedForFecHeaderLength = 1; |
| constexpr int64_t kMaxUnretransmittableFrameIntervalMs = 33 * 4; |
| |
| void BuildRedPayload(const RtpPacketToSend& media_packet, |
| RtpPacketToSend* red_packet) { |
| uint8_t* red_payload = red_packet->AllocatePayload( |
| kRedForFecHeaderLength + media_packet.payload_size()); |
| RTC_DCHECK(red_payload); |
| red_payload[0] = media_packet.PayloadType(); |
| |
| auto media_payload = media_packet.payload(); |
| memcpy(&red_payload[kRedForFecHeaderLength], media_payload.data(), |
| media_payload.size()); |
| } |
| } // namespace |
| |
| RTPSenderVideo::RTPSenderVideo(Clock* clock, |
| RTPSender* rtp_sender, |
| FlexfecSender* flexfec_sender) |
| : rtp_sender_(rtp_sender), |
| clock_(clock), |
| video_type_(kRtpVideoGeneric), |
| retransmission_settings_(kRetransmitBaseLayer | |
| kConditionallyRetransmitHigherLayers), |
| last_rotation_(kVideoRotation_0), |
| red_payload_type_(-1), |
| ulpfec_payload_type_(-1), |
| flexfec_sender_(flexfec_sender), |
| delta_fec_params_{0, 1, kFecMaskRandom}, |
| key_fec_params_{0, 1, kFecMaskRandom}, |
| fec_bitrate_(1000, RateStatistics::kBpsScale), |
| video_bitrate_(1000, RateStatistics::kBpsScale) {} |
| |
| RTPSenderVideo::~RTPSenderVideo() {} |
| |
| void RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes video_type) { |
| video_type_ = video_type; |
| } |
| |
| RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const { |
| return video_type_; |
| } |
| |
| // Static. |
| RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload( |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| int8_t payload_type) { |
| RtpVideoCodecTypes video_type = kRtpVideoGeneric; |
| if (RtpUtility::StringCompare(payload_name, "VP8", 3)) { |
| video_type = kRtpVideoVp8; |
| } else if (RtpUtility::StringCompare(payload_name, "VP9", 3)) { |
| video_type = kRtpVideoVp9; |
| } else if (RtpUtility::StringCompare(payload_name, "H264", 4)) { |
| video_type = kRtpVideoH264; |
| } else if (RtpUtility::StringCompare(payload_name, "I420", 4)) { |
| video_type = kRtpVideoGeneric; |
| } else if (RtpUtility::StringCompare(payload_name, "stereo", 6)) { |
| video_type = kRtpVideoGeneric; |
| } else { |
| video_type = kRtpVideoGeneric; |
| } |
| VideoPayload vp; |
| vp.videoCodecType = video_type; |
| return new RtpUtility::Payload(payload_name, PayloadUnion(vp)); |
| } |
| |
| void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet, |
| StorageType storage) { |
| // Remember some values about the packet before sending it away. |
| size_t packet_size = packet->size(); |
| uint16_t seq_num = packet->SequenceNumber(); |
| uint32_t rtp_timestamp = packet->Timestamp(); |
| if (!rtp_sender_->SendToNetwork(std::move(packet), storage, |
| RtpPacketSender::kLowPriority)) { |
| RTC_LOG(LS_WARNING) << "Failed to send video packet " << seq_num; |
| return; |
| } |
| rtc::CritScope cs(&stats_crit_); |
| video_bitrate_.Update(packet_size, clock_->TimeInMilliseconds()); |
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
| "Video::PacketNormal", "timestamp", rtp_timestamp, |
| "seqnum", seq_num); |
| } |
| |
| void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec( |
| std::unique_ptr<RtpPacketToSend> media_packet, |
| StorageType media_packet_storage, |
| bool protect_media_packet) { |
| uint32_t rtp_timestamp = media_packet->Timestamp(); |
| uint16_t media_seq_num = media_packet->SequenceNumber(); |
| |
| std::unique_ptr<RtpPacketToSend> red_packet( |
| new RtpPacketToSend(*media_packet)); |
| BuildRedPayload(*media_packet, red_packet.get()); |
| |
| std::vector<std::unique_ptr<RedPacket>> fec_packets; |
| StorageType fec_storage = kDontRetransmit; |
| { |
| // Only protect while creating RED and FEC packets, not when sending. |
| rtc::CritScope cs(&crit_); |
| red_packet->SetPayloadType(red_payload_type_); |
| if (ulpfec_enabled()) { |
| if (protect_media_packet) { |
| ulpfec_generator_.AddRtpPacketAndGenerateFec( |
| media_packet->data(), media_packet->payload_size(), |
| media_packet->headers_size()); |
| } |
| uint16_t num_fec_packets = ulpfec_generator_.NumAvailableFecPackets(); |
| if (num_fec_packets > 0) { |
| uint16_t first_fec_sequence_number = |
| rtp_sender_->AllocateSequenceNumber(num_fec_packets); |
| fec_packets = ulpfec_generator_.GetUlpfecPacketsAsRed( |
| red_payload_type_, ulpfec_payload_type_, first_fec_sequence_number); |
| RTC_DCHECK_EQ(num_fec_packets, fec_packets.size()); |
| if (retransmission_settings_ & kRetransmitFECPackets) |
| fec_storage = kAllowRetransmission; |
| } |
| } |
| } |
| // Send |red_packet| instead of |packet| for allocated sequence number. |
| size_t red_packet_size = red_packet->size(); |
| if (rtp_sender_->SendToNetwork(std::move(red_packet), media_packet_storage, |
| RtpPacketSender::kLowPriority)) { |
| rtc::CritScope cs(&stats_crit_); |
| video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds()); |
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
| "Video::PacketRed", "timestamp", rtp_timestamp, |
| "seqnum", media_seq_num); |
| } else { |
| RTC_LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num; |
| } |
| for (const auto& fec_packet : fec_packets) { |
| // TODO(danilchap): Make ulpfec_generator_ generate RtpPacketToSend to avoid |
| // reparsing them. |
| std::unique_ptr<RtpPacketToSend> rtp_packet( |
| new RtpPacketToSend(*media_packet)); |
| RTC_CHECK(rtp_packet->Parse(fec_packet->data(), fec_packet->length())); |
| rtp_packet->set_capture_time_ms(media_packet->capture_time_ms()); |
| uint16_t fec_sequence_number = rtp_packet->SequenceNumber(); |
| if (rtp_sender_->SendToNetwork(std::move(rtp_packet), fec_storage, |
| RtpPacketSender::kLowPriority)) { |
| rtc::CritScope cs(&stats_crit_); |
| fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds()); |
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
| "Video::PacketUlpfec", "timestamp", rtp_timestamp, |
| "seqnum", fec_sequence_number); |
| } else { |
| RTC_LOG(LS_WARNING) << "Failed to send ULPFEC packet " |
| << fec_sequence_number; |
| } |
| } |
| } |
| |
| void RTPSenderVideo::SendVideoPacketWithFlexfec( |
| std::unique_ptr<RtpPacketToSend> media_packet, |
| StorageType media_packet_storage, |
| bool protect_media_packet) { |
| RTC_DCHECK(flexfec_sender_); |
| |
| if (protect_media_packet) |
| flexfec_sender_->AddRtpPacketAndGenerateFec(*media_packet); |
| |
| SendVideoPacket(std::move(media_packet), media_packet_storage); |
| |
| if (flexfec_sender_->FecAvailable()) { |
| std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets = |
| flexfec_sender_->GetFecPackets(); |
| for (auto& fec_packet : fec_packets) { |
| size_t packet_length = fec_packet->size(); |
| uint32_t timestamp = fec_packet->Timestamp(); |
| uint16_t seq_num = fec_packet->SequenceNumber(); |
| if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit, |
| RtpPacketSender::kLowPriority)) { |
| rtc::CritScope cs(&stats_crit_); |
| fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds()); |
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
| "Video::PacketFlexfec", "timestamp", timestamp, |
| "seqnum", seq_num); |
| } else { |
| RTC_LOG(LS_WARNING) << "Failed to send FlexFEC packet " << seq_num; |
| } |
| } |
| } |
| } |
| |
| void RTPSenderVideo::SetUlpfecConfig(int red_payload_type, |
| int ulpfec_payload_type) { |
| // Sanity check. Per the definition of UlpfecConfig (see config.h), |
| // a payload type of -1 means that the corresponding feature is |
| // turned off. |
| RTC_DCHECK_GE(red_payload_type, -1); |
| RTC_DCHECK_LE(red_payload_type, 127); |
| RTC_DCHECK_GE(ulpfec_payload_type, -1); |
| RTC_DCHECK_LE(ulpfec_payload_type, 127); |
| |
| rtc::CritScope cs(&crit_); |
| red_payload_type_ = red_payload_type; |
| ulpfec_payload_type_ = ulpfec_payload_type; |
| |
| // Must not enable ULPFEC without RED. |
| // TODO(brandtr): We currently support enabling RED without ULPFEC. Change |
| // this when we have removed the RED/RTX send-side workaround, so that we |
| // ensure that RED and ULPFEC are only enabled together. |
| RTC_DCHECK(red_enabled() || !ulpfec_enabled()); |
| |
| // Reset FEC parameters. |
| delta_fec_params_ = FecProtectionParams{0, 1, kFecMaskRandom}; |
| key_fec_params_ = FecProtectionParams{0, 1, kFecMaskRandom}; |
| } |
| |
| void RTPSenderVideo::GetUlpfecConfig(int* red_payload_type, |
| int* ulpfec_payload_type) const { |
| rtc::CritScope cs(&crit_); |
| *red_payload_type = red_payload_type_; |
| *ulpfec_payload_type = ulpfec_payload_type_; |
| } |
| |
| size_t RTPSenderVideo::CalculateFecPacketOverhead() const { |
| if (flexfec_enabled()) |
| return flexfec_sender_->MaxPacketOverhead(); |
| |
| size_t overhead = 0; |
| if (red_enabled()) { |
| // The RED overhead is due to a small header. |
| overhead += kRedForFecHeaderLength; |
| } |
| if (ulpfec_enabled()) { |
| // For ULPFEC, the overhead is the FEC headers plus RED for FEC header |
| // (see above) plus anything in RTP header beyond the 12 bytes base header |
| // (CSRC list, extensions...) |
| // This reason for the header extensions to be included here is that |
| // from an FEC viewpoint, they are part of the payload to be protected. |
| // (The base RTP header is already protected by the FEC header.) |
| overhead += ulpfec_generator_.MaxPacketOverhead() + |
| (rtp_sender_->RtpHeaderLength() - kRtpHeaderSize); |
| } |
| return overhead; |
| } |
| |
| void RTPSenderVideo::SetFecParameters(const FecProtectionParams& delta_params, |
| const FecProtectionParams& key_params) { |
| rtc::CritScope cs(&crit_); |
| delta_fec_params_ = delta_params; |
| key_fec_params_ = key_params; |
| } |
| |
| rtc::Optional<uint32_t> RTPSenderVideo::FlexfecSsrc() const { |
| if (flexfec_sender_) { |
| return flexfec_sender_->ssrc(); |
| } |
| return rtc::nullopt; |
| } |
| |
| bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type, |
| FrameType frame_type, |
| int8_t payload_type, |
| uint32_t rtp_timestamp, |
| int64_t capture_time_ms, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* video_header, |
| int64_t expected_retransmission_time_ms) { |
| if (payload_size == 0) |
| return false; |
| |
| // Create header that will be reused in all packets. |
| std::unique_ptr<RtpPacketToSend> rtp_header = rtp_sender_->AllocatePacket(); |
| rtp_header->SetPayloadType(payload_type); |
| rtp_header->SetTimestamp(rtp_timestamp); |
| rtp_header->set_capture_time_ms(capture_time_ms); |
| auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header); |
| |
| size_t fec_packet_overhead; |
| bool red_enabled; |
| int32_t retransmission_settings; |
| { |
| rtc::CritScope cs(&crit_); |
| // According to |
| // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ |
| // ts_126114v120700p.pdf Section 7.4.5: |
| // The MTSI client shall add the payload bytes as defined in this clause |
| // onto the last RTP packet in each group of packets which make up a key |
| // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265 |
| // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP |
| // packet in each group of packets which make up another type of frame |
| // (e.g. a P-Frame) only if the current value is different from the previous |
| // value sent. |
| if (video_header) { |
| // Set rotation when key frame or when changed (to follow standard). |
| // Or when different from 0 (to follow current receiver implementation). |
| VideoRotation current_rotation = video_header->rotation; |
| if (frame_type == kVideoFrameKey || current_rotation != last_rotation_ || |
| current_rotation != kVideoRotation_0) |
| last_packet->SetExtension<VideoOrientation>(current_rotation); |
| last_rotation_ = current_rotation; |
| // Report content type only for key frames. |
| if (frame_type == kVideoFrameKey && |
| video_header->content_type != VideoContentType::UNSPECIFIED) { |
| last_packet->SetExtension<VideoContentTypeExtension>( |
| video_header->content_type); |
| } |
| if (video_header->video_timing.flags != TimingFrameFlags::kInvalid) { |
| last_packet->SetExtension<VideoTimingExtension>( |
| video_header->video_timing); |
| } |
| } |
| |
| // FEC settings. |
| const FecProtectionParams& fec_params = |
| frame_type == kVideoFrameKey ? key_fec_params_ : delta_fec_params_; |
| if (flexfec_enabled()) |
| flexfec_sender_->SetFecParameters(fec_params); |
| if (ulpfec_enabled()) |
| ulpfec_generator_.SetFecParameters(fec_params); |
| |
| fec_packet_overhead = CalculateFecPacketOverhead(); |
| red_enabled = this->red_enabled(); |
| retransmission_settings = retransmission_settings_; |
| } |
| |
| size_t packet_capacity = rtp_sender_->MaxRtpPacketSize() - |
| fec_packet_overhead - |
| (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0); |
| RTC_DCHECK_LE(packet_capacity, rtp_header->capacity()); |
| RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size()); |
| RTC_DCHECK_GT(packet_capacity, last_packet->headers_size()); |
| size_t max_data_payload_length = packet_capacity - rtp_header->headers_size(); |
| RTC_DCHECK_GE(last_packet->headers_size(), rtp_header->headers_size()); |
| size_t last_packet_reduction_len = |
| last_packet->headers_size() - rtp_header->headers_size(); |
| |
| std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( |
| video_type, max_data_payload_length, last_packet_reduction_len, |
| video_header ? &(video_header->codecHeader) : nullptr, frame_type)); |
| |
| const uint8_t temporal_id = |
| video_header ? GetTemporalId(*video_header) : kNoTemporalIdx; |
| StorageType storage = GetStorageType(temporal_id, retransmission_settings, |
| expected_retransmission_time_ms); |
| |
| size_t num_packets = |
| packetizer->SetPayloadData(payload_data, payload_size, fragmentation); |
| |
| if (num_packets == 0) |
| return false; |
| |
| bool first_frame = first_frame_sent_(); |
| for (size_t i = 0; i < num_packets; ++i) { |
| bool last = (i + 1) == num_packets; |
| auto packet = last ? std::move(last_packet) |
| : rtc::MakeUnique<RtpPacketToSend>(*rtp_header); |
| if (!packetizer->NextPacket(packet.get())) |
| return false; |
| RTC_DCHECK_LE(packet->payload_size(), |
| last ? max_data_payload_length - last_packet_reduction_len |
| : max_data_payload_length); |
| if (!rtp_sender_->AssignSequenceNumber(packet.get())) |
| return false; |
| |
| // No FEC protection for upper temporal layers, if used. |
| bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx; |
| |
| // Put packetization finish timestamp into extension. |
| if (packet->HasExtension<VideoTimingExtension>()) { |
| packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds()); |
| // TODO(ilnik): Due to webrtc:7859, packets with timing extensions are not |
| // protected by FEC. It reduces FEC efficiency a bit. When FEC is moved |
| // below the pacer, it can be re-enabled for these packets. |
| // NOTE: Any RTP stream processor in the network, modifying 'network' |
| // timestamps in the timing frames extension have to be an end-point for |
| // FEC, otherwise recovered by FEC packets will be corrupted. |
| protect_packet = false; |
| } |
| |
| if (flexfec_enabled()) { |
| // TODO(brandtr): Remove the FlexFEC code path when FlexfecSender |
| // is wired up to PacedSender instead. |
| SendVideoPacketWithFlexfec(std::move(packet), storage, protect_packet); |
| } else if (red_enabled) { |
| SendVideoPacketAsRedMaybeWithUlpfec(std::move(packet), storage, |
| protect_packet); |
| } else { |
| SendVideoPacket(std::move(packet), storage); |
| } |
| |
| if (first_frame) { |
| if (i == 0) { |
| RTC_LOG(LS_INFO) |
| << "Sent first RTP packet of the first video frame (pre-pacer)"; |
| } |
| if (last) { |
| RTC_LOG(LS_INFO) |
| << "Sent last RTP packet of the first video frame (pre-pacer)"; |
| } |
| } |
| } |
| |
| TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp", |
| rtp_timestamp); |
| return true; |
| } |
| |
| uint32_t RTPSenderVideo::VideoBitrateSent() const { |
| rtc::CritScope cs(&stats_crit_); |
| return video_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0); |
| } |
| |
| uint32_t RTPSenderVideo::FecOverheadRate() const { |
| rtc::CritScope cs(&stats_crit_); |
| return fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0); |
| } |
| |
| int RTPSenderVideo::SelectiveRetransmissions() const { |
| rtc::CritScope cs(&crit_); |
| return retransmission_settings_; |
| } |
| |
| void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) { |
| rtc::CritScope cs(&crit_); |
| retransmission_settings_ = settings; |
| } |
| |
| StorageType RTPSenderVideo::GetStorageType( |
| uint8_t temporal_id, |
| int32_t retransmission_settings, |
| int64_t expected_retransmission_time_ms) { |
| if (retransmission_settings == kRetransmitOff) |
| return StorageType::kDontRetransmit; |
| if (retransmission_settings == kRetransmitAllPackets) |
| return StorageType::kAllowRetransmission; |
| |
| rtc::CritScope cs(&stats_crit_); |
| // Media packet storage. |
| if ((retransmission_settings & kConditionallyRetransmitHigherLayers) && |
| UpdateConditionalRetransmit(temporal_id, |
| expected_retransmission_time_ms)) { |
| retransmission_settings |= kRetransmitHigherLayers; |
| } |
| |
| if (temporal_id == kNoTemporalIdx) |
| return kAllowRetransmission; |
| |
| if ((retransmission_settings & kRetransmitBaseLayer) && temporal_id == 0) |
| return kAllowRetransmission; |
| |
| if ((retransmission_settings & kRetransmitHigherLayers) && temporal_id > 0) |
| return kAllowRetransmission; |
| |
| return kDontRetransmit; |
| } |
| |
| uint8_t RTPSenderVideo::GetTemporalId(const RTPVideoHeader& header) { |
| switch (header.codec) { |
| case kRtpVideoVp8: |
| return header.codecHeader.VP8.temporalIdx; |
| case kRtpVideoVp9: |
| return header.codecHeader.VP9.temporal_idx; |
| default: |
| return kNoTemporalIdx; |
| } |
| } |
| |
| bool RTPSenderVideo::UpdateConditionalRetransmit( |
| uint8_t temporal_id, |
| int64_t expected_retransmission_time_ms) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| // Update stats for any temporal layer. |
| TemporalLayerStats* current_layer_stats = |
| &frame_stats_by_temporal_layer_[temporal_id]; |
| current_layer_stats->frame_rate_fp1000s.Update(1, now_ms); |
| int64_t tl_frame_interval = now_ms - current_layer_stats->last_frame_time_ms; |
| current_layer_stats->last_frame_time_ms = now_ms; |
| |
| // Conditional retransmit only applies to upper layers. |
| if (temporal_id != kNoTemporalIdx && temporal_id > 0) { |
| if (tl_frame_interval >= kMaxUnretransmittableFrameIntervalMs) { |
| // Too long since a retransmittable frame in this layer, enable NACK |
| // protection. |
| return true; |
| } else { |
| // Estimate when the next frame of any lower layer will be sent. |
| const int64_t kUndefined = std::numeric_limits<int64_t>::max(); |
| int64_t expected_next_frame_time = kUndefined; |
| for (int i = temporal_id - 1; i >= 0; --i) { |
| TemporalLayerStats* stats = &frame_stats_by_temporal_layer_[i]; |
| rtc::Optional<uint32_t> rate = stats->frame_rate_fp1000s.Rate(now_ms); |
| if (rate) { |
| int64_t tl_next = stats->last_frame_time_ms + 1000000 / *rate; |
| if (tl_next - now_ms > -expected_retransmission_time_ms && |
| tl_next < expected_next_frame_time) { |
| expected_next_frame_time = tl_next; |
| } |
| } |
| } |
| |
| if (expected_next_frame_time == kUndefined || |
| expected_next_frame_time - now_ms > expected_retransmission_time_ms) { |
| // The next frame in a lower layer is expected at a later time (or |
| // unable to tell due to lack of data) than a retransmission is |
| // estimated to be able to arrive, so allow this packet to be nacked. |
| return true; |
| } |
| } |
| } |
| |
| return false; |
| } |
| |
| } // namespace webrtc |
