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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 #include "webrtc/video/video_send_stream.h" | 10 #include "webrtc/video/video_send_stream.h" |
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| 253 std::string VideoSendStream::StreamStats::ToString() const { | 253 std::string VideoSendStream::StreamStats::ToString() const { |
| 254 std::stringstream ss; | 254 std::stringstream ss; |
| 255 ss << "width: " << width << ", "; | 255 ss << "width: " << width << ", "; |
| 256 ss << "height: " << height << ", "; | 256 ss << "height: " << height << ", "; |
| 257 ss << "key: " << frame_counts.key_frames << ", "; | 257 ss << "key: " << frame_counts.key_frames << ", "; |
| 258 ss << "delta: " << frame_counts.delta_frames << ", "; | 258 ss << "delta: " << frame_counts.delta_frames << ", "; |
| 259 ss << "total_bps: " << total_bitrate_bps << ", "; | 259 ss << "total_bps: " << total_bitrate_bps << ", "; |
| 260 ss << "retransmit_bps: " << retransmit_bitrate_bps << ", "; | 260 ss << "retransmit_bps: " << retransmit_bitrate_bps << ", "; |
| 261 ss << "avg_delay_ms: " << avg_delay_ms << ", "; | 261 ss << "avg_delay_ms: " << avg_delay_ms << ", "; |
| 262 ss << "max_delay_ms: " << max_delay_ms << ", "; | 262 ss << "max_delay_ms: " << max_delay_ms << ", "; |
| 263 ss << "cum_loss: " << rtcp_stats.cumulative_lost << ", "; | 263 ss << "cum_loss: " << rtcp_stats.packets_lost << ", "; |
| 264 ss << "max_ext_seq: " << rtcp_stats.extended_max_sequence_number << ", "; | 264 ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", "; |
| 265 ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", "; | 265 ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", "; |
| 266 ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", "; | 266 ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", "; |
| 267 ss << "pli: " << rtcp_packet_type_counts.pli_packets; | 267 ss << "pli: " << rtcp_packet_type_counts.pli_packets; |
| 268 return ss.str(); | 268 return ss.str(); |
| 269 } | 269 } |
| 270 | 270 |
| 271 namespace { | 271 namespace { |
| 272 | 272 |
| 273 bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) { | 273 bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) { |
| 274 rtc::Optional<VideoCodecType> codecType = | 274 rtc::Optional<VideoCodecType> codecType = |
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| 1368 std::min(config_->rtp.max_packet_size, | 1368 std::min(config_->rtp.max_packet_size, |
| 1369 kPathMTU - transport_overhead_bytes_per_packet_); | 1369 kPathMTU - transport_overhead_bytes_per_packet_); |
| 1370 | 1370 |
| 1371 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1371 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 1372 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1372 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
| 1373 } | 1373 } |
| 1374 } | 1374 } |
| 1375 | 1375 |
| 1376 } // namespace internal | 1376 } // namespace internal |
| 1377 } // namespace webrtc | 1377 } // namespace webrtc |
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