webrtc_m130/call/video_send_stream.cc
Niels Möller 259a497632 Reland "Reland "Move rtp-specific config out of EncoderSettings.""
This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e.

Reason for revert: Intend to investigate and fix perf problems.

Original change's description:
> Revert "Reland "Move rtp-specific config out of EncoderSettings.""
> 
> This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
> 
> Reason for revert: Regression in ramp up perf tests.
> 
> Original change's description:
> > Reland "Move rtp-specific config out of EncoderSettings."
> >
> > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
> >
> > Original change's description:
> > > Move rtp-specific config out of EncoderSettings.
> > >
> > > In VideoSendStream::Config, move payload_name and payload_type from
> > > EncoderSettings to Rtp.
> > >
> > > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > > and should perhaps be renamed in a follow up cl. It's no longer
> > > passed as an argument to VideoCodecInitializer::SetupCodec.
> > >
> > > The latter then needs a different way to know the codec type,
> > > which is provided by a new codec_type member in VideoEncoderConfig.
> > >
> > > Bug: webrtc:8830
> > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22532}
> >
> > Bug: webrtc:8830
> > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> > Reviewed-on: https://webrtc-review.googlesource.com/63721
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22595}
> 
> TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
> 
> Bug: webrtc:8830,chromium:827080
> Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
> Reviewed-on: https://webrtc-review.googlesource.com/65520
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22677}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8830, chromium:827080
Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f
Reviewed-on: https://webrtc-review.googlesource.com/66862
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22751}
2018-04-05 14:30:09 +00:00

186 lines
6.4 KiB
C++

/*
* Copyright (c) 2017 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 "call/video_send_stream.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
VideoSendStream::StreamStats::StreamStats() = default;
VideoSendStream::StreamStats::~StreamStats() = default;
std::string VideoSendStream::StreamStats::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "width: " << width << ", ";
ss << "height: " << height << ", ";
ss << "key: " << frame_counts.key_frames << ", ";
ss << "delta: " << frame_counts.delta_frames << ", ";
ss << "total_bps: " << total_bitrate_bps << ", ";
ss << "retransmit_bps: " << retransmit_bitrate_bps << ", ";
ss << "avg_delay_ms: " << avg_delay_ms << ", ";
ss << "max_delay_ms: " << max_delay_ms << ", ";
ss << "cum_loss: " << rtcp_stats.packets_lost << ", ";
ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", ";
ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", ";
ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", ";
ss << "pli: " << rtcp_packet_type_counts.pli_packets;
return ss.str();
}
VideoSendStream::Stats::Stats() = default;
VideoSendStream::Stats::~Stats() = default;
std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "VideoSendStream stats: " << time_ms << ", {";
ss << "input_fps: " << input_frame_rate << ", ";
ss << "encode_fps: " << encode_frame_rate << ", ";
ss << "encode_ms: " << avg_encode_time_ms << ", ";
ss << "encode_usage_perc: " << encode_usage_percent << ", ";
ss << "target_bps: " << target_media_bitrate_bps << ", ";
ss << "media_bps: " << media_bitrate_bps << ", ";
ss << "preferred_media_bitrate_bps: " << preferred_media_bitrate_bps << ", ";
ss << "suspended: " << (suspended ? "true" : "false") << ", ";
ss << "bw_adapted: " << (bw_limited_resolution ? "true" : "false");
ss << '}';
for (const auto& substream : substreams) {
if (!substream.second.is_rtx && !substream.second.is_flexfec) {
ss << " {ssrc: " << substream.first << ", ";
ss << substream.second.ToString();
ss << '}';
}
}
return ss.str();
}
VideoSendStream::Config::Config(const Config&) = default;
VideoSendStream::Config::Config(Config&&) = default;
VideoSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default;
VideoSendStream::Config::Config::~Config() = default;
std::string VideoSendStream::Config::ToString() const {
char buf[2 * 1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{encoder_settings: " << encoder_settings.ToString();
ss << ", rtp: " << rtp.ToString();
ss << ", rtcp: " << rtcp.ToString();
ss << ", pre_encode_callback: "
<< (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr");
ss << ", post_encode_callback: "
<< (post_encode_callback ? "(EncodedFrameObserver)" : "nullptr");
ss << ", render_delay_ms: " << render_delay_ms;
ss << ", target_delay_ms: " << target_delay_ms;
ss << ", suspend_below_min_bitrate: "
<< (suspend_below_min_bitrate ? "on" : "off");
ss << '}';
return ss.str();
}
std::string VideoSendStream::Config::EncoderSettings::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{encoder_factory: "
<< (encoder_factory ? "(VideoEncoderFactory)" : "(nullptr)");
ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr");
ss << '}';
return ss.str();
}
VideoSendStream::Config::Rtp::Rtp() = default;
VideoSendStream::Config::Rtp::Rtp(const Rtp&) = default;
VideoSendStream::Config::Rtp::~Rtp() = default;
VideoSendStream::Config::Rtp::Flexfec::Flexfec() = default;
VideoSendStream::Config::Rtp::Flexfec::Flexfec(const Flexfec&) = default;
VideoSendStream::Config::Rtp::Flexfec::~Flexfec() = default;
std::string VideoSendStream::Config::Rtp::ToString() const {
char buf[2 * 1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{ssrcs: [";
for (size_t i = 0; i < ssrcs.size(); ++i) {
ss << ssrcs[i];
if (i != ssrcs.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", rtcp_mode: "
<< (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
: "RtcpMode::kReducedSize");
ss << ", max_packet_size: " << max_packet_size;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
ss << ", ulpfec: " << ulpfec.ToString();
ss << ", payload_name: " << payload_name;
ss << ", payload_type: " << payload_type;
ss << ", flexfec: {payload_type: " << flexfec.payload_type;
ss << ", ssrc: " << flexfec.ssrc;
ss << ", protected_media_ssrcs: [";
for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) {
ss << flexfec.protected_media_ssrcs[i];
if (i != flexfec.protected_media_ssrcs.size() - 1)
ss << ", ";
}
ss << "]}";
ss << ", rtx: " << rtx.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
VideoSendStream::Config::Rtp::Rtx::Rtx() = default;
VideoSendStream::Config::Rtp::Rtx::Rtx(const Rtx&) = default;
VideoSendStream::Config::Rtp::Rtx::~Rtx() = default;
std::string VideoSendStream::Config::Rtp::Rtx::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{ssrcs: [";
for (size_t i = 0; i < ssrcs.size(); ++i) {
ss << ssrcs[i];
if (i != ssrcs.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", payload_type: " << payload_type;
ss << '}';
return ss.str();
}
VideoSendStream::Config::Rtcp::Rtcp() = default;
VideoSendStream::Config::Rtcp::Rtcp(const Rtcp&) = default;
VideoSendStream::Config::Rtcp::~Rtcp() = default;
std::string VideoSendStream::Config::Rtcp::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{video_report_interval_ms: " << video_report_interval_ms;
ss << ", audio_report_interval_ms: " << audio_report_interval_ms;
ss << '}';
return ss.str();
}
} // namespace webrtc