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Revert "Add ability to emulate degraded network in Call via field trial"

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This reverts commit 31a12c557dcd84a31f9c3f2d8858d9646c2a3135.

Reason for revert: Breaks downstream project.

Original change's description:
> Add ability to emulate degraded network in Call via field trial
> 
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
> 
> Also includes some refactorings.
> 
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org

Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This commit is contained in:
Ilya Nikolaevskiy
2018-03-14 10:51:50 +00:00
committed by Commit Bot
parent 31a12c557d
commit 16cba5c18d
13 changed files with 482 additions and 1130 deletions
Download Patch File Download Diff File Expand all files Collapse all files

31
call/call.cc
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@ -292,24 +292,19 @@ class Call : public webrtc::Call,
// single mapping from ssrc to a more abstract receive stream, with
// accessor methods for all configuration we need at this level.
struct ReceiveRtpConfig {
explicit ReceiveRtpConfig(const webrtc::AudioReceiveStream::Config& config)
: extensions(config.rtp.extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
explicit ReceiveRtpConfig(const webrtc::VideoReceiveStream::Config& config)
: extensions(config.rtp.extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
explicit ReceiveRtpConfig(const FlexfecReceiveStream::Config& config)
: extensions(config.rtp_header_extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
ReceiveRtpConfig() = default; // Needed by std::map
ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
bool use_send_side_bwe)
: extensions(extensions), use_send_side_bwe(use_send_side_bwe) {}
// Registered RTP header extensions for each stream. Note that RTP header
// extensions are negotiated per track ("m= line") in the SDP, but we have
// no notion of tracks at the Call level. We therefore store the RTP header
// extensions per SSRC instead, which leads to some storage overhead.
const RtpHeaderExtensionMap extensions;
RtpHeaderExtensionMap extensions;
// Set if both RTP extension the RTCP feedback message needed for
// send side BWE are negotiated.
const bool use_send_side_bwe;
bool use_send_side_bwe = false;
};
std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
RTC_GUARDED_BY(receive_crit_);
@ -646,7 +641,8 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
module_process_thread_.get(), config, config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
receive_rtp_config_.emplace(config.rtp.remote_ssrc, config);
receive_rtp_config_[config.rtp.remote_ssrc] =
ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
audio_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group);
@ -795,6 +791,8 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
module_process_thread_.get(), call_stats_.get());
const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
ReceiveRtpConfig receive_config(config.rtp.extensions,
UseSendSideBwe(config));
{
WriteLockScoped write_lock(*receive_crit_);
if (config.rtp.rtx_ssrc) {
@ -802,9 +800,9 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
// stream. Since the transport_send_cc negotiation is per payload
// type, we may get an incorrect value for the rtx stream, but
// that is unlikely to matter in practice.
receive_rtp_config_.emplace(config.rtp.rtx_ssrc, config);
receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
}
receive_rtp_config_.emplace(config.rtp.remote_ssrc, config);
receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
video_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group);
}
@ -867,7 +865,8 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
receive_rtp_config_.end());
receive_rtp_config_.emplace(config.remote_ssrc, config);
receive_rtp_config_[config.remote_ssrc] =
ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
}
// TODO(brandtr): Store config in RtcEventLog here.
@ -1313,7 +1312,7 @@ void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
// deregistering in the |receive_rtp_config_| map is protected by that lock.
// So by not passing the packet on to demuxing in this case, we prevent
// incoming packets to be passed on via the demuxer to a receive stream
// which is being torn down.
// which is being torned down.
return;
}
parsed_packet.IdentifyExtensions(it->second.extensions);