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Clean up dead code in RtpSenderVideo.

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References to PlayoutDelayOracle and the deprecated RtpSenderVideo
constructor have been removed in downstream code, we can now clean the
unused code away.

Bug: webrtc:10809, webrtc:11340
Change-Id: I789274be2079ad4ddd7e83a5fa249b06a32a4e82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168400
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30499}
This commit is contained in:
Erik Språng 2020-02-09 18:07:56 +01:00 committed by Commit Bot
parent 03d909634b
commit d0e1885dbe
4 changed files with 0 additions and 57 deletions
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1
modules/rtp_rtcp/BUILD.gn
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@ -156,7 +156,6 @@ rtc_library("rtp_rtcp") {
"source/forward_error_correction_internal.h",
"source/packet_loss_stats.cc",
"source/packet_loss_stats.h",
"source/playout_delay_oracle.h",
"source/receive_statistics_impl.cc",
"source/receive_statistics_impl.h",
"source/remote_ntp_time_estimator.cc",

24
modules/rtp_rtcp/source/playout_delay_oracle.h
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@ -1,24 +0,0 @@
/*
* Copyright (c) 2016 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
#define MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
namespace webrtc {
// TODO(sprang): Remove once downstream usage is gone.
class PlayoutDelayOracle {
public:
PlayoutDelayOracle() = default;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_

20
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -246,26 +246,6 @@ bool IsNoopDelay(const PlayoutDelay& delay) {
} // namespace
RTPSenderVideo::RTPSenderVideo(Clock* clock,
RTPSender* rtp_sender,
FlexfecSender* flexfec_sender,
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool enable_retransmit_all_layers,
const WebRtcKeyValueConfig& field_trials)
: RTPSenderVideo([&] {
Config config;
config.clock = clock;
config.rtp_sender = rtp_sender;
config.flexfec_sender = flexfec_sender;
config.frame_encryptor = frame_encryptor;
config.require_frame_encryption = require_frame_encryption;
config.enable_retransmit_all_layers = enable_retransmit_all_layers;
config.field_trials = &field_trials;
return config;
}()) {}
RTPSenderVideo::RTPSenderVideo(const Config& config)
: rtp_sender_(config.rtp_sender),
clock_(config.clock),

12
modules/rtp_rtcp/source/rtp_sender_video.h
View File

@ -25,7 +25,6 @@
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
@ -69,8 +68,6 @@ class RTPSenderVideo {
Clock* clock = nullptr;
RTPSender* rtp_sender = nullptr;
FlexfecSender* flexfec_sender = nullptr;
// TODO(sprang): Remove when downstream usage is gone.
PlayoutDelayOracle* playout_delay_oracle = nullptr;
FrameEncryptorInterface* frame_encryptor = nullptr;
bool require_frame_encryption = false;
bool enable_retransmit_all_layers = false;
@ -81,15 +78,6 @@ class RTPSenderVideo {
explicit RTPSenderVideo(const Config& config);
// TODO(bugs.webrtc.org/10809): Remove when downstream usage is gone.
RTPSenderVideo(Clock* clock,
RTPSender* rtpSender,
FlexfecSender* flexfec_sender,
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool enable_retransmit_all_layers,
const WebRtcKeyValueConfig& field_trials);
virtual ~RTPSenderVideo();
// expected_retransmission_time_ms.has_value() -> retransmission allowed.