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Revert "Add ability to set ssrcs of RtpSender at construction time"

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This reverts commit e9d6e658c307fc0241d622756703d5c0d5388d80.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to set ssrcs of RtpSender at construction time
> 
> Bug: webrtc:10774
> Change-Id: I7147a75ccbcd1093dcd2e08047da8900843fdd8d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144037
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28447}

TBR=asapersson@webrtc.org,sprang@webrtc.org

Change-Id: I8b0cba0836e7d86ae1718055196c2c89860b97ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144368
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28453}
This commit is contained in:
Amit Hilbuch
2019-07-02 21:04:57 +00:00
committed by Commit Bot
parent c4421979fe
commit 02d7d353a9
7 changed files with 257 additions and 273 deletions
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10
modules/rtp_rtcp/include/rtp_rtcp.h
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@ -28,6 +28,7 @@
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/deprecation.h"
@ -40,7 +41,6 @@ class RateLimiter;
class ReceiveStatisticsProvider;
class RemoteBitrateEstimator;
class RtcEventLog;
class RTPSender;
class Transport;
class VideoBitrateAllocationObserver;
@ -52,7 +52,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
public:
struct Configuration {
Configuration();
Configuration(Configuration&& rhs);
// True for a audio version of the RTP/RTCP module object false will create
// a video version.
@ -121,11 +120,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
// defaults to webrtc::FieldTrialBasedConfig.
const WebRtcKeyValueConfig* field_trials = nullptr;
// SSRCs for sending media and retransmission, respectively.
// FlexFec SSRC is fetched from |flexfec_sender|.
absl::optional<uint32_t> media_send_ssrc;
absl::optional<uint32_t> rtx_send_ssrc;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};
@ -199,7 +193,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
uint32_t SSRC() const override = 0;
// Sets SSRC, default is a random number.
// TODO(bugs.webrtc.org/10774): Remove.
virtual void SetSSRC(uint32_t ssrc) = 0;
// Sets the value for sending in the RID (and Repaired) RTP header extension.
@ -227,7 +220,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
// Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
// only the SSRC is set.
// TODO(bugs.webrtc.org/10774): Remove.
virtual void SetRtxSsrc(uint32_t ssrc) = 0;
// Sets the payload type to use when sending RTX packets. Note that this