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Reland^2 "Reland: Remove unsupported configuration value, allow_codec_switching"

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This reverts commit 117d847901ea231cd86ca152b359b88619b9de20.

Reason for revert: Downstream error has been corrected.

Original change's description:
> Revert "Reland: Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
>
> Reason for revert: Breaks downstream features
>
> Original change's description:
> > Reland: Remove unsupported configuration value, `allow_codec_switching`
> >
> > This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
> >
> > Reason for revert: Relanding once downstream issues have been addressed
> >
> > Original change's description:
> > > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> > >
> > > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> > >
> > > Reason for revert: breaks downstream
> > >
> > > Original change's description:
> > > > Remove unsupported configuration value, `allow_codec_switching`
> > > >
> > > > Bug: webrtc:11341
> > > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#40995}
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40998}
> >
> > Bug: webrtc:11341
> > Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41032}
>
> Bug: webrtc:11341
> Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41161}

Bug: webrtc:11341
Change-Id: I4a5390a3b8c5e665b742fc564709847ad8853ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41213}
This commit is contained in:
Tomas Gunnarsson 2023-11-21 15:50:34 +00:00 committed by WebRTC LUCI CQ
parent 502afbf510
commit 3a15ba6fbf
6 changed files with 0 additions and 32 deletions
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4
api/peer_connection_interface.h
View File

@ -666,10 +666,6 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// Added to be able to control rollout of this feature.
bool enable_implicit_rollback = false;
// Whether network condition based codec switching is allowed.
// TODO(bugs.webrtc.org/11341): Remove this unsupported config value.
absl::optional<bool> allow_codec_switching;
// The delay before doing a usage histogram report for long-lived
// PeerConnections. Used for testing only.
absl::optional<int> report_usage_pattern_delay_ms;

3
pc/peer_connection.cc
View File

@ -329,7 +329,6 @@ RTCErrorOr<PeerConnectionInterface::RTCConfiguration> ApplyConfiguration(
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.turn_logging_id = configuration.turn_logging_id;
modified_config.allow_codec_switching = configuration.allow_codec_switching;
modified_config.stable_writable_connection_ping_interval_ms =
configuration.stable_writable_connection_ping_interval_ms;
if (configuration != modified_config) {
@ -458,7 +457,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
absl::optional<bool> allow_codec_switching;
absl::optional<int> report_usage_pattern_delay_ms;
absl::optional<int> stable_writable_connection_ping_interval_ms;
VpnPreference vpn_preference;
@ -522,7 +520,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
turn_logging_id == o.turn_logging_id &&
enable_implicit_rollback == o.enable_implicit_rollback &&
allow_codec_switching == o.allow_codec_switching &&
report_usage_pattern_delay_ms == o.report_usage_pattern_delay_ms &&
stable_writable_connection_ping_interval_ms ==
o.stable_writable_connection_ping_interval_ms &&

12
sdk/android/api/org/webrtc/PeerConnection.java
View File

@ -540,11 +540,6 @@ public class PeerConnection {
// every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
public boolean activeResetSrtpParams;
// Whether this client is allowed to switch encoding codec mid-stream. This is a workaround for
// a WebRTC bug where the receiver could get confussed if a codec switch happened mid-call.
// Null indicates no change to currently configured value.
@Nullable public Boolean allowCodecSwitching;
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
@ -611,7 +606,6 @@ public class PeerConnection {
activeResetSrtpParams = false;
cryptoOptions = null;
turnLoggingId = null;
allowCodecSwitching = null;
enableImplicitRollback = false;
offerExtmapAllowMixed = true;
}
@ -801,12 +795,6 @@ public class PeerConnection {
return activeResetSrtpParams;
}
@Nullable
@CalledByNative("RTCConfiguration")
Boolean getAllowCodecSwitching() {
return allowCodecSwitching;
}
@Nullable
@CalledByNative("RTCConfiguration")
CryptoOptions getCryptoOptions() {

4
sdk/android/src/jni/pc/peer_connection.cc
View File

@ -267,10 +267,6 @@ void JavaToNativeRTCConfiguration(
Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
rtc_config->crypto_options =
JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
rtc_config->allow_codec_switching = JavaToNativeOptionalBool(
jni, Java_RTCConfiguration_getAllowCodecSwitching(jni, j_rtc_config));
rtc_config->offer_extmap_allow_mixed =
Java_RTCConfiguration_getOfferExtmapAllowMixed(jni, j_rtc_config);
rtc_config->enable_implicit_rollback =

6
sdk/objc/api/peerconnection/RTCConfiguration.h
View File

@ -184,12 +184,6 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, assign) BOOL activeResetSrtpParams;
/** If the remote side support mid-stream codec switches then allow encoder
* switching to be performed.
*/
@property(nonatomic, assign) BOOL allowCodecSwitching;
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any

3
sdk/objc/api/peerconnection/RTCConfiguration.mm
View File

@ -51,7 +51,6 @@
@synthesize sdpSemantics = _sdpSemantics;
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize turnLoggingId = _turnLoggingId;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@ -139,7 +138,6 @@
_turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
_rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
_rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
_allowCodecSwitching = config.allow_codec_switching.value_or(false);
_enableImplicitRollback = config.enable_implicit_rollback;
_offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
_iceCheckIntervalStrongConnectivity =
@ -286,7 +284,6 @@
nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
nativeConfig->allow_codec_switching = _allowCodecSwitching;
nativeConfig->enable_implicit_rollback = _enableImplicitRollback;
nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
if (_iceCheckIntervalStrongConnectivity != nil) {