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Remove unsupported configuration value, allow_codec_switching

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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}
This commit is contained in:
Tommi 2023-10-24 00:46:44 +02:00 committed by WebRTC LUCI CQ
parent 992d708e8e
commit 8f7a17f80f
6 changed files with 0 additions and 32 deletions
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4
api/peer_connection_interface.h
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@ -661,10 +661,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::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
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@ -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;
webrtc::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
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@ -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
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@ -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) {