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Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling

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Bug: none
Change-Id: I14164546950cc63c37e54544cdc80bfd4eddf211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262962
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36955}
This commit is contained in:
Tommi 2022-05-20 16:17:49 +02:00 committed by WebRTC LUCI CQ
parent e508ebc645
commit 1def899931
11 changed files with 7 additions and 43 deletions
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6
api/audio_options.cc
View file

@ -52,8 +52,6 @@ void AudioOptions::SetAll(const AudioOptions& change) {
change.audio_jitter_buffer_fast_accelerate);
SetFrom(&audio_jitter_buffer_min_delay_ms,
change.audio_jitter_buffer_min_delay_ms);
SetFrom(&audio_jitter_buffer_enable_rtx_handling,
change.audio_jitter_buffer_enable_rtx_handling);
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
@ -74,8 +72,6 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
o.audio_jitter_buffer_fast_accelerate &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
audio_jitter_buffer_enable_rtx_handling ==
o.audio_jitter_buffer_enable_rtx_handling &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
@ -101,8 +97,6 @@ std::string AudioOptions::ToString() const {
audio_jitter_buffer_fast_accelerate);
ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
audio_jitter_buffer_min_delay_ms);
ToStringIfSet(&result, "audio_jitter_buffer_enable_rtx_handling",
audio_jitter_buffer_enable_rtx_handling);
ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);

2
api/audio_options.h
View file

@ -58,8 +58,6 @@ struct RTC_EXPORT AudioOptions {
absl::optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
absl::optional<int> audio_jitter_buffer_min_delay_ms;
// Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
// Enable combined audio+bandwidth BWE.
// TODO(pthatcher): This flag is set from the
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,

4
api/peer_connection_interface.h
View file

@ -491,10 +491,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// The minimum delay in milliseconds for the audio jitter buffer.
int audio_jitter_buffer_min_delay_ms = 0;
// Whether the audio jitter buffer adapts the delay to retransmitted
// packets.
bool audio_jitter_buffer_enable_rtx_handling = false;
// Timeout in milliseconds before an ICE candidate pair is considered to be
// "not receiving", after which a lower priority candidate pair may be
// selected.

7
audio/audio_receive_stream.cc
View file

@ -81,10 +81,9 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
config.rtcp_send_transport, event_log, config.rtp.local_ssrc,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.enable_non_sender_rtt,
config.decoder_factory, config.codec_pair_id,
std::move(config.frame_decryptor), config.crypto_options,
std::move(config.frame_transformer));
config.enable_non_sender_rtt, config.decoder_factory,
config.codec_pair_id, std::move(config.frame_decryptor),
config.crypto_options, std::move(config.frame_transformer));
}
} // namespace

8
audio/channel_receive.cc
View file

@ -94,7 +94,6 @@ class ChannelReceive : public ChannelReceiveInterface,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -523,7 +522,6 @@ ChannelReceive::ChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -1123,7 +1121,6 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -1134,9 +1131,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
clock, neteq_factory, audio_device_module, rtcp_send_transport,
rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets,
jitter_buffer_fast_playout, jitter_buffer_min_delay_ms,
jitter_buffer_enable_rtx_handling, enable_non_sender_rtt, decoder_factory,
codec_pair_id, std::move(frame_decryptor), crypto_options,
std::move(frame_transformer));
enable_non_sender_rtt, decoder_factory, codec_pair_id,
std::move(frame_decryptor), crypto_options, std::move(frame_transformer));
}
} // namespace voe

1
audio/channel_receive.h
View file

@ -182,7 +182,6 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,

1
call/audio_receive_stream.h
View file

@ -127,7 +127,6 @@ class AudioReceiveStream : public MediaReceiveStreamInterface {
size_t jitter_buffer_max_packets = 200;
bool jitter_buffer_fast_accelerate = false;
int jitter_buffer_min_delay_ms = 0;
bool jitter_buffer_enable_rtx_handling = false;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video

14
media/engine/webrtc_voice_engine.cc
View file

@ -260,7 +260,6 @@ webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_accelerate,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
@ -281,7 +280,6 @@ webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig(
config.jitter_buffer_max_packets = jitter_buffer_max_packets;
config.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
config.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms;
config.jitter_buffer_enable_rtx_handling = jitter_buffer_enable_rtx_handling;
config.frame_decryptor = std::move(frame_decryptor);
config.crypto_options = crypto_options;
config.frame_transformer = std::move(frame_transformer);
@ -400,7 +398,6 @@ void WebRtcVoiceEngine::Init() {
options.audio_jitter_buffer_max_packets = 200;
options.audio_jitter_buffer_fast_accelerate = false;
options.audio_jitter_buffer_min_delay_ms = 0;
options.audio_jitter_buffer_enable_rtx_handling = false;
bool error = ApplyOptions(options);
RTC_DCHECK(error);
}
@ -548,12 +545,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
audio_jitter_buffer_min_delay_ms_ =
*options.audio_jitter_buffer_min_delay_ms;
}
if (options.audio_jitter_buffer_enable_rtx_handling) {
RTC_LOG(LS_INFO) << "NetEq handle reordered packets? "
<< *options.audio_jitter_buffer_enable_rtx_handling;
audio_jitter_buffer_enable_rtx_handling_ =
*options.audio_jitter_buffer_enable_rtx_handling;
}
webrtc::AudioProcessing* ap = apm();
if (!ap) {
@ -1958,9 +1949,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
recv_rtp_extensions_, this, engine()->decoder_factory_, decoder_map_,
codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
engine()->audio_jitter_buffer_fast_accelerate_,
engine()->audio_jitter_buffer_min_delay_ms_,
engine()->audio_jitter_buffer_enable_rtx_handling_,
unsignaled_frame_decryptor_, crypto_options_, nullptr);
engine()->audio_jitter_buffer_min_delay_ms_, unsignaled_frame_decryptor_,
crypto_options_, nullptr);
recv_streams_.insert(std::make_pair(
ssrc, new WebRtcAudioReceiveStream(std::move(config), call_)));

1
media/engine/webrtc_voice_engine.h
View file

@ -128,7 +128,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
size_t audio_jitter_buffer_max_packets_ = 200;
bool audio_jitter_buffer_fast_accelerate_ = false;
int audio_jitter_buffer_min_delay_ms_ = 0;
bool audio_jitter_buffer_enable_rtx_handling_ = false;
const bool minimized_remsampling_on_mobile_trial_enabled_;
};

3
pc/peer_connection.cc
View file

@ -315,7 +315,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
int audio_jitter_buffer_min_delay_ms;
bool audio_jitter_buffer_enable_rtx_handling;
int ice_connection_receiving_timeout;
int ice_backup_candidate_pair_ping_interval;
ContinualGatheringPolicy continual_gathering_policy;
@ -362,8 +361,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
o.audio_jitter_buffer_fast_accelerate &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
audio_jitter_buffer_enable_rtx_handling ==
o.audio_jitter_buffer_enable_rtx_handling &&
ice_connection_receiving_timeout ==
o.ice_connection_receiving_timeout &&
ice_backup_candidate_pair_ping_interval ==

3
pc/sdp_offer_answer.cc
View file

@ -1211,9 +1211,6 @@ void SdpOfferAnswerHandler::Initialize(
audio_options_.audio_jitter_buffer_min_delay_ms =
configuration.audio_jitter_buffer_min_delay_ms;
audio_options_.audio_jitter_buffer_enable_rtx_handling =
configuration.audio_jitter_buffer_enable_rtx_handling;
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
if (!configuration.certificates.empty()) {