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audio/red: cleanup killswitch

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this has been enable by default since M96

BUG=webrtc:11640

Change-Id: I5d310d3929882007211eae12bc3ac1366107ca87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36123}
This commit is contained in:
Philipp Hancke 2022-03-03 12:09:16 +01:00 committed by WebRTC LUCI CQ
parent 45623a3c0f
commit ff05c5c805
3 changed files with 13 additions and 41 deletions
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37
media/engine/webrtc_voice_engine.cc
View File

@ -210,11 +210,6 @@ bool IsEnabled(const webrtc::WebRtcKeyValueConfig& config,
return absl::StartsWith(config.Lookup(trial), "Enabled");
}
bool IsDisabled(const webrtc::WebRtcKeyValueConfig& config,
absl::string_view trial) {
return absl::StartsWith(config.Lookup(trial), "Disabled");
}
struct AdaptivePtimeConfig {
bool enabled = false;
webrtc::DataRate min_payload_bitrate = webrtc::DataRate::KilobitsPerSec(16);
@ -311,8 +306,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
audio_mixer_(audio_mixer),
apm_(audio_processing),
audio_frame_processor_(audio_frame_processor),
audio_red_for_opus_enabled_(
!IsDisabled(trials, "WebRTC-Audio-Red-For-Opus")),
minimized_remsampling_on_mobile_trial_enabled_(
IsEnabled(trials, "WebRTC-Audio-MinimizeResamplingOnMobile")) {
// This may be called from any thread, so detach thread checkers.
@ -738,7 +731,7 @@ std::vector<AudioCodec> WebRtcVoiceEngine::CollectCodecs(
out.push_back(codec);
if (codec.name == kOpusCodecName && audio_red_for_opus_enabled_) {
if (codec.name == kOpusCodecName) {
std::string redFmtp =
rtc::ToString(codec.id) + "/" + rtc::ToString(codec.id);
map_format({kRedCodecName, 48000, 2, {{"", redFmtp}}}, &out);
@ -1311,9 +1304,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
engine_(engine),
call_(call),
audio_config_(config.audio),
crypto_options_(crypto_options),
audio_red_for_opus_enabled_(
!IsDisabled(call->trials(), "WebRTC-Audio-Red-For-Opus")) {
crypto_options_(crypto_options) {
network_thread_checker_.Detach();
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
@ -1574,7 +1565,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
}
auto format = AudioCodecToSdpAudioFormat(codec);
if (!IsCodec(codec, kCnCodecName) && !IsCodec(codec, kDtmfCodecName) &&
(!audio_red_for_opus_enabled_ || !IsCodec(codec, kRedCodecName)) &&
!IsCodec(codec, kRedCodecName) &&
!engine()->decoder_factory_->IsSupportedDecoder(format)) {
RTC_LOG(LS_ERROR) << "Unsupported codec: " << rtc::ToString(format);
return false;
@ -1760,19 +1751,17 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
}
}
if (audio_red_for_opus_enabled_) {
// Loop through the codecs to find the RED codec that matches opus
// with respect to clockrate and number of channels.
size_t red_codec_position = 0;
for (const AudioCodec& red_codec : codecs) {
if (red_codec_position < send_codec_position &&
IsCodec(red_codec, kRedCodecName) &&
CheckRedParameters(red_codec, *send_codec_spec)) {
send_codec_spec->red_payload_type = red_codec.id;
break;
}
red_codec_position++;
// Loop through the codecs to find the RED codec that matches opus
// with respect to clockrate and number of channels.
size_t red_codec_position = 0;
for (const AudioCodec& red_codec : codecs) {
if (red_codec_position < send_codec_position &&
IsCodec(red_codec, kRedCodecName) &&
CheckRedParameters(red_codec, *send_codec_spec)) {
send_codec_spec->red_payload_type = red_codec.id;
break;
}
red_codec_position++;
}
if (send_codec_spec_ != send_codec_spec) {

5
media/engine/webrtc_voice_engine.h
View File

@ -130,9 +130,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
int audio_jitter_buffer_min_delay_ms_ = 0;
bool audio_jitter_buffer_enable_rtx_handling_ = false;
// If this field is enabled, we will negotiate and use RFC 2198
// redundancy for opus audio.
const bool audio_red_for_opus_enabled_;
const bool minimized_remsampling_on_mobile_trial_enabled_;
};
@ -325,8 +322,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
// Unsignaled streams have an option to have a frame decryptor set on them.
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
unsignaled_frame_decryptor_;
const bool audio_red_for_opus_enabled_;
};
} // namespace cricket

12
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -1035,18 +1035,6 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) {
{112, {"red", 48000, 2, {{"", "111/111"}}}}})));
}
// Test that we disable Opus/Red with the kill switch.
TEST_P(WebRtcVoiceEngineTestFake, RecvRed) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-Red-For-Opus/Disabled/");
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kRed48000Codec);
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_P(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
EXPECT_TRUE(SetupSendStream());