diff options
Diffstat (limited to 'third_party/libwebrtc/media/engine/webrtc_voice_engine.cc')
| -rw-r--r-- | third_party/libwebrtc/media/engine/webrtc_voice_engine.cc | 29 |
1 files changed, 8 insertions, 21 deletions
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc index adf8b5c51d..adf662074d 100644 --- a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc +++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc @@ -66,7 +66,6 @@ #include "rtc_base/checks.h" #include "rtc_base/dscp.h" #include "rtc_base/experiments/struct_parameters_parser.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/logging.h" #include "rtc_base/race_checker.h" #include "rtc_base/string_encode.h" @@ -79,13 +78,12 @@ #include "system_wrappers/include/metrics.h" #if WEBRTC_ENABLE_PROTOBUF -RTC_PUSH_IGNORING_WUNDEF() #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h" #else #include "modules/audio_coding/audio_network_adaptor/config.pb.h" #endif -RTC_POP_IGNORING_WUNDEF() + #endif namespace cricket { @@ -147,12 +145,10 @@ bool IsCodec(const AudioCodec& codec, const char* ref_name) { return absl::EqualsIgnoreCase(codec.name, ref_name); } -absl::optional<AudioCodec> FindCodec( - const std::vector<AudioCodec>& codecs, - const AudioCodec& codec, - const webrtc::FieldTrialsView* field_trials) { +absl::optional<AudioCodec> FindCodec(const std::vector<AudioCodec>& codecs, + const AudioCodec& codec) { for (const AudioCodec& c : codecs) { - if (c.Matches(codec, field_trials)) { + if (c.Matches(codec)) { return c; } } @@ -344,10 +340,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine( const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing, - // TODO(bugs.webrtc.org/15111): - // Remove the raw AudioFrameProcessor pointer in the follow-up. - webrtc::AudioFrameProcessor* audio_frame_processor, - std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor, + std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor, const webrtc::FieldTrialsView& trials) : task_queue_factory_(task_queue_factory), adm_(adm), @@ -355,8 +348,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine( decoder_factory_(decoder_factory), audio_mixer_(audio_mixer), apm_(audio_processing), - audio_frame_processor_(audio_frame_processor), - owned_audio_frame_processor_(std::move(owned_audio_frame_processor)), + audio_frame_processor_(std::move(audio_frame_processor)), minimized_remsampling_on_mobile_trial_enabled_( IsEnabled(trials, "WebRTC-Audio-MinimizeResamplingOnMobile")) { RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; @@ -425,11 +417,7 @@ void WebRtcVoiceEngine::Init() { if (audio_frame_processor_) { config.async_audio_processing_factory = rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>( - *audio_frame_processor_, *task_queue_factory_); - } else if (owned_audio_frame_processor_) { - config.async_audio_processing_factory = - rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>( - std::move(owned_audio_frame_processor_), *task_queue_factory_); + std::move(audio_frame_processor_), *task_queue_factory_); } audio_state_ = webrtc::AudioState::Create(config); } @@ -2151,8 +2139,7 @@ bool WebRtcVoiceReceiveChannel::SetRecvCodecs( for (const AudioCodec& codec : codecs) { // Log a warning if a codec's payload type is changing. This used to be // treated as an error. It's abnormal, but not really illegal. - absl::optional<AudioCodec> old_codec = - FindCodec(recv_codecs_, codec, &call_->trials()); + absl::optional<AudioCodec> old_codec = FindCodec(recv_codecs_, codec); if (old_codec && old_codec->id != codec.id) { RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type (" << codec.id << ", was already mapped to " |