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Optional: Use nullopt and implicit construction in /modules/audio_coding

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Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=kwiberg@webrtc.org

Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
This commit is contained in:
Oskar Sundbom
2017-11-16 15:31:38 +01:00
committed by Commit Bot
parent f715c53bca
commit 12ab00b4d8
49 changed files with 435 additions and 570 deletions
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38
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -108,9 +108,10 @@ float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
rtc::Optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
const std::string& param) {
auto it = format.parameters.find(param);
return (it == format.parameters.end())
? rtc::Optional<std::string>()
: rtc::Optional<std::string>(it->second);
if (it == format.parameters.end())
return rtc::nullopt;
return it->second;
}
template <typename T>
@ -255,9 +256,9 @@ rtc::Optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
return rtc::Optional<AudioCodecInfo>(info);
return info;
}
return rtc::Optional<AudioCodecInfo>();
return rtc::nullopt;
}
AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
@ -265,7 +266,7 @@ AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
config.bitrate_bps = codec_inst.rate;
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
@ -277,7 +278,7 @@ rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
return rtc::Optional<AudioEncoderOpusConfig>();
return rtc::nullopt;
}
AudioEncoderOpusConfig config;
@ -287,9 +288,9 @@ rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
config.bitrate_bps = rtc::Optional<int>(
config.bitrate_bps =
CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
GetFormatParameter(format, "maxaveragebitrate")));
GetFormatParameter(format, "maxaveragebitrate"));
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
@ -309,7 +310,7 @@ rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config.supported_frame_lengths_ms);
RTC_DCHECK(config.IsOk());
return rtc::Optional<AudioEncoderOpusConfig>(config);
return config;
}
rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
@ -321,11 +322,11 @@ rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
return rtc::Optional<int>();
return rtc::nullopt;
} else {
return rtc::Optional<int>(bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
: config.complexity);
return bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
: config.complexity;
}
}
@ -552,8 +553,7 @@ void AudioEncoderOpusImpl::OnReceivedOverhead(
audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
ApplyAudioNetworkAdaptor();
} else {
overhead_bytes_per_packet_ =
rtc::Optional<size_t>(overhead_bytes_per_packet);
overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
}
@ -707,9 +707,9 @@ void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
}
void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
config_.bitrate_bps = rtc::SafeClamp<int>(
bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps));
AudioEncoderOpusConfig::kMaxBitrateBps);
RTC_DCHECK(config_.IsOk());
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
const auto new_complexity = GetNewComplexity(config_);
@ -759,7 +759,7 @@ void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
bitrate_smoother_last_update_time_ = now_ms;
}
}
}