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Don't use rvalue reference function arguments in the audio coding module

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Rvalue reference arguments are generally banned by the style guide.

Bug: None
Change-Id: I4314859623ffcf056f53c42087b59696b5e71690
Reviewed-on: https://chromium-review.googlesource.com/531028
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19338}
This commit is contained in:
minyue-webrtc 2017-08-14 14:33:32 +02:00 committed by Commit Bot
parent c5d9e63c2b
commit 5d6891000f
4 changed files with 25 additions and 22 deletions
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2
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
View File

@ -302,7 +302,7 @@ ControllerManagerImpl::ControllerManagerImpl(const Config& config)
ControllerManagerImpl::ControllerManagerImpl(
const Config& config,
std::vector<std::unique_ptr<Controller>>&& controllers,
std::vector<std::unique_ptr<Controller>> controllers,
const std::map<const Controller*, std::pair<int, float>>& scoring_points)
: config_(config),
controllers_(std::move(controllers)),

2
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
View File

@ -74,7 +74,7 @@ class ControllerManagerImpl final : public ControllerManager {
// Dependency injection for testing.
ControllerManagerImpl(
const Config& config,
std::vector<std::unique_ptr<Controller>>&& controllers,
std::vector<std::unique_ptr<Controller>> controllers,
const std::map<const Controller*, std::pair<int, float>>&
chracteristic_points);

34
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -368,31 +368,31 @@ class AudioEncoderOpus::PacketLossFractionSmoother {
AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config)
: AudioEncoderOpus(config, config.payload_type) {}
AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config,
int payload_type)
: AudioEncoderOpus(
config,
payload_type,
[this](const ProtoString& config_string, RtcEventLog* event_log) {
return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
},
// We choose 5sec as initial time constant due to empirical data.
rtc::MakeUnique<SmoothingFilterImpl>(5000)) {}
AudioEncoderOpus::AudioEncoderOpus(
const AudioEncoderOpusConfig& config,
int payload_type,
AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother)
: payload_type_(payload_type),
send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
"WebRTC-SendSideBwe-WithOverhead")),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
packet_loss_rate_(0.0),
inst_(nullptr),
packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
audio_network_adaptor_creator_(
audio_network_adaptor_creator
? std::move(audio_network_adaptor_creator)
: [this](const ProtoString& config_string,
RtcEventLog* event_log) {
return DefaultAudioNetworkAdaptorCreator(config_string,
event_log);
}),
bitrate_smoother_(bitrate_smoother
? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
// We choose 5sec as initial time constant due to empirical data.
new SmoothingFilterImpl(5000))) {
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
audio_network_adaptor_creator_(audio_network_adaptor_creator),
bitrate_smoother_(std::move(bitrate_smoother)) {
RTC_DCHECK(0 <= payload_type && payload_type <= 127);
// Sanity check of the redundant payload type field that we want to get rid
// of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847

9
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -67,11 +67,14 @@ class AudioEncoderOpus final : public AudioEncoder {
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
AudioEncoderOpus(const AudioEncoderOpusConfig& config);
AudioEncoderOpus(const AudioEncoderOpusConfig& config, int payload_type);
// Dependency injection for testing.
AudioEncoderOpus(
const AudioEncoderOpusConfig& config,
int payload_type,
AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother);
explicit AudioEncoderOpus(const CodecInst& codec_inst);
AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
@ -164,7 +167,7 @@ class AudioEncoderOpus final : public AudioEncoder {
int next_frame_length_ms_;
int complexity_;
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
rtc::Optional<size_t> overhead_bytes_per_packet_;
const std::unique_ptr<SmoothingFilter> bitrate_smoother_;