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Make CreateRtcLogStreamConfig() return a std::unique_ptr

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This is done for two reasons:
1. This will be useful in an upcoming CL, when all logging will be done with one function that accepts a std::unique_ptr<RtcEvent>.
2. Ensure no unnecessary copies when the compiler fails to optimize.

BUG=webrtc:8111
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/3010273002
Cr-Commit-Position: refs/heads/master@{#19746}
This commit is contained in:
eladalon 2017-09-08 06:15:52 -07:00 committed by Commit Bot
parent f901661f30
commit 8ec568a9ff
1 changed files with 33 additions and 33 deletions
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66
webrtc/call/call.cc
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@ -93,59 +93,59 @@ const int* FindKeyByValue(const std::map<int, int>& m, int v) {
return nullptr;
}
rtclog::StreamConfig CreateRtcLogStreamConfig(
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const VideoReceiveStream::Config& config) {
rtclog::StreamConfig rtclog_config;
rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
rtclog_config.local_ssrc = config.rtp.local_ssrc;
rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc;
rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
rtclog_config.remb = config.rtp.remb;
rtclog_config.rtp_extensions = config.rtp.extensions;
auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
rtclog_config->local_ssrc = config.rtp.local_ssrc;
rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc;
rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
rtclog_config->remb = config.rtp.remb;
rtclog_config->rtp_extensions = config.rtp.extensions;
for (const auto& d : config.decoders) {
const int* search =
FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type);
rtclog_config.codecs.emplace_back(d.payload_name, d.payload_type,
rtclog_config->codecs.emplace_back(d.payload_name, d.payload_type,
search ? *search : 0);
}
return rtclog_config;
}
rtclog::StreamConfig CreateRtcLogStreamConfig(
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const VideoSendStream::Config& config,
size_t ssrc_index) {
rtclog::StreamConfig rtclog_config;
rtclog_config.local_ssrc = config.rtp.ssrcs[ssrc_index];
auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index];
if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
rtclog_config.rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
}
rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
rtclog_config.rtp_extensions = config.rtp.extensions;
rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
rtclog_config->rtp_extensions = config.rtp.extensions;
rtclog_config.codecs.emplace_back(config.encoder_settings.payload_name,
config.encoder_settings.payload_type,
config.rtp.rtx.payload_type);
rtclog_config->codecs.emplace_back(config.encoder_settings.payload_name,
config.encoder_settings.payload_type,
config.rtp.rtx.payload_type);
return rtclog_config;
}
rtclog::StreamConfig CreateRtcLogStreamConfig(
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const AudioReceiveStream::Config& config) {
rtclog::StreamConfig rtclog_config;
rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
rtclog_config.local_ssrc = config.rtp.local_ssrc;
rtclog_config.rtp_extensions = config.rtp.extensions;
auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
rtclog_config->local_ssrc = config.rtp.local_ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
return rtclog_config;
}
rtclog::StreamConfig CreateRtcLogStreamConfig(
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const AudioSendStream::Config& config) {
rtclog::StreamConfig rtclog_config;
rtclog_config.local_ssrc = config.rtp.ssrc;
rtclog_config.rtp_extensions = config.rtp.extensions;
auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
if (config.send_codec_spec) {
rtclog_config.codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
}
return rtclog_config;
}
@ -605,7 +605,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
event_log_->LogAudioSendStreamConfig(*CreateRtcLogStreamConfig(config));
rtc::Optional<RtpState> suspended_rtp_state;
{
@ -671,7 +671,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
event_log_->LogAudioReceiveStreamConfig(*CreateRtcLogStreamConfig(config));
AudioReceiveStream* receive_stream = new AudioReceiveStream(
&audio_receiver_controller_, transport_send_->packet_router(), config,
config_.audio_state, event_log_);
@ -732,7 +732,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
++ssrc_index) {
event_log_->LogVideoSendStreamConfig(
CreateRtcLogStreamConfig(config, ssrc_index));
*CreateRtcLogStreamConfig(config, ssrc_index));
}
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
@ -822,7 +822,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
}
receive_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
event_log_->LogVideoReceiveStreamConfig(CreateRtcLogStreamConfig(config));
event_log_->LogVideoReceiveStreamConfig(*CreateRtcLogStreamConfig(config));
return receive_stream;
}