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Remove unnecessary proxy+lock code around RtcpRttStats pointer

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Change-Id: I9c7fdc695be1e424488fa46962d459c66cf4d1e7
Bug: webrtc:9068
Reviewed-on: https://webrtc-review.googlesource.com/64721
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22603}
This commit is contained in:
Tommi 2018-03-26 13:28:26 +02:00 committed by Commit Bot
parent 1df1bf8551
commit 5f22365dd7
9 changed files with 18 additions and 60 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
audio/audio_receive_stream.cc
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@ -73,7 +73,7 @@ std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
return std::unique_ptr<voe::ChannelProxy>(
new voe::ChannelProxy(std::unique_ptr<voe::Channel>(new voe::Channel(
module_process_thread, internal_audio_state->audio_device_module(),
config.jitter_buffer_max_packets,
nullptr /* RtcpRttStats */, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.decoder_factory,
config.codec_pair_id))));
}

17
audio/audio_send_stream.cc
View File

@ -46,15 +46,15 @@ void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
webrtc::AudioState* audio_state,
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread) {
ProcessThread* module_process_thread,
RtcpRttStats* rtcp_rtt_stats) {
RTC_DCHECK(audio_state);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy(
std::unique_ptr<voe::Channel>(new voe::Channel(
worker_queue,
module_process_thread,
internal_audio_state->audio_device_module()))));
return std::unique_ptr<voe::ChannelProxy>(
new voe::ChannelProxy(std::unique_ptr<voe::Channel>(new voe::Channel(
worker_queue, module_process_thread,
internal_audio_state->audio_device_module(), rtcp_rtt_stats))));
}
} // namespace
@ -103,7 +103,8 @@ AudioSendStream::AudioSendStream(
overall_call_lifetime,
CreateChannelAndProxy(audio_state.get(),
worker_queue,
module_process_thread)) {}
module_process_thread,
rtcp_rtt_stats)) {}
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
@ -138,7 +139,6 @@ AudioSendStream::AudioSendStream(
RTC_DCHECK(overall_call_lifetime_);
channel_proxy_->SetRtcEventLog(event_log_);
channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
channel_proxy_->SetRTCPStatus(true);
RtpReceiver* rtpReceiver = nullptr; // Unused, but required for call.
channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
@ -159,7 +159,6 @@ AudioSendStream::~AudioSendStream() {
channel_proxy_->RegisterTransport(nullptr);
channel_proxy_->ResetSenderCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
channel_proxy_->SetRtcpRttStats(nullptr);
// Lifetime can only be updated after deregistering
// |timed_send_transport_adapter_| in the underlying channel object to avoid
// data races in |active_lifetime_|.

3
audio/audio_send_stream_unittest.cc
View File

@ -228,9 +228,6 @@ struct ConfigHelper {
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1); // Destructor resets the event log
EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull()))
.Times(1); // Destructor resets the rtt stats.
}
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {

40
audio/channel.cc
View File

@ -94,34 +94,6 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog {
RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
};
class RtcpRttStatsProxy final : public RtcpRttStats {
public:
RtcpRttStatsProxy() : rtcp_rtt_stats_(nullptr) {}
void OnRttUpdate(int64_t rtt) override {
rtc::CritScope lock(&crit_);
if (rtcp_rtt_stats_)
rtcp_rtt_stats_->OnRttUpdate(rtt);
}
int64_t LastProcessedRtt() const override {
rtc::CritScope lock(&crit_);
if (!rtcp_rtt_stats_)
return 0;
return rtcp_rtt_stats_->LastProcessedRtt();
}
void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
rtc::CritScope lock(&crit_);
rtcp_rtt_stats_ = rtcp_rtt_stats;
}
private:
rtc::CriticalSection crit_;
RtcpRttStats* rtcp_rtt_stats_ RTC_GUARDED_BY(crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy);
};
class TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
@ -539,9 +511,11 @@ int Channel::PreferredSampleRate() const {
Channel::Channel(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module)
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats)
: Channel(module_process_thread,
audio_device_module,
rtcp_rtt_stats,
0,
false,
rtc::scoped_refptr<AudioDecoderFactory>(),
@ -552,12 +526,12 @@ Channel::Channel(rtc::TaskQueue* encoder_queue,
Channel::Channel(ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
rtc::Optional<AudioCodecPairId> codec_pair_id)
: event_log_proxy_(new RtcEventLogProxy()),
rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()),
rtp_payload_registry_(new RTPPayloadRegistry()),
rtp_receive_statistics_(
ReceiveStatistics::Create(Clock::GetRealTimeClock())),
@ -620,7 +594,7 @@ Channel::Channel(ProcessThread* module_process_thread,
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
configuration.event_log = &(*event_log_proxy_);
configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_);
configuration.rtt_stats = rtcp_rtt_stats;
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
@ -1313,10 +1287,6 @@ void Channel::SetRtcEventLog(RtcEventLog* event_log) {
event_log_proxy_->SetEventLog(event_log);
}
void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
}
void Channel::UpdateOverheadForEncoder() {
size_t overhead_per_packet =
transport_overhead_per_packet_ + rtp_overhead_per_packet_;

7
audio/channel.h
View File

@ -90,7 +90,6 @@ struct ReportBlock {
namespace voe {
class RtcEventLogProxy;
class RtcpRttStatsProxy;
class RtpPacketSenderProxy;
class TransportFeedbackProxy;
class TransportSequenceNumberProxy;
@ -151,10 +150,12 @@ class Channel
// Used for send streams.
Channel(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module);
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats);
// Used for receive streams.
Channel(ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
RtcpRttStats* rtcp_rtt_stats,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
@ -285,7 +286,6 @@ class Channel
// Set a RtcEventLog logging object.
void SetRtcEventLog(RtcEventLog* event_log);
void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
void SetTransportOverhead(size_t transport_overhead_per_packet);
// From OverheadObserver in the RTP/RTCP module
@ -341,7 +341,6 @@ class Channel
ChannelState channel_state_;
std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_;
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;

5
audio/channel_proxy.cc
View File

@ -283,11 +283,6 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
}
}
void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel_->SetRtcpRttStats(rtcp_rtt_stats);
}
bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel_->GetRecCodec(*codec_inst) == 0;

1
audio/channel_proxy.h
View File

@ -110,7 +110,6 @@ class ChannelProxy : public RtpPacketSinkInterface {
RtpReceiver** rtp_receiver) const;
virtual uint32_t GetPlayoutTimestamp() const;
virtual void SetMinimumPlayoutDelay(int delay_ms);
virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
virtual bool GetRecCodec(CodecInst* codec_inst) const;
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
virtual void OnRecoverableUplinkPacketLossRate(

1
audio/mock_voe_channel_proxy.h
View File

@ -69,7 +69,6 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
MOCK_METHOD2(GetAudioFrameWithInfo,
AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
AudioFrame* audio_frame));

2
modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -343,7 +343,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
RemoteBitrateEstimator* remote_bitrate_;
RtcpRttStats* rtt_stats_;
RtcpRttStats* const rtt_stats_;
PacketLossStats send_loss_stats_;
PacketLossStats receive_loss_stats_;