diff options
| author | Tommi <tommi@webrtc.org> | 2020-04-27 12:01:11 +0200 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2020-04-27 13:59:45 +0000 |
| commit | 9abc6bd8aa85b23079221956463379d80d5426c2 (patch) | |
| tree | 8775010c0168fb77c480d11e6be827577ecc525d /audio | |
| parent | b37e59d198add002a0776b6a44603f9846f65f56 (diff) | |
| download | webrtc-9abc6bd8aa85b23079221956463379d80d5426c2.tar.gz | |
Reduce audiosendstream dependency on rttstats (dead code).
Change-Id: I4b05321548b6584424f23c45b0e95b4c03fe67c1
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148529
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31139}
Diffstat (limited to 'audio')
| -rw-r--r-- | audio/audio_send_stream.cc | 2 | ||||
| -rw-r--r-- | audio/audio_send_stream.h | 1 | ||||
| -rw-r--r-- | audio/audio_send_stream_unittest.cc | 4 |
3 files changed, 1 insertions, 6 deletions
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index a44b55f95f..36010d80c3 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -114,7 +114,6 @@ AudioSendStream::AudioSendStream( rtp_transport, bitrate_allocator, event_log, - rtcp_rtt_stats, suspended_rtp_state, voe::CreateChannelSend(clock, task_queue_factory, @@ -138,7 +137,6 @@ AudioSendStream::AudioSendStream( RtpTransportControllerSendInterface* rtp_transport, BitrateAllocatorInterface* bitrate_allocator, RtcEventLog* event_log, - RtcpRttStats* rtcp_rtt_stats, const absl::optional<RtpState>& suspended_rtp_state, std::unique_ptr<voe::ChannelSendInterface> channel_send) : clock_(clock), diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index 1dbb134921..909d4e9fcc 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -73,7 +73,6 @@ class AudioSendStream final : public webrtc::AudioSendStream, RtpTransportControllerSendInterface* rtp_transport, BitrateAllocatorInterface* bitrate_allocator, RtcEventLog* event_log, - RtcpRttStats* rtcp_rtt_stats, const absl::optional<RtpState>& suspended_rtp_state, std::unique_ptr<voe::ChannelSendInterface> channel_send); ~AudioSendStream() override; diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 8f33d29f70..ea753f7d0b 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -29,7 +29,6 @@ #include "modules/audio_processing/include/audio_processing_statistics.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h" -#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "rtc_base/task_queue_for_test.h" #include "system_wrappers/include/clock.h" @@ -192,7 +191,7 @@ struct ConfigHelper { new internal::AudioSendStream( Clock::GetRealTimeClock(), stream_config_, audio_state_, task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_, - &event_log_, &rtcp_rtt_stats_, absl::nullopt, + &event_log_, absl::nullopt, std::unique_ptr<voe::ChannelSendInterface>(channel_send_))); } @@ -331,7 +330,6 @@ struct ConfigHelper { ::testing::NiceMock<MockRtcEventLog> event_log_; ::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_; ::testing::NiceMock<MockRtpRtcp> rtp_rtcp_; - MockRtcpRttStats rtcp_rtt_stats_; ::testing::NiceMock<MockLimitObserver> limit_observer_; BitrateAllocator bitrate_allocator_; // |worker_queue| is defined last to ensure all pending tasks are cancelled |