43e83d44f0
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.
Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1411083006
Cr-Commit-Position: refs/heads/master@{#10340}
52 lines
1.7 KiB
C++
52 lines
1.7 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include "webrtc/audio_receive_stream.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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namespace webrtc {
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class RemoteBitrateEstimator;
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namespace internal {
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class AudioReceiveStream : public webrtc::AudioReceiveStream {
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public:
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AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
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const webrtc::AudioReceiveStream::Config& config);
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~AudioReceiveStream() override;
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// webrtc::ReceiveStream implementation.
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void Start() override;
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void Stop() override;
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void SignalNetworkState(NetworkState state) override;
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bool DeliverRtcp(const uint8_t* packet, size_t length) override;
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bool DeliverRtp(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time) override;
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// webrtc::AudioReceiveStream implementation.
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webrtc::AudioReceiveStream::Stats GetStats() const override;
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const webrtc::AudioReceiveStream::Config& config() const;
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private:
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RemoteBitrateEstimator* const remote_bitrate_estimator_;
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const webrtc::AudioReceiveStream::Config config_;
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rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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