| Index: webrtc/video/audio_receive_stream.h
|
| diff --git a/webrtc/video/audio_receive_stream.h b/webrtc/video/audio_receive_stream.h
|
| deleted file mode 100644
|
| index c9ac04af397ade4edd719472686732f055d8a58d..0000000000000000000000000000000000000000
|
| --- a/webrtc/video/audio_receive_stream.h
|
| +++ /dev/null
|
| @@ -1,53 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#ifndef WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
|
| -#define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
|
| -
|
| -#include "webrtc/audio_receive_stream.h"
|
| -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -class RemoteBitrateEstimator;
|
| -
|
| -namespace internal {
|
| -
|
| -class AudioReceiveStream : public webrtc::AudioReceiveStream {
|
| - public:
|
| - AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
|
| - const webrtc::AudioReceiveStream::Config& config);
|
| - ~AudioReceiveStream() override {}
|
| -
|
| - // webrtc::ReceiveStream implementation.
|
| - void Start() override;
|
| - void Stop() override;
|
| - void SignalNetworkState(NetworkState state) override;
|
| - bool DeliverRtcp(const uint8_t* packet, size_t length) override;
|
| - bool DeliverRtp(const uint8_t* packet,
|
| - size_t length,
|
| - const PacketTime& packet_time) override;
|
| -
|
| - // webrtc::AudioReceiveStream implementation.
|
| - webrtc::AudioReceiveStream::Stats GetStats() const override;
|
| -
|
| - const webrtc::AudioReceiveStream::Config& config() const {
|
| - return config_;
|
| - }
|
| -
|
| - private:
|
| - RemoteBitrateEstimator* const remote_bitrate_estimator_;
|
| - const webrtc::AudioReceiveStream::Config config_;
|
| - rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
|
| -};
|
| -} // namespace internal
|
| -} // namespace webrtc
|
| -
|
| -#endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
|
|
|