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_ |