webrtc_m130/webrtc/audio/audio_receive_stream.h
deadbeef 2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13 20:00:29 +00:00

68 lines
2.2 KiB
C++

/*
* 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_AUDIO_AUDIO_RECEIVE_STREAM_H_
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
class CongestionController;
class RemoteBitrateEstimator;
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioReceiveStream final : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
~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;
void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
const webrtc::AudioReceiveStream::Config& config() const;
private:
VoiceEngine* voice_engine() const;
rtc::ThreadChecker thread_checker_;
RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};
} // namespace internal
} // namespace webrtc
#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_