b86d4e4a8d
This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
62 lines
1.9 KiB
C++
62 lines
1.9 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_SEND_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#include "webrtc/audio_send_stream.h"
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#include "webrtc/audio_state.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/base/scoped_ptr.h"
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namespace webrtc {
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class CongestionController;
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class VoiceEngine;
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namespace voe {
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class ChannelProxy;
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} // namespace voe
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namespace internal {
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class AudioSendStream final : public webrtc::AudioSendStream {
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public:
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AudioSendStream(const webrtc::AudioSendStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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CongestionController* congestion_controller);
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~AudioSendStream() override;
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// webrtc::SendStream 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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// webrtc::AudioSendStream implementation.
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bool SendTelephoneEvent(int payload_type, uint8_t event,
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uint32_t duration_ms) override;
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webrtc::AudioSendStream::Stats GetStats() const override;
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const webrtc::AudioSendStream::Config& config() const;
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private:
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VoiceEngine* voice_engine() const;
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rtc::ThreadChecker thread_checker_;
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const webrtc::AudioSendStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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