c7a8b08a7c
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
71 lines
2.2 KiB
C++
71 lines
2.2 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_SEND_STREAM_H_
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#define WEBRTC_AUDIO_SEND_STREAM_H_
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#include <string>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/config.h"
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/stream.h"
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#include "webrtc/transport.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioSendStream : public SendStream {
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public:
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struct Stats {};
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struct Config {
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Config() = delete;
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explicit Config(Transport* send_transport)
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: send_transport(send_transport) {}
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std::string ToString() const;
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// Receive-stream specific RTP settings.
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struct Rtp {
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std::string ToString() const;
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// Sender SSRC.
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uint32_t ssrc = 0;
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// RTP header extensions used for the received stream.
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std::vector<RtpExtension> extensions;
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} rtp;
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// Transport for outgoing packets. The transport is expected to exist for
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// the entire life of the AudioSendStream and is owned by the API client.
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Transport* send_transport = nullptr;
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// Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
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// components.
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// TODO(solenberg): Remove when VoiceEngine channels are created outside
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// of Call.
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int voe_channel_id = -1;
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// Ownership of the encoder object is transferred to Call when the config is
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// passed to Call::CreateAudioSendStream().
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// TODO(solenberg): Implement, once we configure codecs through the new API.
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// rtc::scoped_ptr<AudioEncoder> encoder;
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int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
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int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
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};
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virtual Stats GetStats() const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_SEND_STREAM_H_
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