114 lines
3.5 KiB
C
114 lines
3.5 KiB
C
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/*
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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 <memory>
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#include <string>
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#include <vector>
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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/transport.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
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class AudioSendStream {
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public:
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struct Stats {
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// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
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uint32_t local_ssrc = 0;
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int64_t bytes_sent = 0;
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int32_t packets_sent = 0;
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int32_t packets_lost = -1;
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float fraction_lost = -1.0f;
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std::string codec_name;
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int32_t ext_seqnum = -1;
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int32_t jitter_ms = -1;
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int64_t rtt_ms = -1;
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int32_t audio_level = -1;
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float aec_quality_min = -1.0f;
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int32_t echo_delay_median_ms = -1;
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int32_t echo_delay_std_ms = -1;
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int32_t echo_return_loss = -100;
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int32_t echo_return_loss_enhancement = -100;
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bool typing_noise_detected = false;
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};
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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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// Send-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 sent stream.
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std::vector<RtpExtension> extensions;
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// See NackConfig for description.
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NackConfig nack;
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// RTCP CNAME, see RFC 3550.
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std::string c_name;
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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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// std::unique_ptr<AudioEncoder> encoder;
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int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
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};
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// Starts stream activity.
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// When a stream is active, it can receive, process and deliver packets.
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virtual void Start() = 0;
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// Stops stream activity.
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// When a stream is stopped, it can't receive, process or deliver packets.
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virtual void Stop() = 0;
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// TODO(solenberg): Make payload_type a config property instead.
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virtual bool SendTelephoneEvent(int payload_type, int event,
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int duration_ms) = 0;
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virtual void SetMuted(bool muted) = 0;
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virtual Stats GetStats() const = 0;
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protected:
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virtual ~AudioSendStream() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_SEND_STREAM_H_
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