Move webrtc/audio_*.h to webrtc/api/call

webrtc/audio_send_stream.h

-/*
- *  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_SEND_STREAM_H_
-#define WEBRTC_AUDIO_SEND_STREAM_H_
-
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/transport.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// WORK IN PROGRESS
-// This class is under development and is not yet intended for for use outside
-// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
-
-class AudioSendStream {
- public:
-  struct Stats {
-    // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
-    uint32_t local_ssrc = 0;
-    int64_t bytes_sent = 0;
-    int32_t packets_sent = 0;
-    int32_t packets_lost = -1;
-    float fraction_lost = -1.0f;
-    std::string codec_name;
-    int32_t ext_seqnum = -1;
-    int32_t jitter_ms = -1;
-    int64_t rtt_ms = -1;
-    int32_t audio_level = -1;
-    float aec_quality_min = -1.0f;
-    int32_t echo_delay_median_ms = -1;
-    int32_t echo_delay_std_ms = -1;
-    int32_t echo_return_loss = -100;
-    int32_t echo_return_loss_enhancement = -100;
-    bool typing_noise_detected = false;
-  };
-
-  struct Config {
-    Config() = delete;
-    explicit Config(Transport* send_transport)
-        : send_transport(send_transport) {}
-
-    std::string ToString() const;
-
-    // Send-stream specific RTP settings.
-    struct Rtp {
-      std::string ToString() const;
-
-      // Sender SSRC.
-      uint32_t ssrc = 0;
-
-      // RTP header extensions used for the sent stream.
-      std::vector<RtpExtension> extensions;
-
-      // See NackConfig for description.
-      NackConfig nack;
-
-      // RTCP CNAME, see RFC 3550.
-      std::string c_name;
-    } rtp;
-
-    // Transport for outgoing packets. The transport is expected to exist for
-    // the entire life of the AudioSendStream and is owned by the API client.
-    Transport* send_transport = nullptr;
-
-    // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
-    // components.
-    // TODO(solenberg): Remove when VoiceEngine channels are created outside
-    // of Call.
-    int voe_channel_id = -1;
-
-    // Ownership of the encoder object is transferred to Call when the config is
-    // passed to Call::CreateAudioSendStream().
-    // TODO(solenberg): Implement, once we configure codecs through the new API.
-    // std::unique_ptr<AudioEncoder> encoder;
-    int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
-
-    // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
-    // disable audio bitrate adaptation.
-    // Note: This is still an experimental feature and not ready for real usage.
-    int min_bitrate_kbps = -1;
-    int max_bitrate_kbps = -1;
-  };
-
-  // Starts stream activity.
-  // When a stream is active, it can receive, process and deliver packets.
-  virtual void Start() = 0;
-  // Stops stream activity.
-  // When a stream is stopped, it can't receive, process or deliver packets.
-  virtual void Stop() = 0;
-
-  // TODO(solenberg): Make payload_type a config property instead.
-  virtual bool SendTelephoneEvent(int payload_type, int event,
-                                  int duration_ms) = 0;
-
-  virtual void SetMuted(bool muted) = 0;
-
-  virtual Stats GetStats() const = 0;
-
- protected:
-  virtual ~AudioSendStream() {}
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_SEND_STREAM_H_