text: /*
   *  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 <string>
  #include <vector>
  
  #include "webrtc/base/scoped_ptr.h"
  #include "webrtc/config.h"
  #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
  #include "webrtc/stream.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 SendStream {
   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;
  
      // Receive-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;
  
        // 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.
      // rtc::scoped_ptr<AudioEncoder> encoder;
      int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
      int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
    };
  
    // TODO(solenberg): Make payload_type a config property instead.
    virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
                                    uint32_t duration_ms) = 0;
    virtual Stats GetStats() const = 0;
  };
  }  // namespace webrtc
  
  #endif  // WEBRTC_AUDIO_SEND_STREAM_H_
  