webrtc/api/call/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_API_CALL_AUDIO_SEND_STREAM_H_
#define WEBRTC_API_CALL_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;
    float residual_echo_likelihood = -1.0f;
    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;

    // 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;

    struct SendCodecSpec {
      SendCodecSpec() {
        webrtc::CodecInst empty_inst = {0};
        codec_inst = empty_inst;
        codec_inst.pltype = -1;
      }
      bool operator==(const SendCodecSpec& rhs) const {
        {
          if (nack_enabled != rhs.nack_enabled) {
            return false;
          }
          if (transport_cc_enabled != rhs.transport_cc_enabled) {
            return false;
          }
          if (enable_codec_fec != rhs.enable_codec_fec) {
            return false;
          }
          if (enable_opus_dtx != rhs.enable_opus_dtx) {
            return false;
          }
          if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
            return false;
          }
          if (cng_payload_type != rhs.cng_payload_type) {
            return false;
          }
          if (cng_plfreq != rhs.cng_plfreq) {
            return false;
          }
          if (codec_inst != rhs.codec_inst) {
            return false;
          }
          return true;
        }
      }
      bool operator!=(const SendCodecSpec& rhs) const {
        return !(*this == rhs);
      }

      bool nack_enabled = false;
      bool transport_cc_enabled = false;
      bool enable_codec_fec = false;
      bool enable_opus_dtx = false;
      int opus_max_playback_rate = 0;
      int cng_payload_type = -1;
      int cng_plfreq = -1;
      webrtc::CodecInst codec_inst;
    } send_codec_spec;
  };

  // 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_API_CALL_AUDIO_SEND_STREAM_H_