Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
      12             : #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
      13             : 
      14             : #include <memory>
      15             : #include <string>
      16             : #include <vector>
      17             : 
      18             : #include "webrtc/api/call/transport.h"
      19             : #include "webrtc/base/optional.h"
      20             : #include "webrtc/config.h"
      21             : #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
      22             : #include "webrtc/typedefs.h"
      23             : 
      24             : namespace webrtc {
      25             : 
      26             : // WORK IN PROGRESS
      27             : // This class is under development and is not yet intended for for use outside
      28             : // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
      29             : // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
      30             : 
      31           0 : class AudioSendStream {
      32             :  public:
      33           0 :   struct Stats {
      34             :     Stats();
      35             :     ~Stats();
      36             : 
      37             :     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
      38             :     uint32_t local_ssrc = 0;
      39             :     int64_t bytes_sent = 0;
      40             :     int32_t packets_sent = 0;
      41             :     int32_t packets_lost = -1;
      42             :     float fraction_lost = -1.0f;
      43             :     std::string codec_name;
      44             :     rtc::Optional<int> codec_payload_type;
      45             :     int32_t ext_seqnum = -1;
      46             :     int32_t jitter_ms = -1;
      47             :     int64_t rtt_ms = -1;
      48             :     int32_t audio_level = -1;
      49             :     float aec_quality_min = -1.0f;
      50             :     int32_t echo_delay_median_ms = -1;
      51             :     int32_t echo_delay_std_ms = -1;
      52             :     int32_t echo_return_loss = -100;
      53             :     int32_t echo_return_loss_enhancement = -100;
      54             :     float residual_echo_likelihood = -1.0f;
      55             :     float residual_echo_likelihood_recent_max = -1.0f;
      56             :     bool typing_noise_detected = false;
      57             :   };
      58             : 
      59           0 :   struct Config {
      60             :     Config() = delete;
      61             :     explicit Config(Transport* send_transport);
      62             :     ~Config();
      63             :     std::string ToString() const;
      64             : 
      65             :     // Send-stream specific RTP settings.
      66           0 :     struct Rtp {
      67             :       Rtp();
      68             :       ~Rtp();
      69             :       std::string ToString() const;
      70             : 
      71             :       // Sender SSRC.
      72             :       uint32_t ssrc = 0;
      73             : 
      74             :       // RTP header extensions used for the sent stream.
      75             :       std::vector<RtpExtension> extensions;
      76             : 
      77             :       // See NackConfig for description.
      78             :       NackConfig nack;
      79             : 
      80             :       // RTCP CNAME, see RFC 3550.
      81             :       std::string c_name;
      82             :     } rtp;
      83             : 
      84             :     // Transport for outgoing packets. The transport is expected to exist for
      85             :     // the entire life of the AudioSendStream and is owned by the API client.
      86             :     Transport* send_transport = nullptr;
      87             : 
      88             :     // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
      89             :     // components.
      90             :     // TODO(solenberg): Remove when VoiceEngine channels are created outside
      91             :     // of Call.
      92             :     int voe_channel_id = -1;
      93             : 
      94             :     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
      95             :     // disable audio bitrate adaptation.
      96             :     // Note: This is still an experimental feature and not ready for real usage.
      97             :     int min_bitrate_bps = -1;
      98             :     int max_bitrate_bps = -1;
      99             : 
     100             :     // Defines whether to turn on audio network adaptor, and defines its config
     101             :     // string.
     102             :     rtc::Optional<std::string> audio_network_adaptor_config;
     103             : 
     104             :     struct SendCodecSpec {
     105             :       SendCodecSpec();
     106             :       std::string ToString() const;
     107             : 
     108             :       bool operator==(const SendCodecSpec& rhs) const;
     109             :       bool operator!=(const SendCodecSpec& rhs) const {
     110             :         return !(*this == rhs);
     111             :       }
     112             : 
     113             :       bool nack_enabled = false;
     114             :       bool transport_cc_enabled = false;
     115             :       bool enable_codec_fec = false;
     116             :       bool enable_opus_dtx = false;
     117             :       int opus_max_playback_rate = 0;
     118             :       int cng_payload_type = -1;
     119             :       int cng_plfreq = -1;
     120             :       int max_ptime_ms = -1;
     121             :       int min_ptime_ms = -1;
     122             :       webrtc::CodecInst codec_inst;
     123             :     } send_codec_spec;
     124             :   };
     125             : 
     126             :   // Starts stream activity.
     127             :   // When a stream is active, it can receive, process and deliver packets.
     128             :   virtual void Start() = 0;
     129             :   // Stops stream activity.
     130             :   // When a stream is stopped, it can't receive, process or deliver packets.
     131             :   virtual void Stop() = 0;
     132             : 
     133             :   // TODO(solenberg): Make payload_type a config property instead.
     134             :   virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
     135             :                                   int event, int duration_ms) = 0;
     136             : 
     137             :   virtual void SetMuted(bool muted) = 0;
     138             : 
     139             :   virtual Stats GetStats() const = 0;
     140             : 
     141             :  protected:
     142           0 :   virtual ~AudioSendStream() {}
     143             : };
     144             : }  // namespace webrtc
     145             : 
     146             : #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_