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      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_AUDIO_RECEIVE_STREAM_H_
     12 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
     13 
     14 #include <map>
     15 #include <string>
     16 #include <vector>
     17 
     18 #include "webrtc/base/scoped_ptr.h"
     19 #include "webrtc/config.h"
     20 #include "webrtc/stream.h"
     21 #include "webrtc/transport.h"
     22 #include "webrtc/typedefs.h"
     23 
     24 namespace webrtc {
     25 
     26 class AudioDecoder;
     27 class AudioSinkInterface;
     28 
     29 // WORK IN PROGRESS
     30 // This class is under development and is not yet intended for for use outside
     31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
     32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
     33 
     34 class AudioReceiveStream : public ReceiveStream {
     35  public:
     36   struct Stats {
     37     uint32_t remote_ssrc = 0;
     38     int64_t bytes_rcvd = 0;
     39     uint32_t packets_rcvd = 0;
     40     uint32_t packets_lost = 0;
     41     float fraction_lost = 0.0f;
     42     std::string codec_name;
     43     uint32_t ext_seqnum = 0;
     44     uint32_t jitter_ms = 0;
     45     uint32_t jitter_buffer_ms = 0;
     46     uint32_t jitter_buffer_preferred_ms = 0;
     47     uint32_t delay_estimate_ms = 0;
     48     int32_t audio_level = -1;
     49     float expand_rate = 0.0f;
     50     float speech_expand_rate = 0.0f;
     51     float secondary_decoded_rate = 0.0f;
     52     float accelerate_rate = 0.0f;
     53     float preemptive_expand_rate = 0.0f;
     54     int32_t decoding_calls_to_silence_generator = 0;
     55     int32_t decoding_calls_to_neteq = 0;
     56     int32_t decoding_normal = 0;
     57     int32_t decoding_plc = 0;
     58     int32_t decoding_cng = 0;
     59     int32_t decoding_plc_cng = 0;
     60     int64_t capture_start_ntp_time_ms = 0;
     61   };
     62 
     63   struct Config {
     64     std::string ToString() const;
     65 
     66     // Receive-stream specific RTP settings.
     67     struct Rtp {
     68       std::string ToString() const;
     69 
     70       // Synchronization source (stream identifier) to be received.
     71       uint32_t remote_ssrc = 0;
     72 
     73       // Sender SSRC used for sending RTCP (such as receiver reports).
     74       uint32_t local_ssrc = 0;
     75 
     76       // Enable feedback for send side bandwidth estimation.
     77       // See
     78       // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
     79       // for details.
     80       bool transport_cc = false;
     81 
     82       // RTP header extensions used for the received stream.
     83       std::vector<RtpExtension> extensions;
     84     } rtp;
     85 
     86     Transport* receive_transport = nullptr;
     87     Transport* rtcp_send_transport = nullptr;
     88 
     89     // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
     90     // level components.
     91     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     92     // of Call.
     93     int voe_channel_id = -1;
     94 
     95     // Identifier for an A/V synchronization group. Empty string to disable.
     96     // TODO(pbos): Synchronize streams in a sync group, not just one video
     97     // stream to one audio stream. Tracked by issue webrtc:4762.
     98     std::string sync_group;
     99 
    100     // Decoders for every payload that we can receive. Call owns the
    101     // AudioDecoder instances once the Config is submitted to
    102     // Call::CreateReceiveStream().
    103     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
    104     std::map<uint8_t, AudioDecoder*> decoder_map;
    105 
    106     // TODO(pbos): Remove config option once combined A/V BWE is always on.
    107     bool combined_audio_video_bwe = false;
    108   };
    109 
    110   virtual Stats GetStats() const = 0;
    111 
    112   // Sets an audio sink that receives unmixed audio from the receive stream.
    113   // Ownership of the sink is passed to the stream and can be used by the
    114   // caller to do lifetime management (i.e. when the sink's dtor is called).
    115   // Only one sink can be set and passing a null sink, clears an existing one.
    116   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
    117   // to stream through this sink. In practice, this happens if mixed audio
    118   // is being pulled+rendered and/or if audio is being pulled for the purposes
    119   // of feeding to the AEC.
    120   virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
    121 };
    122 }  // namespace webrtc
    123 
    124 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_
    125