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 24namespace webrtc { 25 26class AudioDecoder; 27class 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 34class 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