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 CALL_AUDIO_RECEIVE_STREAM_H_ 12 #define CALL_AUDIO_RECEIVE_STREAM_H_ 13 14 #include <map> 15 #include <memory> 16 #include <string> 17 #include <vector> 18 19 #include "api/audio_codecs/audio_decoder_factory.h" 20 #include "api/call/transport.h" 21 #include "api/optional.h" 22 #include "api/rtpparameters.h" 23 #include "api/rtpreceiverinterface.h" 24 #include "call/rtp_config.h" 25 #include "common_types.h" // NOLINT(build/include) 26 #include "rtc_base/scoped_ref_ptr.h" 27 #include "typedefs.h" // NOLINT(build/include) 28 29 namespace webrtc { 30 class AudioSinkInterface; 31 32 // WORK IN PROGRESS 33 // This class is under development and is not yet intended for for use outside 34 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 35 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 36 37 class AudioReceiveStream { 38 public: 39 struct Stats { 40 uint32_t remote_ssrc = 0; 41 int64_t bytes_rcvd = 0; 42 uint32_t packets_rcvd = 0; 43 uint32_t packets_lost = 0; 44 float fraction_lost = 0.0f; 45 std::string codec_name; 46 rtc::Optional<int> codec_payload_type; 47 uint32_t ext_seqnum = 0; 48 uint32_t jitter_ms = 0; 49 uint32_t jitter_buffer_ms = 0; 50 uint32_t jitter_buffer_preferred_ms = 0; 51 uint32_t delay_estimate_ms = 0; 52 int32_t audio_level = -1; 53 // Stats below correspond to similarly-named fields in the WebRTC stats 54 // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats 55 double total_output_energy = 0.0; 56 uint64_t total_samples_received = 0; 57 double total_output_duration = 0.0; 58 uint64_t concealed_samples = 0; 59 uint64_t concealment_events = 0; 60 double jitter_buffer_delay_seconds = 0.0; 61 // Stats below DO NOT correspond directly to anything in the WebRTC stats 62 float expand_rate = 0.0f; 63 float speech_expand_rate = 0.0f; 64 float secondary_decoded_rate = 0.0f; 65 float secondary_discarded_rate = 0.0f; 66 float accelerate_rate = 0.0f; 67 float preemptive_expand_rate = 0.0f; 68 int32_t decoding_calls_to_silence_generator = 0; 69 int32_t decoding_calls_to_neteq = 0; 70 int32_t decoding_normal = 0; 71 int32_t decoding_plc = 0; 72 int32_t decoding_cng = 0; 73 int32_t decoding_plc_cng = 0; 74 int32_t decoding_muted_output = 0; 75 int64_t capture_start_ntp_time_ms = 0; 76 }; 77 78 struct Config { 79 std::string ToString() const; 80 81 // Receive-stream specific RTP settings. 82 struct Rtp { 83 std::string ToString() const; 84 85 // Synchronization source (stream identifier) to be received. 86 uint32_t remote_ssrc = 0; 87 88 // Sender SSRC used for sending RTCP (such as receiver reports). 89 uint32_t local_ssrc = 0; 90 91 // Enable feedback for send side bandwidth estimation. 92 // See 93 // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions 94 // for details. 95 bool transport_cc = false; 96 97 // See NackConfig for description. 98 NackConfig nack; 99 100 // RTP header extensions used for the received stream. 101 std::vector<RtpExtension> extensions; 102 } rtp; 103 104 Transport* rtcp_send_transport = nullptr; 105 106 // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- 107 // level components. 108 // TODO(solenberg): Remove when VoiceEngine channels are created outside 109 // of Call. 110 int voe_channel_id = -1; 111 112 // Identifier for an A/V synchronization group. Empty string to disable. 113 // TODO(pbos): Synchronize streams in a sync group, not just one video 114 // stream to one audio stream. Tracked by issue webrtc:4762. 115 std::string sync_group; 116 117 // Decoder specifications for every payload type that we can receive. 118 std::map<int, SdpAudioFormat> decoder_map; 119 120 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; 121 }; 122 123 // Starts stream activity. 124 // When a stream is active, it can receive, process and deliver packets. 125 virtual void Start() = 0; 126 // Stops stream activity. 127 // When a stream is stopped, it can't receive, process or deliver packets. 128 virtual void Stop() = 0; 129 130 virtual Stats GetStats() const = 0; 131 // TODO(solenberg): Remove, once AudioMonitor is gone. 132 virtual int GetOutputLevel() const = 0; 133 134 // Sets an audio sink that receives unmixed audio from the receive stream. 135 // Ownership of the sink is passed to the stream and can be used by the 136 // caller to do lifetime management (i.e. when the sink's dtor is called). 137 // Only one sink can be set and passing a null sink clears an existing one. 138 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 139 // to stream through this sink. In practice, this happens if mixed audio 140 // is being pulled+rendered and/or if audio is being pulled for the purposes 141 // of feeding to the AEC. 142 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; 143 144 // Sets playback gain of the stream, applied when mixing, and thus after it 145 // is potentially forwarded to any attached AudioSinkInterface implementation. 146 virtual void SetGain(float gain) = 0; 147 148 virtual std::vector<RtpSource> GetSources() const = 0; 149 150 protected: ~AudioReceiveStream()151 virtual ~AudioReceiveStream() {} 152 }; 153 } // namespace webrtc 154 155 #endif // CALL_AUDIO_RECEIVE_STREAM_H_ 156