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_RECEIVE_STREAM_H_ 12 #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ 13 14 #include <map> 15 #include <memory> 16 #include <string> 17 #include <vector> 18 19 #include "webrtc/api/call/transport.h" 20 #include "webrtc/base/optional.h" 21 #include "webrtc/base/scoped_ref_ptr.h" 22 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h" 23 #include "webrtc/common_types.h" 24 #include "webrtc/config.h" 25 #include "webrtc/typedefs.h" 26 27 namespace webrtc { 28 class AudioSinkInterface; 29 30 // WORK IN PROGRESS 31 // This class is under development and is not yet intended for for use outside 32 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 33 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 34 35 class AudioReceiveStream { 36 public: 37 struct Stats { 38 uint32_t remote_ssrc = 0; 39 int64_t bytes_rcvd = 0; 40 uint32_t packets_rcvd = 0; 41 uint32_t packets_lost = 0; 42 float fraction_lost = 0.0f; 43 std::string codec_name; 44 rtc::Optional<int> codec_payload_type; 45 uint32_t ext_seqnum = 0; 46 uint32_t jitter_ms = 0; 47 uint32_t jitter_buffer_ms = 0; 48 uint32_t jitter_buffer_preferred_ms = 0; 49 uint32_t delay_estimate_ms = 0; 50 int32_t audio_level = -1; 51 float expand_rate = 0.0f; 52 float speech_expand_rate = 0.0f; 53 float secondary_decoded_rate = 0.0f; 54 float accelerate_rate = 0.0f; 55 float preemptive_expand_rate = 0.0f; 56 int32_t decoding_calls_to_silence_generator = 0; 57 int32_t decoding_calls_to_neteq = 0; 58 int32_t decoding_normal = 0; 59 int32_t decoding_plc = 0; 60 int32_t decoding_cng = 0; 61 int32_t decoding_plc_cng = 0; 62 int32_t decoding_muted_output = 0; 63 int64_t capture_start_ntp_time_ms = 0; 64 }; 65 66 struct Config { 67 std::string ToString() const; 68 69 // Receive-stream specific RTP settings. 70 struct Rtp { 71 std::string ToString() const; 72 73 // Synchronization source (stream identifier) to be received. 74 uint32_t remote_ssrc = 0; 75 76 // Sender SSRC used for sending RTCP (such as receiver reports). 77 uint32_t local_ssrc = 0; 78 79 // Enable feedback for send side bandwidth estimation. 80 // See 81 // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions 82 // for details. 83 bool transport_cc = false; 84 85 // See NackConfig for description. 86 NackConfig nack; 87 88 // RTP header extensions used for the received stream. 89 std::vector<RtpExtension> extensions; 90 } rtp; 91 92 Transport* rtcp_send_transport = nullptr; 93 94 // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- 95 // level components. 96 // TODO(solenberg): Remove when VoiceEngine channels are created outside 97 // of Call. 98 int voe_channel_id = -1; 99 100 // Identifier for an A/V synchronization group. Empty string to disable. 101 // TODO(pbos): Synchronize streams in a sync group, not just one video 102 // stream to one audio stream. Tracked by issue webrtc:4762. 103 std::string sync_group; 104 105 // Decoders for every payload that we can receive. Call owns the 106 // AudioDecoder instances once the Config is submitted to 107 // Call::CreateReceiveStream(). 108 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. 109 std::map<uint8_t, AudioDecoder*> decoder_map; 110 111 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; 112 }; 113 114 // Starts stream activity. 115 // When a stream is active, it can receive, process and deliver packets. 116 virtual void Start() = 0; 117 // Stops stream activity. 118 // When a stream is stopped, it can't receive, process or deliver packets. 119 virtual void Stop() = 0; 120 121 virtual Stats GetStats() const = 0; 122 123 // Sets an audio sink that receives unmixed audio from the receive stream. 124 // Ownership of the sink is passed to the stream and can be used by the 125 // caller to do lifetime management (i.e. when the sink's dtor is called). 126 // Only one sink can be set and passing a null sink clears an existing one. 127 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 128 // to stream through this sink. In practice, this happens if mixed audio 129 // is being pulled+rendered and/or if audio is being pulled for the purposes 130 // of feeding to the AEC. 131 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; 132 133 // Sets playback gain of the stream, applied when mixing, and thus after it 134 // is potentially forwarded to any attached AudioSinkInterface implementation. 135 virtual void SetGain(float gain) = 0; 136 137 protected: ~AudioReceiveStream()138 virtual ~AudioReceiveStream() {} 139 }; 140 } // namespace webrtc 141 142 #endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ 143