Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 11 | #ifndef CALL_AUDIO_RECEIVE_STREAM_H_ |
| 12 | #define CALL_AUDIO_RECEIVE_STREAM_H_ |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 13 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 14 | #include <map> |
kwiberg | fffa42b | 2016-02-23 10:46:32 -0800 | [diff] [blame] | 15 | #include <memory> |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 16 | #include <string> |
| 17 | #include <vector> |
| 18 | |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 19 | #include "absl/types/optional.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 20 | #include "api/audio_codecs/audio_decoder_factory.h" |
| 21 | #include "api/call/transport.h" |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 22 | #include "api/crypto/cryptooptions.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 23 | #include "api/rtpparameters.h" |
| 24 | #include "api/rtpreceiverinterface.h" |
| 25 | #include "call/rtp_config.h" |
Mirko Bonadei | 7120742 | 2017-09-15 13:58:09 +0200 | [diff] [blame] | 26 | #include "common_types.h" // NOLINT(build/include) |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 27 | #include "rtc_base/scoped_ref_ptr.h" |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 28 | |
| 29 | namespace webrtc { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 30 | class AudioSinkInterface; |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 31 | class FrameDecryptorInterface; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 32 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 33 | class AudioReceiveStream { |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 34 | public: |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 35 | struct Stats { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 36 | Stats(); |
| 37 | ~Stats(); |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 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; |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 44 | absl::optional<int> codec_payload_type; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 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; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 51 | // Stats below correspond to similarly-named fields in the WebRTC stats |
| 52 | // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 53 | double total_output_energy = 0.0; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 54 | uint64_t total_samples_received = 0; |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 55 | double total_output_duration = 0.0; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 56 | uint64_t concealed_samples = 0; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 57 | uint64_t concealment_events = 0; |
Gustaf Ullberg | b0a0207 | 2017-10-02 12:00:34 +0200 | [diff] [blame] | 58 | double jitter_buffer_delay_seconds = 0.0; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 59 | // Stats below DO NOT correspond directly to anything in the WebRTC stats |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 60 | float expand_rate = 0.0f; |
| 61 | float speech_expand_rate = 0.0f; |
| 62 | float secondary_decoded_rate = 0.0f; |
minyue-webrtc | 0e320ec | 2017-08-28 13:51:27 +0200 | [diff] [blame] | 63 | float secondary_discarded_rate = 0.0f; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 64 | float accelerate_rate = 0.0f; |
| 65 | float preemptive_expand_rate = 0.0f; |
| 66 | int32_t decoding_calls_to_silence_generator = 0; |
| 67 | int32_t decoding_calls_to_neteq = 0; |
| 68 | int32_t decoding_normal = 0; |
| 69 | int32_t decoding_plc = 0; |
| 70 | int32_t decoding_cng = 0; |
| 71 | int32_t decoding_plc_cng = 0; |
henrik.lundin | 6348978 | 2016-09-20 01:47:12 -0700 | [diff] [blame] | 72 | int32_t decoding_muted_output = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 73 | int64_t capture_start_ntp_time_ms = 0; |
| 74 | }; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 75 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 76 | struct Config { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 77 | Config(); |
| 78 | ~Config(); |
| 79 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 80 | std::string ToString() const; |
| 81 | |
| 82 | // Receive-stream specific RTP settings. |
| 83 | struct Rtp { |
Paulina Hensman | 11b34f4 | 2018-04-09 14:24:52 +0200 | [diff] [blame] | 84 | Rtp(); |
| 85 | ~Rtp(); |
| 86 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 87 | std::string ToString() const; |
| 88 | |
| 89 | // Synchronization source (stream identifier) to be received. |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 90 | uint32_t remote_ssrc = 0; |
| 91 | |
| 92 | // Sender SSRC used for sending RTCP (such as receiver reports). |
| 93 | uint32_t local_ssrc = 0; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 94 | |
Stefan Holmer | 3842c5c | 2016-01-12 13:55:00 +0100 | [diff] [blame] | 95 | // Enable feedback for send side bandwidth estimation. |
| 96 | // See |
| 97 | // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| 98 | // for details. |
| 99 | bool transport_cc = false; |
| 100 | |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 101 | // See NackConfig for description. |
| 102 | NackConfig nack; |
| 103 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 104 | // RTP header extensions used for the received stream. |
| 105 | std::vector<RtpExtension> extensions; |
| 106 | } rtp; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 107 | |
solenberg | cf18b34 | 2015-10-01 08:13:42 -0700 | [diff] [blame] | 108 | Transport* rtcp_send_transport = nullptr; |
| 109 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 110 | // NetEq settings. |
| 111 | size_t jitter_buffer_max_packets = 50; |
| 112 | bool jitter_buffer_fast_accelerate = false; |
| 113 | |
pbos | 8fc7fa7 | 2015-07-15 08:02:58 -0700 | [diff] [blame] | 114 | // Identifier for an A/V synchronization group. Empty string to disable. |
| 115 | // TODO(pbos): Synchronize streams in a sync group, not just one video |
| 116 | // stream to one audio stream. Tracked by issue webrtc:4762. |
| 117 | std::string sync_group; |
| 118 | |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 119 | // Decoder specifications for every payload type that we can receive. |
| 120 | std::map<int, SdpAudioFormat> decoder_map; |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 121 | |
| 122 | rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; |
Karl Wiberg | 0812634 | 2018-03-20 19:18:55 +0100 | [diff] [blame] | 123 | |
Danil Chapovalov | b9b146c | 2018-06-15 12:28:07 +0200 | [diff] [blame] | 124 | absl::optional<AudioCodecPairId> codec_pair_id; |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 125 | |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 126 | // Per PeerConnection crypto options. |
| 127 | webrtc::CryptoOptions crypto_options; |
| 128 | |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 129 | // An optional custom frame decryptor that allows the entire frame to be |
| 130 | // decrypted in whatever way the caller choses. This is not required by |
| 131 | // default. |
| 132 | rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 133 | }; |
| 134 | |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 135 | // Reconfigure the stream according to the Configuration. |
| 136 | virtual void Reconfigure(const Config& config) = 0; |
| 137 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 138 | // Starts stream activity. |
| 139 | // When a stream is active, it can receive, process and deliver packets. |
| 140 | virtual void Start() = 0; |
| 141 | // Stops stream activity. |
| 142 | // When a stream is stopped, it can't receive, process or deliver packets. |
| 143 | virtual void Stop() = 0; |
| 144 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 145 | virtual Stats GetStats() const = 0; |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 146 | |
| 147 | // Sets an audio sink that receives unmixed audio from the receive stream. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 148 | // Ownership of the sink is managed by the caller. |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 149 | // Only one sink can be set and passing a null sink clears an existing one. |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 150 | // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
| 151 | // to stream through this sink. In practice, this happens if mixed audio |
| 152 | // is being pulled+rendered and/or if audio is being pulled for the purposes |
| 153 | // of feeding to the AEC. |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 154 | virtual void SetSink(AudioSinkInterface* sink) = 0; |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 155 | |
solenberg | 217fb66 | 2016-06-17 08:30:54 -0700 | [diff] [blame] | 156 | // Sets playback gain of the stream, applied when mixing, and thus after it |
| 157 | // is potentially forwarded to any attached AudioSinkInterface implementation. |
| 158 | virtual void SetGain(float gain) = 0; |
| 159 | |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 160 | virtual std::vector<RtpSource> GetSources() const = 0; |
| 161 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 162 | protected: |
| 163 | virtual ~AudioReceiveStream() {} |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 164 | }; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 165 | } // namespace webrtc |
| 166 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 167 | #endif // CALL_AUDIO_RECEIVE_STREAM_H_ |