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 | |
kjellander | a69d973 | 2016-08-31 07:33:05 -0700 | [diff] [blame] | 11 | #ifndef WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ |
| 12 | #define WEBRTC_API_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 | |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 19 | #include "webrtc/base/scoped_ref_ptr.h" |
| 20 | #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h" |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 21 | #include "webrtc/common_types.h" |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 22 | #include "webrtc/config.h" |
solenberg | cf18b34 | 2015-10-01 08:13:42 -0700 | [diff] [blame] | 23 | #include "webrtc/transport.h" |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 24 | #include "webrtc/typedefs.h" |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 25 | |
| 26 | namespace webrtc { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 27 | class AudioSinkInterface; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 28 | |
Fredrik Solenberg | a4527c8 | 2015-12-03 13:06:20 +0100 | [diff] [blame] | 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 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 34 | class AudioReceiveStream { |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 35 | public: |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 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; |
henrik.lundin | 6348978 | 2016-09-20 01:47:12 -0700 | [diff] [blame^] | 60 | int32_t decoding_muted_output = 0; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 61 | int64_t capture_start_ntp_time_ms = 0; |
| 62 | }; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 63 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 64 | struct Config { |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 65 | std::string ToString() const; |
| 66 | |
| 67 | // Receive-stream specific RTP settings. |
| 68 | struct Rtp { |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 69 | std::string ToString() const; |
| 70 | |
| 71 | // Synchronization source (stream identifier) to be received. |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 72 | uint32_t remote_ssrc = 0; |
| 73 | |
| 74 | // Sender SSRC used for sending RTCP (such as receiver reports). |
| 75 | uint32_t local_ssrc = 0; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 76 | |
Stefan Holmer | 3842c5c | 2016-01-12 13:55:00 +0100 | [diff] [blame] | 77 | // Enable feedback for send side bandwidth estimation. |
| 78 | // See |
| 79 | // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| 80 | // for details. |
| 81 | bool transport_cc = false; |
| 82 | |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 83 | // See NackConfig for description. |
| 84 | NackConfig nack; |
| 85 | |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 86 | // RTP header extensions used for the received stream. |
| 87 | std::vector<RtpExtension> extensions; |
| 88 | } rtp; |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 89 | |
solenberg | cf18b34 | 2015-10-01 08:13:42 -0700 | [diff] [blame] | 90 | Transport* rtcp_send_transport = nullptr; |
| 91 | |
| 92 | // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- |
| 93 | // level components. |
| 94 | // TODO(solenberg): Remove when VoiceEngine channels are created outside |
| 95 | // of Call. |
pbos | 8fc7fa7 | 2015-07-15 08:02:58 -0700 | [diff] [blame] | 96 | int voe_channel_id = -1; |
| 97 | |
| 98 | // Identifier for an A/V synchronization group. Empty string to disable. |
| 99 | // TODO(pbos): Synchronize streams in a sync group, not just one video |
| 100 | // stream to one audio stream. Tracked by issue webrtc:4762. |
| 101 | std::string sync_group; |
| 102 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 103 | // Decoders for every payload that we can receive. Call owns the |
| 104 | // AudioDecoder instances once the Config is submitted to |
| 105 | // Call::CreateReceiveStream(). |
| 106 | // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. |
| 107 | std::map<uint8_t, AudioDecoder*> decoder_map; |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 108 | |
| 109 | rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 110 | }; |
| 111 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 112 | // Starts stream activity. |
| 113 | // When a stream is active, it can receive, process and deliver packets. |
| 114 | virtual void Start() = 0; |
| 115 | // Stops stream activity. |
| 116 | // When a stream is stopped, it can't receive, process or deliver packets. |
| 117 | virtual void Stop() = 0; |
| 118 | |
Fredrik Solenberg | 04f4931 | 2015-06-08 13:04:56 +0200 | [diff] [blame] | 119 | virtual Stats GetStats() const = 0; |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 120 | |
| 121 | // Sets an audio sink that receives unmixed audio from the receive stream. |
| 122 | // Ownership of the sink is passed to the stream and can be used by the |
| 123 | // caller to do lifetime management (i.e. when the sink's dtor is called). |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 124 | // 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] | 125 | // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
| 126 | // to stream through this sink. In practice, this happens if mixed audio |
| 127 | // is being pulled+rendered and/or if audio is being pulled for the purposes |
| 128 | // of feeding to the AEC. |
kwiberg | fffa42b | 2016-02-23 10:46:32 -0800 | [diff] [blame] | 129 | virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 130 | |
solenberg | 217fb66 | 2016-06-17 08:30:54 -0700 | [diff] [blame] | 131 | // Sets playback gain of the stream, applied when mixing, and thus after it |
| 132 | // is potentially forwarded to any attached AudioSinkInterface implementation. |
| 133 | virtual void SetGain(float gain) = 0; |
| 134 | |
pbos | 1ba8d39 | 2016-05-01 20:18:34 -0700 | [diff] [blame] | 135 | protected: |
| 136 | virtual ~AudioReceiveStream() {} |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 137 | }; |
Fredrik Solenberg | 23fba1f | 2015-04-29 15:24:01 +0200 | [diff] [blame] | 138 | } // namespace webrtc |
| 139 | |
kjellander | a69d973 | 2016-08-31 07:33:05 -0700 | [diff] [blame] | 140 | #endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ |