| OLD | NEW |
|---|
| (Empty) |
| 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_API_CALL_AUDIO_RECEIVE_STREAM_H_ | |
| 12 #define WEBRTC_API_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-extens
ions | |
| 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: | |
| 138 virtual ~AudioReceiveStream() {} | |
| 139 }; | |
| 140 } // namespace webrtc | |
| 141 | |
| 142 #endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2550273003:
Moved call.h and most of api/call/* into call/ (Closed)
