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_AUDIO_SEND_STREAM_H_ 12 #define WEBRTC_AUDIO_SEND_STREAM_H_ 13 14 #include <string> 15 #include <vector> 16 17 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/config.h" 19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" 20 #include "webrtc/stream.h" 21 #include "webrtc/transport.h" 22 #include "webrtc/typedefs.h" 23 24 namespace webrtc { 25 26 // WORK IN PROGRESS 27 // This class is under development and is not yet intended for for use outside 28 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 29 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 30 31 class AudioSendStream : public SendStream { 32 public: 33 struct Stats { 34 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. 35 uint32_t local_ssrc = 0; 36 int64_t bytes_sent = 0; 37 int32_t packets_sent = 0; 38 int32_t packets_lost = -1; 39 float fraction_lost = -1.0f; 40 std::string codec_name; 41 int32_t ext_seqnum = -1; 42 int32_t jitter_ms = -1; 43 int64_t rtt_ms = -1; 44 int32_t audio_level = -1; 45 float aec_quality_min = -1.0f; 46 int32_t echo_delay_median_ms = -1; 47 int32_t echo_delay_std_ms = -1; 48 int32_t echo_return_loss = -100; 49 int32_t echo_return_loss_enhancement = -100; 50 bool typing_noise_detected = false; 51 }; 52 53 struct Config { 54 Config() = delete; 55 explicit Config(Transport* send_transport) 56 : send_transport(send_transport) {} 57 58 std::string ToString() const; 59 60 // Receive-stream specific RTP settings. 61 struct Rtp { 62 std::string ToString() const; 63 64 // Sender SSRC. 65 uint32_t ssrc = 0; 66 67 // RTP header extensions used for the sent stream. 68 std::vector<RtpExtension> extensions; 69 70 // RTCP CNAME, see RFC 3550. 71 std::string c_name; 72 } rtp; 73 74 // Transport for outgoing packets. The transport is expected to exist for 75 // the entire life of the AudioSendStream and is owned by the API client. 76 Transport* send_transport = nullptr; 77 78 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level 79 // components. 80 // TODO(solenberg): Remove when VoiceEngine channels are created outside 81 // of Call. 82 int voe_channel_id = -1; 83 84 // Ownership of the encoder object is transferred to Call when the config is 85 // passed to Call::CreateAudioSendStream(). 86 // TODO(solenberg): Implement, once we configure codecs through the new API. 87 // rtc::scoped_ptr<AudioEncoder> encoder; 88 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. 89 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. 90 }; 91 92 // TODO(solenberg): Make payload_type a config property instead. 93 virtual bool SendTelephoneEvent(int payload_type, uint8_t event, 94 uint32_t duration_ms) = 0; 95 virtual Stats GetStats() const = 0; 96 }; 97 } // namespace webrtc 98 99 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ 100