| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_AUDIO_SEND_STREAM_H_ |
| #define WEBRTC_AUDIO_SEND_STREAM_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/config.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
| #include "webrtc/transport.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // WORK IN PROGRESS |
| // This class is under development and is not yet intended for for use outside |
| // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
| |
| class AudioSendStream { |
| public: |
| struct Stats { |
| // TODO(solenberg): Harmonize naming and defaults with receive stream stats. |
| uint32_t local_ssrc = 0; |
| int64_t bytes_sent = 0; |
| int32_t packets_sent = 0; |
| int32_t packets_lost = -1; |
| float fraction_lost = -1.0f; |
| std::string codec_name; |
| int32_t ext_seqnum = -1; |
| int32_t jitter_ms = -1; |
| int64_t rtt_ms = -1; |
| int32_t audio_level = -1; |
| float aec_quality_min = -1.0f; |
| int32_t echo_delay_median_ms = -1; |
| int32_t echo_delay_std_ms = -1; |
| int32_t echo_return_loss = -100; |
| int32_t echo_return_loss_enhancement = -100; |
| bool typing_noise_detected = false; |
| }; |
| |
| struct Config { |
| Config() = delete; |
| explicit Config(Transport* send_transport) |
| : send_transport(send_transport) {} |
| |
| std::string ToString() const; |
| |
| // Send-stream specific RTP settings. |
| struct Rtp { |
| std::string ToString() const; |
| |
| // Sender SSRC. |
| uint32_t ssrc = 0; |
| |
| // RTP header extensions used for the sent stream. |
| std::vector<RtpExtension> extensions; |
| |
| // See NackConfig for description. |
| NackConfig nack; |
| |
| // RTCP CNAME, see RFC 3550. |
| std::string c_name; |
| } rtp; |
| |
| // Transport for outgoing packets. The transport is expected to exist for |
| // the entire life of the AudioSendStream and is owned by the API client. |
| Transport* send_transport = nullptr; |
| |
| // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level |
| // components. |
| // TODO(solenberg): Remove when VoiceEngine channels are created outside |
| // of Call. |
| int voe_channel_id = -1; |
| |
| // Ownership of the encoder object is transferred to Call when the config is |
| // passed to Call::CreateAudioSendStream(). |
| // TODO(solenberg): Implement, once we configure codecs through the new API. |
| // std::unique_ptr<AudioEncoder> encoder; |
| int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. |
| |
| // Bitrate limits used for variable audio bitrate streams. Set both to -1 to |
| // disable audio bitrate adaptation. |
| // Note: This is still an experimental feature and not ready for real usage. |
| int min_bitrate_kbps = -1; |
| int max_bitrate_kbps = -1; |
| }; |
| |
| // Starts stream activity. |
| // When a stream is active, it can receive, process and deliver packets. |
| virtual void Start() = 0; |
| // Stops stream activity. |
| // When a stream is stopped, it can't receive, process or deliver packets. |
| virtual void Stop() = 0; |
| |
| // TODO(solenberg): Make payload_type a config property instead. |
| virtual bool SendTelephoneEvent(int payload_type, int event, |
| int duration_ms) = 0; |
| |
| virtual void SetMuted(bool muted) = 0; |
| |
| virtual Stats GetStats() const = 0; |
| |
| protected: |
| virtual ~AudioSendStream() {} |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_SEND_STREAM_H_ |