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/*
* 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 <string>
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/stream.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 SendStream {
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;
// Receive-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;
// 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.
// rtc::scoped_ptr<AudioEncoder> encoder;
int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
};
// TODO(solenberg): Make payload_type a config property instead.
virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) = 0;
virtual Stats GetStats() const = 0;
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_SEND_STREAM_H_
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