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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 CALL_AUDIO_SEND_STREAM_H_
#define CALL_AUDIO_SEND_STREAM_H_
#include <memory>
#include <string>
#include <vector>
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/transport.h"
#include "api/optional.h"
#include "api/rtpparameters.h"
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "typedefs.h" // NOLINT(build/include)
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 {
Stats();
~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;
rtc::Optional<int> codec_payload_type;
int32_t ext_seqnum = -1;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
int32_t audio_level = -1;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
double total_input_energy = 0.0;
double total_input_duration = 0.0;
bool typing_noise_detected = false;
ANAStats ana_statistics;
AudioProcessingStats apm_statistics;
};
struct Config {
Config() = delete;
explicit Config(Transport* send_transport);
~Config();
std::string ToString() const;
// Send-stream specific RTP settings.
struct Rtp {
Rtp();
~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;
// 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_bps = -1;
int max_bitrate_bps = -1;
// Defines whether to turn on audio network adaptor, and defines its config
// string.
rtc::Optional<std::string> audio_network_adaptor_config;
struct SendCodecSpec {
SendCodecSpec(int payload_type, const SdpAudioFormat& format);
~SendCodecSpec();
std::string ToString() const;
bool operator==(const SendCodecSpec& rhs) const;
bool operator!=(const SendCodecSpec& rhs) const {
return !(*this == rhs);
}
int payload_type;
SdpAudioFormat format;
bool nack_enabled = false;
bool transport_cc_enabled = false;
rtc::Optional<int> cng_payload_type;
// If unset, use the encoder's default target bitrate.
rtc::Optional<int> target_bitrate_bps;
};
rtc::Optional<SendCodecSpec> send_codec_spec;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
// Track ID as specified during track creation.
std::string track_id;
};
virtual ~AudioSendStream() = default;
virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
// Reconfigure the stream according to the Configuration.
virtual void Reconfigure(const Config& config) = 0;
// 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 payload_frequency,
int event, int duration_ms) = 0;
virtual void SetMuted(bool muted) = 0;
virtual Stats GetStats() const = 0;
virtual Stats GetStats(bool has_remote_tracks) const = 0;
};
} // namespace webrtc
#endif // CALL_AUDIO_SEND_STREAM_H_
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