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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 "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#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/crypto/crypto_options.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "call/audio_sender.h"
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
namespace webrtc {
class AudioSendStream : public AudioSender {
public:
struct Stats {
Stats();
~Stats();
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
int64_t payload_bytes_sent = 0;
int64_t header_and_padding_bytes_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent = 0;
int32_t packets_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
TimeDelta total_packet_send_delay = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent = 0;
int32_t packets_lost = -1;
float fraction_lost = -1.0f;
std::string codec_name;
absl::optional<int> codec_payload_type;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
int16_t audio_level = 0;
// 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;
ANAStats ana_statistics;
AudioProcessingStats apm_statistics;
RtcpPacketTypeCounter rtcp_packet_type_counts;
int64_t target_bitrate_bps = 0;
// A snapshot of Report Blocks with additional data of interest to
// statistics. Within this list, the sender-source SSRC pair is unique and
// per-pair the ReportBlockData represents the latest Report Block that was
// received for that pair.
std::vector<ReportBlockData> report_block_datas;
uint32_t nacks_received = 0;
};
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;
// The value to send in the RID RTP header extension if the extension is
// included in the list of extensions.
std::string rid;
// The value to send in the MID RTP header extension if the extension is
// included in the list of extensions.
std::string mid;
// Corresponds to the SDP attribute extmap-allow-mixed.
bool extmap_allow_mixed = false;
// RTP header extensions used for the sent stream.
std::vector<RtpExtension> extensions;
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
// Time interval between RTCP report for audio
int rtcp_report_interval_ms = 5000;
// 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;
// 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;
double bitrate_priority = 1.0;
bool has_dscp = false;
// Defines whether to turn on audio network adaptor, and defines its config
// string.
absl::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;
bool enable_non_sender_rtt = false;
absl::optional<int> cng_payload_type;
absl::optional<int> red_payload_type;
// If unset, use the encoder's default target bitrate.
absl::optional<int> target_bitrate_bps;
};
absl::optional<SendCodecSpec> send_codec_spec;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
absl::optional<AudioCodecPairId> codec_pair_id;
// Track ID as specified during track creation.
std::string track_id;
// Per PeerConnection crypto options.
webrtc::CryptoOptions crypto_options;
// An optional custom frame encryptor that allows the entire frame to be
// encryptor in whatever way the caller choses. This is not required by
// default.
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
// An optional frame transformer used by insertable streams to transform
// encoded frames.
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
};
virtual ~AudioSendStream() = default;
virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
// Reconfigure the stream according to the Configuration.
virtual void Reconfigure(const Config& config,
SetParametersCallback callback) = 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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