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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_RECEIVE_STREAM_H_
#define CALL_AUDIO_RECEIVE_STREAM_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "call/receive_stream.h"
#include "call/rtp_config.h"
namespace webrtc {
class AudioSinkInterface;
class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
public:
struct Stats {
Stats();
~Stats();
uint32_t remote_ssrc = 0;
int64_t payload_bytes_rcvd = 0;
int64_t header_and_padding_bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
int32_t packets_lost = 0;
uint64_t packets_discarded = 0;
uint32_t nacks_sent = 0;
std::string codec_name;
absl::optional<int> codec_payload_type;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
uint32_t jitter_buffer_preferred_ms = 0;
uint32_t delay_estimate_ms = 0;
int32_t audio_level = -1;
// Stats below correspond to similarly-named fields in the WebRTC stats
// spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
double total_output_energy = 0.0;
uint64_t total_samples_received = 0;
double total_output_duration = 0.0;
uint64_t concealed_samples = 0;
uint64_t silent_concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
double jitter_buffer_target_delay_seconds = 0.0;
double jitter_buffer_minimum_delay_seconds = 0.0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;
float secondary_decoded_rate = 0.0f;
float secondary_discarded_rate = 0.0f;
float accelerate_rate = 0.0f;
float preemptive_expand_rate = 0.0f;
uint64_t delayed_packet_outage_samples = 0;
int32_t decoding_calls_to_silence_generator = 0;
int32_t decoding_calls_to_neteq = 0;
int32_t decoding_normal = 0;
// TODO(alexnarest): Consider decoding_neteq_plc for consistency
int32_t decoding_plc = 0;
int32_t decoding_codec_plc = 0;
int32_t decoding_cng = 0;
int32_t decoding_plc_cng = 0;
int32_t decoding_muted_output = 0;
int64_t capture_start_ntp_time_ms = 0;
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
absl::optional<int64_t> last_packet_received_timestamp_ms;
uint64_t jitter_buffer_flushes = 0;
double relative_packet_arrival_delay_seconds = 0.0;
int32_t interruption_count = 0;
int32_t total_interruption_duration_ms = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
absl::optional<int64_t> last_sender_report_timestamp_ms;
absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
uint32_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
absl::optional<TimeDelta> round_trip_time;
TimeDelta total_round_trip_time = TimeDelta::Zero();
int round_trip_time_measurements;
};
struct Config {
Config();
~Config();
std::string ToString() const;
// Receive-stream specific RTP settings.
struct Rtp : public ReceiveStreamRtpConfig {
Rtp();
~Rtp();
std::string ToString() const;
// See NackConfig for description.
NackConfig nack;
RtcpEventObserver* rtcp_event_observer = nullptr;
} rtp;
// Receive-side RTT.
bool enable_non_sender_rtt = false;
Transport* rtcp_send_transport = nullptr;
// NetEq settings.
size_t jitter_buffer_max_packets = 200;
bool jitter_buffer_fast_accelerate = false;
int jitter_buffer_min_delay_ms = 0;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video
// stream to one audio stream. Tracked by issue webrtc:4762.
std::string sync_group;
// Decoder specifications for every payload type that we can receive.
std::map<int, SdpAudioFormat> decoder_map;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
absl::optional<AudioCodecPairId> codec_pair_id;
// Per PeerConnection crypto options.
webrtc::CryptoOptions crypto_options;
// An optional custom frame decryptor that allows the entire frame to be
// decrypted in whatever way the caller choses. This is not required by
// default.
// TODO(tommi): Remove this member variable from the struct. It's not
// a part of the AudioReceiveStreamInterface state but rather a pass through
// variable.
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor;
// An optional frame transformer used by insertable streams to transform
// encoded frames.
// TODO(tommi): Remove this member variable from the struct. It's not
// a part of the AudioReceiveStreamInterface state but rather a pass through
// variable.
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
};
// Methods that support reconfiguring the stream post initialization.
virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0;
virtual void SetNackHistory(int history_ms) = 0;
virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
// Returns true if the stream has been started.
virtual bool IsRunning() const = 0;
virtual Stats GetStats(bool get_and_clear_legacy_stats) const = 0;
Stats GetStats() { return GetStats(/*get_and_clear_legacy_stats=*/true); }
// Sets an audio sink that receives unmixed audio from the receive stream.
// Ownership of the sink is managed by the caller.
// Only one sink can be set and passing a null sink clears an existing one.
// NOTE: Audio must still somehow be pulled through AudioTransport for audio
// to stream through this sink. In practice, this happens if mixed audio
// is being pulled+rendered and/or if audio is being pulled for the purposes
// of feeding to the AEC.
virtual void SetSink(AudioSinkInterface* sink) = 0;
// Sets playback gain of the stream, applied when mixing, and thus after it
// is potentially forwarded to any attached AudioSinkInterface implementation.
virtual void SetGain(float gain) = 0;
// Sets a base minimum for the playout delay. Base minimum delay sets lower
// bound on minimum delay value determining lower bound on playout delay.
//
// Returns true if value was successfully set, false overwise.
virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
// Synchronization source (stream identifier) to be received.
// This member will not change mid-stream and can be assumed to be const
// post initialization.
virtual uint32_t remote_ssrc() const = 0;
protected:
virtual ~AudioReceiveStreamInterface() {}
};
} // namespace webrtc
#endif // CALL_AUDIO_RECEIVE_STREAM_H_
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