| /* |
| * 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 "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; |
| uint32_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; |
| 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; |
| } 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; |
| |
| // Access the currently set rtp extensions. Must be called on the packet |
| // delivery thread. |
| // TODO(tommi): This is currently only called from |
| // `WebRtcAudioReceiveStream::GetRtpParameters()`. See if we can remove it. |
| virtual const std::vector<RtpExtension>& GetRtpExtensions() const = 0; |
| |
| protected: |
| virtual ~AudioReceiveStreamInterface() {} |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_AUDIO_RECEIVE_STREAM_H_ |