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Diffstat (limited to 'third_party/libwebrtc/call/audio_receive_stream.h')
| -rw-r--r-- | third_party/libwebrtc/call/audio_receive_stream.h | 210 |
1 files changed, 210 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/audio_receive_stream.h b/third_party/libwebrtc/call/audio_receive_stream.h new file mode 100644 index 0000000000..6fc93b2d9a --- /dev/null +++ b/third_party/libwebrtc/call/audio_receive_stream.h @@ -0,0 +1,210 @@ +/* + * 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_ |