| /* |
| * Copyright (c) 2012 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 AUDIO_CHANNEL_RECEIVE_H_ |
| #define AUDIO_CHANNEL_RECEIVE_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/call/audio_sink.h" |
| #include "api/call/transport.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/media_transport_config.h" |
| #include "api/media_transport_interface.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "call/rtp_packet_sink_interface.h" |
| #include "call/syncable.h" |
| #include "modules/audio_coding/include/audio_coding_module.h" |
| |
| // TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence |
| // warnings about use of unsigned short. |
| // These need cleanup, in a separate cl. |
| |
| namespace rtc { |
| class TimestampWrapAroundHandler; |
| } |
| |
| namespace webrtc { |
| |
| class AudioDeviceModule; |
| class FrameDecryptorInterface; |
| class PacketRouter; |
| class ProcessThread; |
| class RateLimiter; |
| class ReceiveStatistics; |
| class RtcEventLog; |
| class RtpPacketReceived; |
| class RtpRtcp; |
| |
| struct CallReceiveStatistics { |
| unsigned int cumulativeLost; |
| unsigned int extendedMax; |
| unsigned int jitterSamples; |
| int64_t rttMs; |
| size_t bytesReceived; |
| int packetsReceived; |
| // The capture ntp time (in local timebase) of the first played out audio |
| // frame. |
| int64_t capture_start_ntp_time_ms_; |
| // 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; |
| }; |
| |
| namespace voe { |
| |
| class ChannelSendInterface; |
| |
| // Interface class needed for AudioReceiveStream tests that use a |
| // MockChannelReceive. |
| |
| class ChannelReceiveInterface : public RtpPacketSinkInterface { |
| public: |
| virtual ~ChannelReceiveInterface() = default; |
| |
| virtual void SetSink(AudioSinkInterface* sink) = 0; |
| |
| virtual void SetReceiveCodecs( |
| const std::map<int, SdpAudioFormat>& codecs) = 0; |
| |
| virtual void StartPlayout() = 0; |
| virtual void StopPlayout() = 0; |
| |
| // Payload type and format of last received RTP packet, if any. |
| virtual absl::optional<std::pair<int, SdpAudioFormat>> |
| GetReceiveCodec() const = 0; |
| |
| virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0; |
| |
| virtual void SetChannelOutputVolumeScaling(float scaling) = 0; |
| virtual int GetSpeechOutputLevelFullRange() const = 0; |
| // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy |
| virtual double GetTotalOutputEnergy() const = 0; |
| virtual double GetTotalOutputDuration() const = 0; |
| |
| // Stats. |
| virtual NetworkStatistics GetNetworkStatistics() const = 0; |
| virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0; |
| |
| // Audio+Video Sync. |
| virtual uint32_t GetDelayEstimate() const = 0; |
| virtual void SetMinimumPlayoutDelay(int delay_ms) = 0; |
| virtual uint32_t GetPlayoutTimestamp() const = 0; |
| |
| // Audio quality. |
| // Base minimum delay sets lower bound on minimum delay value which |
| // determines minimum delay until audio playout. |
| virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; |
| virtual int GetBaseMinimumPlayoutDelayMs() const = 0; |
| |
| // Produces the transport-related timestamps; current_delay_ms is left unset. |
| virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0; |
| |
| // RTP+RTCP |
| virtual void SetLocalSSRC(uint32_t ssrc) = 0; |
| |
| virtual void RegisterReceiverCongestionControlObjects( |
| PacketRouter* packet_router) = 0; |
| virtual void ResetReceiverCongestionControlObjects() = 0; |
| |
| virtual CallReceiveStatistics GetRTCPStatistics() const = 0; |
| virtual void SetNACKStatus(bool enable, int max_packets) = 0; |
| |
| virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
| int sample_rate_hz, |
| AudioFrame* audio_frame) = 0; |
| |
| virtual int PreferredSampleRate() const = 0; |
| |
| // Associate to a send channel. |
| // Used for obtaining RTT for a receive-only channel. |
| virtual void SetAssociatedSendChannel( |
| const ChannelSendInterface* channel) = 0; |
| |
| virtual std::vector<RtpSource> GetSources() const = 0; |
| }; |
| |
| std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive( |
| Clock* clock, |
| ProcessThread* module_process_thread, |
| AudioDeviceModule* audio_device_module, |
| const MediaTransportConfig& media_transport_config, |
| Transport* rtcp_send_transport, |
| RtcEventLog* rtc_event_log, |
| uint32_t remote_ssrc, |
| size_t jitter_buffer_max_packets, |
| bool jitter_buffer_fast_playout, |
| int jitter_buffer_min_delay_ms, |
| bool jitter_buffer_enable_rtx_handling, |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| const webrtc::CryptoOptions& crypto_options); |
| |
| } // namespace voe |
| } // namespace webrtc |
| |
| #endif // AUDIO_CHANNEL_RECEIVE_H_ |