From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/audio/channel_receive.h | 196 ++++++++++++++++++++++++++ 1 file changed, 196 insertions(+) create mode 100644 third_party/libwebrtc/audio/channel_receive.h (limited to 'third_party/libwebrtc/audio/channel_receive.h') diff --git a/third_party/libwebrtc/audio/channel_receive.h b/third_party/libwebrtc/audio/channel_receive.h new file mode 100644 index 0000000000..5713d97aaa --- /dev/null +++ b/third_party/libwebrtc/audio/channel_receive.h @@ -0,0 +1,196 @@ +/* + * 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 +#include +#include +#include + +#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/frame_transformer_interface.h" +#include "api/neteq/neteq_factory.h" +#include "api/transport/rtp/rtp_source.h" +#include "call/rtp_packet_sink_interface.h" +#include "call/syncable.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/source_tracker.h" +#include "system_wrappers/include/clock.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 RateLimiter; +class ReceiveStatistics; +class RtcEventLog; +class RtpPacketReceived; +class RtpRtcp; + +struct CallReceiveStatistics { + int cumulativeLost; + unsigned int jitterSamples; + int64_t payload_bytes_received = 0; + int64_t header_and_padding_bytes_received = 0; + int packetsReceived; + uint32_t nacks_sent = 0; + // 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 last_packet_received; + // Remote outbound stats derived by the received RTCP sender reports. + // Note that the timestamps below correspond to the time elapsed since the + // Unix epoch. + // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* + absl::optional last_sender_report_timestamp_ms; + absl::optional last_sender_report_remote_timestamp_ms; + uint64_t sender_reports_packets_sent = 0; + uint64_t sender_reports_bytes_sent = 0; + uint64_t sender_reports_reports_count = 0; + absl::optional round_trip_time; + TimeDelta total_round_trip_time = TimeDelta::Zero(); + int round_trip_time_measurements; +}; + +namespace voe { + +class ChannelSendInterface; + +// Interface class needed for AudioReceiveStreamInterface 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& codecs) = 0; + + virtual void StartPlayout() = 0; + virtual void StopPlayout() = 0; + + // Payload type and format of last received RTP packet, if any. + virtual absl::optional> 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( + bool get_and_clear_legacy_stats) const = 0; + virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0; + + // Audio+Video Sync. + virtual uint32_t GetDelayEstimate() const = 0; + virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0; + virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, + int64_t* time_ms) const = 0; + virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms, + int64_t time_ms) = 0; + virtual absl::optional GetCurrentEstimatedPlayoutNtpTimestampMs( + int64_t now_ms) 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 GetSyncInfo() const = 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 void SetNonSenderRttMeasurement(bool enabled) = 0; + + virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( + int sample_rate_hz, + AudioFrame* audio_frame) = 0; + + virtual int PreferredSampleRate() const = 0; + + // Sets the source tracker to notify about "delivered" packets when output is + // muted. + virtual void SetSourceTracker(SourceTracker* source_tracker) = 0; + + // Associate to a send channel. + // Used for obtaining RTT for a receive-only channel. + virtual void SetAssociatedSendChannel( + const ChannelSendInterface* channel) = 0; + + // Sets a frame transformer between the depacketizer and the decoder, to + // transform the received frames before decoding them. + virtual void SetDepacketizerToDecoderFrameTransformer( + rtc::scoped_refptr + frame_transformer) = 0; + + virtual void SetFrameDecryptor( + rtc::scoped_refptr frame_decryptor) = 0; + + virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0; + virtual uint32_t GetLocalSsrc() const = 0; +}; + +std::unique_ptr CreateChannelReceive( + Clock* clock, + NetEqFactory* neteq_factory, + AudioDeviceModule* audio_device_module, + Transport* rtcp_send_transport, + RtcEventLog* rtc_event_log, + uint32_t local_ssrc, + uint32_t remote_ssrc, + size_t jitter_buffer_max_packets, + bool jitter_buffer_fast_playout, + int jitter_buffer_min_delay_ms, + bool enable_non_sender_rtt, + rtc::scoped_refptr decoder_factory, + absl::optional codec_pair_id, + rtc::scoped_refptr frame_decryptor, + const webrtc::CryptoOptions& crypto_options, + rtc::scoped_refptr frame_transformer, + RtcpEventObserver* rtcp_event_observer); + +} // namespace voe +} // namespace webrtc + +#endif // AUDIO_CHANNEL_RECEIVE_H_ -- cgit v1.2.3