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172 lines
6.0 KiB
172 lines
6.0 KiB
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_CHANNEL_RECEIVE_H_
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#define AUDIO_CHANNEL_RECEIVE_H_
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#include <map>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio/audio_mixer.h"
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/call/audio_sink.h"
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#include "api/call/transport.h"
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#include "api/crypto/crypto_options.h"
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#include "api/frame_transformer_interface.h"
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#include "api/neteq/neteq_factory.h"
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#include "api/transport/rtp/rtp_source.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "system_wrappers/include/clock.h"
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// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
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// warnings about use of unsigned short.
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// These need cleanup, in a separate cl.
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namespace rtc {
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class TimestampWrapAroundHandler;
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}
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namespace webrtc {
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class AudioDeviceModule;
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class FrameDecryptorInterface;
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class PacketRouter;
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class ProcessThread;
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class RateLimiter;
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class ReceiveStatistics;
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class RtcEventLog;
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class RtpPacketReceived;
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class RtpRtcp;
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struct CallReceiveStatistics {
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unsigned int cumulativeLost;
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unsigned int jitterSamples;
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int64_t rttMs;
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int64_t payload_bytes_rcvd = 0;
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int64_t header_and_padding_bytes_rcvd = 0;
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int packetsReceived;
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// The capture ntp time (in local timebase) of the first played out audio
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// frame.
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int64_t capture_start_ntp_time_ms_;
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// The timestamp at which the last packet was received, i.e. the time of the
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// local clock when it was received - not the RTP timestamp of that packet.
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// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
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absl::optional<int64_t> last_packet_received_timestamp_ms;
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};
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namespace voe {
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class ChannelSendInterface;
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// Interface class needed for AudioReceiveStream tests that use a
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// MockChannelReceive.
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class ChannelReceiveInterface : public RtpPacketSinkInterface {
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public:
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virtual ~ChannelReceiveInterface() = default;
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virtual void SetSink(AudioSinkInterface* sink) = 0;
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virtual void SetReceiveCodecs(
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const std::map<int, SdpAudioFormat>& codecs) = 0;
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virtual void StartPlayout() = 0;
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virtual void StopPlayout() = 0;
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// Payload type and format of last received RTP packet, if any.
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virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
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const = 0;
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virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
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virtual void SetChannelOutputVolumeScaling(float scaling) = 0;
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virtual int GetSpeechOutputLevelFullRange() const = 0;
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// See description of "totalAudioEnergy" in the WebRTC stats spec:
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
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virtual double GetTotalOutputEnergy() const = 0;
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virtual double GetTotalOutputDuration() const = 0;
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// Stats.
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virtual NetworkStatistics GetNetworkStatistics() const = 0;
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virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0;
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// Audio+Video Sync.
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virtual uint32_t GetDelayEstimate() const = 0;
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virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
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virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
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int64_t* time_ms) const = 0;
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virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
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int64_t time_ms) = 0;
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virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
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int64_t now_ms) const = 0;
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// Audio quality.
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// Base minimum delay sets lower bound on minimum delay value which
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// determines minimum delay until audio playout.
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virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
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virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
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// Produces the transport-related timestamps; current_delay_ms is left unset.
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virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
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virtual void RegisterReceiverCongestionControlObjects(
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PacketRouter* packet_router) = 0;
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virtual void ResetReceiverCongestionControlObjects() = 0;
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virtual CallReceiveStatistics GetRTCPStatistics() const = 0;
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virtual void SetNACKStatus(bool enable, int max_packets) = 0;
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virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
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int sample_rate_hz,
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AudioFrame* audio_frame) = 0;
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virtual int PreferredSampleRate() const = 0;
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// Associate to a send channel.
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// Used for obtaining RTT for a receive-only channel.
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virtual void SetAssociatedSendChannel(
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const ChannelSendInterface* channel) = 0;
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// Sets a frame transformer between the depacketizer and the decoder, to
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// transform the received frames before decoding them.
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virtual void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface>
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frame_transformer) = 0;
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};
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std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
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Clock* clock,
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ProcessThread* module_process_thread,
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NetEqFactory* neteq_factory,
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AudioDeviceModule* audio_device_module,
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Transport* rtcp_send_transport,
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RtcEventLog* rtc_event_log,
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uint32_t local_ssrc,
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uint32_t remote_ssrc,
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size_t jitter_buffer_max_packets,
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bool jitter_buffer_fast_playout,
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int jitter_buffer_min_delay_ms,
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bool jitter_buffer_enable_rtx_handling,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
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absl::optional<AudioCodecPairId> codec_pair_id,
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
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const webrtc::CryptoOptions& crypto_options,
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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} // namespace voe
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} // namespace webrtc
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#endif // AUDIO_CHANNEL_RECEIVE_H_
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