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/*
* 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/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/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 {
unsigned int cumulativeLost;
unsigned int jitterSamples;
int64_t rttMs;
int64_t payload_bytes_rcvd = 0;
int64_t header_and_padding_bytes_rcvd = 0;
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;
// 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<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;
};
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(
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<int64_t> 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<Syncable::Info> 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 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<webrtc::FrameTransformerInterface>
frame_transformer) = 0;
virtual void SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0;
virtual uint32_t GetLocalSsrc() const = 0;
};
std::unique_ptr<ChannelReceiveInterface> 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 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,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
} // namespace voe
} // namespace webrtc
#endif // AUDIO_CHANNEL_RECEIVE_H_
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