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/*
* Copyright (c) 2020 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_VOIP_AUDIO_INGRESS_H_
#define AUDIO_VOIP_AUDIO_INGRESS_H_
#include <algorithm>
#include <atomic>
#include <map>
#include <memory>
#include <utility>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_mixer.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
#include "api/voip/voip_statistics.h"
#include "audio/audio_level.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
// AudioIngress handles incoming RTP/RTCP packets from the remote
// media endpoint. Received RTP packets are injected into AcmReceiver and
// when audio output thread requests for audio samples to play through system
// output such as speaker device, AudioIngress provides the samples via its
// implementation on AudioMixer::Source interface.
//
// Note that this class is originally based on ChannelReceive in
// audio/channel_receive.cc with non-audio related logic trimmed as aimed for
// smaller footprint.
class AudioIngress : public AudioMixer::Source {
public:
AudioIngress(RtpRtcpInterface* rtp_rtcp,
Clock* clock,
ReceiveStatistics* receive_statistics,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
~AudioIngress() override;
// Start or stop receiving operation of AudioIngress.
bool StartPlay();
void StopPlay() {
playing_ = false;
output_audio_level_.ResetLevelFullRange();
}
// Query the state of the AudioIngress.
bool IsPlaying() const { return playing_; }
// Set the decoder formats and payload type for AcmReceiver where the
// key type (int) of the map is the payload type of SdpAudioFormat.
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
// APIs to handle received RTP/RTCP packets from caller.
void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet);
void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet);
// See comments on LevelFullRange, TotalEnergy, TotalDuration from
// audio/audio_level.h.
int GetOutputAudioLevel() const {
return output_audio_level_.LevelFullRange();
}
double GetOutputTotalEnergy() { return output_audio_level_.TotalEnergy(); }
double GetOutputTotalDuration() {
return output_audio_level_.TotalDuration();
}
NetworkStatistics GetNetworkStatistics() const {
NetworkStatistics stats;
acm_receiver_.GetNetworkStatistics(&stats,
/*get_and_clear_legacy_stats=*/false);
return stats;
}
ChannelStatistics GetChannelStatistics();
// Implementation of AudioMixer::Source interface.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sampling_rate,
AudioFrame* audio_frame) override;
int Ssrc() const override {
return rtc::dchecked_cast<int>(remote_ssrc_.load());
}
int PreferredSampleRate() const override {
// If we haven't received any RTP packet from remote and thus
// last_packet_sampling_rate is not available then use NetEq's sampling
// rate as that would be what would be used for audio output sample.
return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
acm_receiver_.last_output_sample_rate_hz());
}
private:
// Indicates AudioIngress status as caller invokes Start/StopPlaying.
// If not playing, incoming RTP data processing is skipped, thus
// producing no data to output device.
std::atomic<bool> playing_;
// Currently active remote ssrc from remote media endpoint.
std::atomic<uint32_t> remote_ssrc_;
// The first rtp timestamp of the output audio frame that is used to
// calculate elasped time for subsequent audio frames.
std::atomic<int64_t> first_rtp_timestamp_;
// Synchronizaton is handled internally by ReceiveStatistics.
ReceiveStatistics* const rtp_receive_statistics_;
// Synchronizaton is handled internally by RtpRtcpInterface.
RtpRtcpInterface* const rtp_rtcp_;
// Synchronizaton is handled internally by acm2::AcmReceiver.
acm2::AcmReceiver acm_receiver_;
// Synchronizaton is handled internally by voe::AudioLevel.
voe::AudioLevel output_audio_level_;
Mutex lock_;
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
// For receiving RTP statistics, this tracks the sampling rate value
// per payload type set when caller set via SetReceiveCodecs.
std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
#endif // AUDIO_VOIP_AUDIO_INGRESS_H_
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