| /* |
| * 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 "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/synchronization/mutex.h" |
| #include "rtc_base/time_utils.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; |
| } |
| |
| // 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: |
| // Returns network round trip time (RTT) measued by RTCP exchange with |
| // remote media endpoint. Returns absl::nullopt when it's not initialized. |
| absl::optional<int64_t> GetRoundTripTime(); |
| |
| // 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_); |
| |
| rtc::TimestampWrapAroundHandler timestamp_wrap_handler_ RTC_GUARDED_BY(lock_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // AUDIO_VOIP_AUDIO_INGRESS_H_ |