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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/audio/voip/audio_ingress.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/voip/audio_ingress.h')
-rw-r--r-- | third_party/libwebrtc/audio/voip/audio_ingress.h | 145 |
1 files changed, 145 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/voip/audio_ingress.h b/third_party/libwebrtc/audio/voip/audio_ingress.h new file mode 100644 index 0000000000..11bde7ce28 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/audio_ingress.h @@ -0,0 +1,145 @@ +/* + * 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_ |