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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_channel.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_channel.h')
-rw-r--r-- | third_party/libwebrtc/audio/voip/audio_channel.h | 131 |
1 files changed, 131 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/voip/audio_channel.h b/third_party/libwebrtc/audio/voip/audio_channel.h new file mode 100644 index 0000000000..7338d9faab --- /dev/null +++ b/third_party/libwebrtc/audio/voip/audio_channel.h @@ -0,0 +1,131 @@ +/* + * 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_CHANNEL_H_ +#define AUDIO_VOIP_AUDIO_CHANNEL_H_ + +#include <map> +#include <memory> +#include <queue> +#include <utility> + +#include "api/task_queue/task_queue_factory.h" +#include "api/voip/voip_base.h" +#include "api/voip/voip_statistics.h" +#include "audio/voip/audio_egress.h" +#include "audio/voip/audio_ingress.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "rtc_base/ref_count.h" + +namespace webrtc { + +// AudioChannel represents a single media session and provides APIs over +// AudioIngress and AudioEgress. Note that a single RTP stack is shared with +// these two classes as it has both sending and receiving capabilities. +class AudioChannel : public rtc::RefCountInterface { + public: + AudioChannel(Transport* transport, + uint32_t local_ssrc, + TaskQueueFactory* task_queue_factory, + AudioMixer* audio_mixer, + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory); + ~AudioChannel() override; + + // Set and get ChannelId that this audio channel belongs for debugging and + // logging purpose. + void SetId(ChannelId id) { id_ = id; } + ChannelId GetId() const { return id_; } + + // APIs to start/stop audio channel on each direction. + // StartSend/StartPlay returns false if encoder/decoders + // have not been set, respectively. + bool StartSend(); + void StopSend(); + bool StartPlay(); + void StopPlay(); + + // APIs relayed to AudioEgress. + bool IsSendingMedia() const { return egress_->IsSending(); } + AudioSender* GetAudioSender() { return egress_.get(); } + void SetEncoder(int payload_type, + const SdpAudioFormat& encoder_format, + std::unique_ptr<AudioEncoder> encoder) { + egress_->SetEncoder(payload_type, encoder_format, std::move(encoder)); + } + absl::optional<SdpAudioFormat> GetEncoderFormat() const { + return egress_->GetEncoderFormat(); + } + void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz) { + egress_->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz); + } + bool SendTelephoneEvent(int dtmf_event, int duration_ms) { + return egress_->SendTelephoneEvent(dtmf_event, duration_ms); + } + void SetMute(bool enable) { egress_->SetMute(enable); } + + // APIs relayed to AudioIngress. + bool IsPlaying() const { return ingress_->IsPlaying(); } + void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) { + ingress_->ReceivedRTPPacket(rtp_packet); + } + void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet) { + ingress_->ReceivedRTCPPacket(rtcp_packet); + } + void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) { + ingress_->SetReceiveCodecs(codecs); + } + IngressStatistics GetIngressStatistics(); + ChannelStatistics GetChannelStatistics(); + + // See comments on the methods used from AudioEgress and AudioIngress. + // Conversion to double is following what is done in + // DoubleAudioLevelFromIntAudioLevel method in rtc_stats_collector.cc to be + // consistent. + double GetInputAudioLevel() const { + return egress_->GetInputAudioLevel() / 32767.0; + } + double GetInputTotalEnergy() const { return egress_->GetInputTotalEnergy(); } + double GetInputTotalDuration() const { + return egress_->GetInputTotalDuration(); + } + double GetOutputAudioLevel() const { + return ingress_->GetOutputAudioLevel() / 32767.0; + } + double GetOutputTotalEnergy() const { + return ingress_->GetOutputTotalEnergy(); + } + double GetOutputTotalDuration() const { + return ingress_->GetOutputTotalDuration(); + } + + // Internal API for testing purpose. + void SendRTCPReportForTesting(RTCPPacketType type) { + int32_t result = rtp_rtcp_->SendRTCP(type); + RTC_DCHECK(result == 0); + } + + private: + // ChannelId that this audio channel belongs for logging purpose. + ChannelId id_; + + // Synchronization is handled internally by AudioMixer. + AudioMixer* audio_mixer_; + + // Listed in order for safe destruction of AudioChannel object. + // Synchronization for these are handled internally. + std::unique_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; + std::unique_ptr<AudioIngress> ingress_; + std::unique_ptr<AudioEgress> egress_; +}; + +} // namespace webrtc + +#endif // AUDIO_VOIP_AUDIO_CHANNEL_H_ |