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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_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_
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