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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_EGRESS_H_
#define AUDIO_VOIP_AUDIO_EGRESS_H_
#include <memory>
#include <string>
#include "api/audio_codecs/audio_format.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
#include "audio/audio_level.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/audio_sender.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
// AudioEgress receives input samples from AudioDeviceModule via
// AudioTransportImpl through AudioSender interface. Once it encodes the sample
// via selected encoder through AudioPacketizationCallback interface, the
// encoded payload will be packetized by the RTP stack, resulting in ready to
// send RTP packet to remote endpoint.
//
// TaskQueue is used to encode and send RTP asynchrounously as some OS platform
// uses the same thread for both audio input and output sample deliveries which
// can affect audio quality.
//
// Note that this class is originally based on ChannelSend in
// audio/channel_send.cc with non-audio related logic trimmed as aimed for
// smaller footprint.
class AudioEgress : public AudioSender, public AudioPacketizationCallback {
public:
AudioEgress(RtpRtcpInterface* rtp_rtcp,
Clock* clock,
TaskQueueFactory* task_queue_factory);
~AudioEgress() override;
// Set the encoder format and payload type for AudioCodingModule.
// It's possible to change the encoder type during its active usage.
// `payload_type` must be the type that is negotiated with peer through
// offer/answer.
void SetEncoder(int payload_type,
const SdpAudioFormat& encoder_format,
std::unique_ptr<AudioEncoder> encoder);
// Start or stop sending operation of AudioEgress. This will start/stop
// the RTP stack also causes encoder queue thread to start/stop
// processing input audio samples. StartSend will return false if
// a send codec has not been set.
bool StartSend();
void StopSend();
// Query the state of the RTP stack. This returns true if StartSend()
// called and false if StopSend() is called.
bool IsSending() const;
// Enable or disable Mute state.
void SetMute(bool mute);
// Retrieve current encoder format info. This returns encoder format set
// by SetEncoder() and if encoder is not set, this will return nullopt.
absl::optional<SdpAudioFormat> GetEncoderFormat() const {
MutexLock lock(&lock_);
return encoder_format_;
}
// Register the payload type and sample rate for DTMF (RFC 4733) payload.
void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz);
// Send DTMF named event as specified by
// https://tools.ietf.org/html/rfc4733#section-3.2
// `duration_ms` specifies the duration of DTMF packets that will be emitted
// in place of real RTP packets instead.
// This will return true when requested dtmf event is successfully scheduled
// otherwise false when the dtmf queue reached maximum of 20 events.
bool SendTelephoneEvent(int dtmf_event, int duration_ms);
// See comments on LevelFullRange, TotalEnergy, TotalDuration from
// audio/audio_level.h.
int GetInputAudioLevel() const { return input_audio_level_.LevelFullRange(); }
double GetInputTotalEnergy() const {
return input_audio_level_.TotalEnergy();
}
double GetInputTotalDuration() const {
return input_audio_level_.TotalDuration();
}
// Implementation of AudioSender interface.
void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
// Implementation of AudioPacketizationCallback interface.
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size) override;
private:
void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
MutexLock lock(&lock_);
encoder_format_ = encoder_format;
}
mutable Mutex lock_;
// Current encoder format selected by caller.
absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
// Synchronization is handled internally by RtpRtcp.
RtpRtcpInterface* const rtp_rtcp_;
// Synchronization is handled internally by RTPSenderAudio.
RTPSenderAudio rtp_sender_audio_;
// Synchronization is handled internally by AudioCodingModule.
const std::unique_ptr<AudioCodingModule> audio_coding_;
// Synchronization is handled internally by voe::AudioLevel.
voe::AudioLevel input_audio_level_;
// Struct that holds all variables used by encoder task queue.
struct EncoderContext {
// Offset used to mark rtp timestamp in sample rate unit in
// newly received audio frame from AudioTransport.
uint32_t frame_rtp_timestamp_ = 0;
// Flag to track mute state from caller. `previously_muted_` is used to
// track previous state as part of input to AudioFrameOperations::Mute
// to implement fading effect when (un)mute is invoked.
bool mute_ = false;
bool previously_muted_ = false;
};
EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_);
// Defined last to ensure that there are no running tasks when the other
// members are destroyed.
rtc::TaskQueue encoder_queue_;
};
} // namespace webrtc
#endif // AUDIO_VOIP_AUDIO_EGRESS_H_
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