626015d7f8
In order to have correct overhead adjustments for ANA in AudioSendStream we need to know both transport and audio packetization overheads in AudioSendStream. This change makes AudioSendStream to be OverheadObserver instead of ChannelSend. This change is required for https://webrtc-review.googlesource.com/c/src/+/115082. This change was also suggested earlier in TODO: // TODO(solenberg): Make AudioSendStream an OverheadObserver instead. https://cs.chromium.org/chromium/src/third_party/webrtc/audio/channel_send.cc?rcl=71b5a7df7794bbc4103296fcd8bd740acebdc901&l=1181 I think we should also consider moving TargetTransferRate observer to AudioSendStream. Since AudioSendStream owns encoder and configures ANA, it makes sense to consolidate all rate (and overhead) calculation there. Added TODO to clean it up in next chanelists. Bug: webrtc:10150 Change-Id: I48791b998ea00ffde9ec75c6bca8b6dc83001b42 Reviewed-on: https://webrtc-review.googlesource.com/c/119121 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26540}
133 lines
4.7 KiB
C++
133 lines
4.7 KiB
C++
/*
|
|
* Copyright (c) 2012 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_CHANNEL_SEND_H_
|
|
#define AUDIO_CHANNEL_SEND_H_
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "api/audio/audio_frame.h"
|
|
#include "api/audio_codecs/audio_encoder.h"
|
|
#include "api/crypto/crypto_options.h"
|
|
#include "api/media_transport_interface.h"
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
|
|
#include "rtc_base/function_view.h"
|
|
#include "rtc_base/task_queue.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class FrameEncryptorInterface;
|
|
class ProcessThread;
|
|
class RtcEventLog;
|
|
class RtpRtcp;
|
|
class RtpTransportControllerSendInterface;
|
|
|
|
struct CallSendStatistics {
|
|
int64_t rttMs;
|
|
size_t bytesSent;
|
|
int packetsSent;
|
|
};
|
|
|
|
// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
|
|
struct ReportBlock {
|
|
uint32_t sender_SSRC; // SSRC of sender
|
|
uint32_t source_SSRC;
|
|
uint8_t fraction_lost;
|
|
int32_t cumulative_num_packets_lost;
|
|
uint32_t extended_highest_sequence_number;
|
|
uint32_t interarrival_jitter;
|
|
uint32_t last_SR_timestamp;
|
|
uint32_t delay_since_last_SR;
|
|
};
|
|
|
|
namespace voe {
|
|
|
|
class ChannelSendInterface {
|
|
public:
|
|
virtual ~ChannelSendInterface() = default;
|
|
|
|
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
|
|
|
|
virtual CallSendStatistics GetRTCPStatistics() const = 0;
|
|
|
|
virtual bool SetEncoder(int payload_type,
|
|
std::unique_ptr<AudioEncoder> encoder) = 0;
|
|
virtual void ModifyEncoder(
|
|
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
|
|
|
|
virtual void SetLocalSSRC(uint32_t ssrc) = 0;
|
|
// Use 0 to indicate that the extension should not be registered.
|
|
virtual void SetRid(const std::string& rid,
|
|
int extension_id,
|
|
int repaired_extension_id) = 0;
|
|
virtual void SetMid(const std::string& mid, int extension_id) = 0;
|
|
virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
|
|
virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
|
|
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
|
|
virtual void EnableSendTransportSequenceNumber(int id) = 0;
|
|
virtual void RegisterSenderCongestionControlObjects(
|
|
RtpTransportControllerSendInterface* transport,
|
|
RtcpBandwidthObserver* bandwidth_observer) = 0;
|
|
virtual void ResetSenderCongestionControlObjects() = 0;
|
|
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const = 0;
|
|
virtual ANAStats GetANAStatistics() const = 0;
|
|
virtual bool SetSendTelephoneEventPayloadType(int payload_type,
|
|
int payload_frequency) = 0;
|
|
virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
|
|
virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
|
|
virtual int GetBitrate() const = 0;
|
|
virtual void SetInputMute(bool muted) = 0;
|
|
|
|
virtual void ProcessAndEncodeAudio(
|
|
std::unique_ptr<AudioFrame> audio_frame) = 0;
|
|
virtual RtpRtcp* GetRtpRtcp() const = 0;
|
|
|
|
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) = 0;
|
|
virtual void OnRecoverableUplinkPacketLossRate(
|
|
float recoverable_packet_loss_rate) = 0;
|
|
// In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform
|
|
// about RTT.
|
|
// In media transport we rely on the TargetTransferRateObserver instead.
|
|
// In other words, if you are using RTP, you should expect
|
|
// |ReceivedRTCPPacket| to be called, if you are using media transport,
|
|
// |OnTargetTransferRate| will be called.
|
|
//
|
|
// In future, RTP media will move to the media transport implementation and
|
|
// these conditions will be removed.
|
|
// Returns the RTT in milliseconds.
|
|
virtual int64_t GetRTT() const = 0;
|
|
virtual void StartSend() = 0;
|
|
virtual void StopSend() = 0;
|
|
|
|
// E2EE Custom Audio Frame Encryption (Optional)
|
|
virtual void SetFrameEncryptor(
|
|
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
|
|
};
|
|
|
|
std::unique_ptr<ChannelSendInterface> CreateChannelSend(
|
|
rtc::TaskQueue* encoder_queue,
|
|
ProcessThread* module_process_thread,
|
|
MediaTransportInterface* media_transport,
|
|
OverheadObserver* overhead_observer,
|
|
Transport* rtp_transport,
|
|
RtcpRttStats* rtcp_rtt_stats,
|
|
RtcEventLog* rtc_event_log,
|
|
FrameEncryptorInterface* frame_encryptor,
|
|
const webrtc::CryptoOptions& crypto_options,
|
|
bool extmap_allow_mixed,
|
|
int rtcp_report_interval_ms);
|
|
|
|
} // namespace voe
|
|
} // namespace webrtc
|
|
|
|
#endif // AUDIO_CHANNEL_SEND_H_
|