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/*
* Copyright (c) 2015 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_AUDIO_SEND_STREAM_H_
#define AUDIO_AUDIO_SEND_STREAM_H_
#include <memory>
#include <vector>
#include "audio/channel_send.h"
#include "audio/transport_feedback_packet_loss_tracker.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/audio_allocation_settings.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class RtpTransportControllerSendInterface;
namespace internal {
class AudioState;
class AudioSendStream final : public webrtc::AudioSendStream,
public webrtc::BitrateAllocatorObserver,
public webrtc::PacketFeedbackObserver,
public webrtc::OverheadObserver {
public:
AudioSendStream(Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state);
// For unit tests, which need to supply a mock ChannelSend.
AudioSendStream(Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send);
~AudioSendStream() override;
// webrtc::AudioSendStream implementation.
const webrtc::AudioSendStream::Config& GetConfig() const override;
void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
void Start() override;
void Stop() override;
void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
bool SendTelephoneEvent(int payload_type,
int payload_frequency,
int event,
int duration_ms) override;
void SetMuted(bool muted) override;
webrtc::AudioSendStream::Stats GetStats() const override;
webrtc::AudioSendStream::Stats GetStats(
bool has_remote_tracks) const override;
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
// Implements BitrateAllocatorObserver.
uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
// From PacketFeedbackObserver.
void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
void OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) override;
void SetTransportOverhead(int transport_overhead_per_packet_bytes);
// OverheadObserver override reports audio packetization overhead from
// RTP/RTCP module or Media Transport.
void OnOverheadChanged(size_t overhead_bytes_per_packet_bytes) override;
RtpState GetRtpState() const;
const voe::ChannelSendInterface* GetChannel() const;
// Returns combined per-packet overhead.
size_t TestOnlyGetPerPacketOverheadBytes() const
RTC_LOCKS_EXCLUDED(overhead_per_packet_lock_);
private:
class TimedTransport;
internal::AudioState* audio_state();
const internal::AudioState* audio_state() const;
void StoreEncoderProperties(int sample_rate_hz, size_t num_channels);
// These are all static to make it less likely that (the old) config_ is
// accessed unintentionally.
static void ConfigureStream(AudioSendStream* stream,
const Config& new_config,
bool first_time);
static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
static bool ReconfigureSendCodec(AudioSendStream* stream,
const Config& new_config);
static void ReconfigureANA(AudioSendStream* stream, const Config& new_config);
static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config);
static void ReconfigureBitrateObserver(AudioSendStream* stream,
const Config& new_config);
void ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps,
double bitrate_priority);
void RemoveBitrateObserver();
// Sets per-packet overhead on encoded (for ANA) based on current known values
// of transport and packetization overheads.
void UpdateOverheadForEncoder()
RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
// Returns combined per-packet overhead.
size_t GetPerPacketOverheadBytes() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
void RegisterCngPayloadType(int payload_type, int clockrate_hz);
Clock* clock_;
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker pacer_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
rtc::TaskQueue* worker_queue_;
const AudioAllocationSettings allocation_settings_;
webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
RtcEventLog* const event_log_;
int encoder_sample_rate_hz_ = 0;
size_t encoder_num_channels_ = 0;
bool sending_ = false;
BitrateAllocatorInterface* const bitrate_allocator_;
RtpTransportControllerSendInterface* const rtp_transport_;
rtc::CriticalSection packet_loss_tracker_cs_;
TransportFeedbackPacketLossTracker packet_loss_tracker_
RTC_GUARDED_BY(&packet_loss_tracker_cs_);
RtpRtcp* rtp_rtcp_module_;
absl::optional<RtpState> const suspended_rtp_state_;
// RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
// reserved for padding and MUST NOT be used as a local identifier.
// So it should be safe to use 0 here to indicate "not configured".
struct ExtensionIds {
int audio_level = 0;
int transport_sequence_number = 0;
int mid = 0;
int rid = 0;
int repaired_rid = 0;
};
static ExtensionIds FindExtensionIds(
const std::vector<RtpExtension>& extensions);
static int TransportSeqNumId(const Config& config);
rtc::CriticalSection overhead_per_packet_lock_;
// Current transport overhead (ICE, TURN, etc.)
size_t transport_overhead_per_packet_bytes_
RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
// Current audio packetization overhead (RTP or Media Transport).
size_t audio_overhead_per_packet_bytes_
RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_SEND_STREAM_H_