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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 <utility>
#include <vector>
#include "api/sequence_checker.h"
#include "audio/audio_level.h"
#include "audio/channel_send.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
namespace webrtc {
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class RtpTransportControllerSendInterface;
struct AudioAllocationConfig {
static constexpr char kKey[] = "WebRTC-Audio-Allocation";
// Field Trial configured bitrates to use as overrides over default/user
// configured bitrate range when audio bitrate allocation is enabled.
absl::optional<DataRate> min_bitrate;
absl::optional<DataRate> max_bitrate;
DataRate priority_bitrate = DataRate::Zero();
// By default the priority_bitrate is compensated for packet overhead.
// Use this flag to configure a raw value instead.
absl::optional<DataRate> priority_bitrate_raw;
absl::optional<double> bitrate_priority;
std::unique_ptr<StructParametersParser> Parser();
AudioAllocationConfig();
};
namespace internal {
class AudioState;
class AudioSendStream final : public webrtc::AudioSendStream,
public webrtc::BitrateAllocatorObserver {
public:
AudioSendStream(Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
TaskQueueFactory* task_queue_factory,
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,
TaskQueueFactory* task_queue_factory,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send);
AudioSendStream() = delete;
AudioSendStream(const AudioSendStream&) = delete;
AudioSendStream& operator=(const AudioSendStream&) = delete;
~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 DeliverRtcp(const uint8_t* packet, size_t length);
// Implements BitrateAllocatorObserver.
uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
void SetTransportOverhead(int transport_overhead_per_packet_bytes);
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;
// Constraints including overhead.
struct TargetAudioBitrateConstraints {
DataRate min;
DataRate max;
};
internal::AudioState* audio_state();
const internal::AudioState* audio_state() const;
void StoreEncoderProperties(int sample_rate_hz, size_t num_channels)
RTC_RUN_ON(worker_thread_checker_);
void ConfigureStream(const Config& new_config, bool first_time)
RTC_RUN_ON(worker_thread_checker_);
bool SetupSendCodec(const Config& new_config)
RTC_RUN_ON(worker_thread_checker_);
bool ReconfigureSendCodec(const Config& new_config)
RTC_RUN_ON(worker_thread_checker_);
void ReconfigureANA(const Config& new_config)
RTC_RUN_ON(worker_thread_checker_);
void ReconfigureCNG(const Config& new_config)
RTC_RUN_ON(worker_thread_checker_);
void ReconfigureBitrateObserver(const Config& new_config)
RTC_RUN_ON(worker_thread_checker_);
void ConfigureBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
void RemoveBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
// Returns bitrate constraints, maybe including overhead when enabled by
// field trial.
absl::optional<TargetAudioBitrateConstraints> GetMinMaxBitrateConstraints()
const RTC_RUN_ON(worker_thread_checker_);
// 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)
RTC_RUN_ON(worker_thread_checker_);
void UpdateCachedTargetAudioBitrateConstraints()
RTC_RUN_ON(worker_thread_checker_);
Clock* clock_;
SequenceChecker worker_thread_checker_;
SequenceChecker pacer_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
rtc::TaskQueue* rtp_transport_queue_;
const bool allocate_audio_without_feedback_;
const bool force_no_audio_feedback_ = allocate_audio_without_feedback_;
const bool enable_audio_alr_probing_;
const bool send_side_bwe_with_overhead_;
const AudioAllocationConfig allocation_settings_;
webrtc::AudioSendStream::Config config_
RTC_GUARDED_BY(worker_thread_checker_);
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
RtcEventLog* const event_log_;
const bool use_legacy_overhead_calculation_;
int encoder_sample_rate_hz_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
size_t encoder_num_channels_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
bool sending_ RTC_GUARDED_BY(worker_thread_checker_) = false;
mutable Mutex audio_level_lock_;
// Keeps track of audio level, total audio energy and total samples duration.
// https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy
webrtc::voe::AudioLevel audio_level_ RTC_GUARDED_BY(audio_level_lock_);
BitrateAllocatorInterface* const bitrate_allocator_
RTC_GUARDED_BY(rtp_transport_queue_);
// Constrains cached to be accessed from |rtp_transport_queue_|.
absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
cached_constraints_ RTC_GUARDED_BY(rtp_transport_queue_) = absl::nullopt;
RtpTransportControllerSendInterface* const rtp_transport_;
RtpRtcpInterface* const 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 abs_send_time = 0;
int abs_capture_time = 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);
mutable Mutex overhead_per_packet_lock_;
size_t overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
// Current transport overhead (ICE, TURN, etc.)
size_t transport_overhead_per_packet_bytes_
RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
bool registered_with_allocator_ RTC_GUARDED_BY(worker_thread_checker_) =
false;
size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_thread_checker_) =
0;
absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
RTC_GUARDED_BY(worker_thread_checker_);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_SEND_STREAM_H_
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