| /* |
| * 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 CALL_AUDIO_SEND_STREAM_H_ |
| #define CALL_AUDIO_SEND_STREAM_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/audio_codecs/audio_codec_pair_id.h" |
| #include "api/audio_codecs/audio_encoder.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/call/transport.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/crypto/frame_encryptor_interface.h" |
| #include "api/rtp_parameters.h" |
| #include "api/scoped_refptr.h" |
| #include "call/rtp_config.h" |
| #include "modules/audio_processing/include/audio_processing_statistics.h" |
| #include "modules/rtp_rtcp/include/report_block_data.h" |
| |
| namespace webrtc { |
| |
| class AudioFrame; |
| |
| class AudioSendStream { |
| public: |
| struct Stats { |
| Stats(); |
| ~Stats(); |
| |
| // TODO(solenberg): Harmonize naming and defaults with receive stream stats. |
| uint32_t local_ssrc = 0; |
| int64_t payload_bytes_sent = 0; |
| int64_t header_and_padding_bytes_sent = 0; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent |
| uint64_t retransmitted_bytes_sent = 0; |
| int32_t packets_sent = 0; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent |
| uint64_t retransmitted_packets_sent = 0; |
| int32_t packets_lost = -1; |
| float fraction_lost = -1.0f; |
| std::string codec_name; |
| absl::optional<int> codec_payload_type; |
| int32_t jitter_ms = -1; |
| int64_t rtt_ms = -1; |
| int16_t audio_level = 0; |
| // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy |
| double total_input_energy = 0.0; |
| double total_input_duration = 0.0; |
| bool typing_noise_detected = false; |
| |
| ANAStats ana_statistics; |
| AudioProcessingStats apm_statistics; |
| |
| int64_t target_bitrate_bps = 0; |
| // A snapshot of Report Blocks with additional data of interest to |
| // statistics. Within this list, the sender-source SSRC pair is unique and |
| // per-pair the ReportBlockData represents the latest Report Block that was |
| // received for that pair. |
| std::vector<ReportBlockData> report_block_datas; |
| }; |
| |
| struct Config { |
| Config() = delete; |
| explicit Config(Transport* send_transport); |
| ~Config(); |
| std::string ToString() const; |
| |
| // Send-stream specific RTP settings. |
| struct Rtp { |
| Rtp(); |
| ~Rtp(); |
| std::string ToString() const; |
| |
| // Sender SSRC. |
| uint32_t ssrc = 0; |
| |
| // The value to send in the RID RTP header extension if the extension is |
| // included in the list of extensions. |
| std::string rid; |
| |
| // The value to send in the MID RTP header extension if the extension is |
| // included in the list of extensions. |
| std::string mid; |
| |
| // Corresponds to the SDP attribute extmap-allow-mixed. |
| bool extmap_allow_mixed = false; |
| |
| // RTP header extensions used for the sent stream. |
| std::vector<RtpExtension> extensions; |
| |
| // RTCP CNAME, see RFC 3550. |
| std::string c_name; |
| } rtp; |
| |
| // Time interval between RTCP report for audio |
| int rtcp_report_interval_ms = 5000; |
| |
| // Transport for outgoing packets. The transport is expected to exist for |
| // the entire life of the AudioSendStream and is owned by the API client. |
| Transport* send_transport = nullptr; |
| |
| // Bitrate limits used for variable audio bitrate streams. Set both to -1 to |
| // disable audio bitrate adaptation. |
| // Note: This is still an experimental feature and not ready for real usage. |
| int min_bitrate_bps = -1; |
| int max_bitrate_bps = -1; |
| |
| double bitrate_priority = 1.0; |
| bool has_dscp = false; |
| |
| // Defines whether to turn on audio network adaptor, and defines its config |
| // string. |
| absl::optional<std::string> audio_network_adaptor_config; |
| |
| struct SendCodecSpec { |
| SendCodecSpec(int payload_type, const SdpAudioFormat& format); |
| ~SendCodecSpec(); |
| std::string ToString() const; |
| |
| bool operator==(const SendCodecSpec& rhs) const; |
| bool operator!=(const SendCodecSpec& rhs) const { |
| return !(*this == rhs); |
| } |
| |
| int payload_type; |
| SdpAudioFormat format; |
| bool nack_enabled = false; |
| bool transport_cc_enabled = false; |
| absl::optional<int> cng_payload_type; |
| // If unset, use the encoder's default target bitrate. |
| absl::optional<int> target_bitrate_bps; |
| }; |
| |
| absl::optional<SendCodecSpec> send_codec_spec; |
| rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
| absl::optional<AudioCodecPairId> codec_pair_id; |
| |
| // Track ID as specified during track creation. |
| std::string track_id; |
| |
| // Per PeerConnection crypto options. |
| webrtc::CryptoOptions crypto_options; |
| |
| // An optional custom frame encryptor that allows the entire frame to be |
| // encryptor in whatever way the caller choses. This is not required by |
| // default. |
| rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor; |
| }; |
| |
| virtual ~AudioSendStream() = default; |
| |
| virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0; |
| |
| // Reconfigure the stream according to the Configuration. |
| virtual void Reconfigure(const Config& config) = 0; |
| |
| // Starts stream activity. |
| // When a stream is active, it can receive, process and deliver packets. |
| virtual void Start() = 0; |
| // Stops stream activity. |
| // When a stream is stopped, it can't receive, process or deliver packets. |
| virtual void Stop() = 0; |
| |
| // Encode and send audio. |
| virtual void SendAudioData( |
| std::unique_ptr<webrtc::AudioFrame> audio_frame) = 0; |
| |
| // TODO(solenberg): Make payload_type a config property instead. |
| virtual bool SendTelephoneEvent(int payload_type, |
| int payload_frequency, |
| int event, |
| int duration_ms) = 0; |
| |
| virtual void SetMuted(bool muted) = 0; |
| |
| virtual Stats GetStats() const = 0; |
| virtual Stats GetStats(bool has_remote_tracks) const = 0; |
| }; |
| } // namespace webrtc |
| |
| #endif // CALL_AUDIO_SEND_STREAM_H_ |