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Diffstat (limited to 'third_party/libwebrtc/call/audio_send_stream.h')
-rw-r--r-- | third_party/libwebrtc/call/audio_send_stream.h | 203 |
1 files changed, 203 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/audio_send_stream.h b/third_party/libwebrtc/call/audio_send_stream.h new file mode 100644 index 0000000000..187ec65ed8 --- /dev/null +++ b/third_party/libwebrtc/call/audio_send_stream.h @@ -0,0 +1,203 @@ +/* + * 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/frame_transformer_interface.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_setparameters_callback.h" +#include "api/scoped_refptr.h" +#include "call/audio_sender.h" +#include "call/rtp_config.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" + +namespace webrtc { + +class AudioSendStream : public AudioSender { + 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-totalpacketsenddelay + TimeDelta total_packet_send_delay = TimeDelta::Zero(); + // 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; + + ANAStats ana_statistics; + AudioProcessingStats apm_statistics; + RtcpPacketTypeCounter rtcp_packet_type_counts; + + 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; + uint32_t nacks_rcvd = 0; + }; + + 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; + bool enable_non_sender_rtt = false; + absl::optional<int> cng_payload_type; + absl::optional<int> red_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; + + // An optional frame transformer used by insertable streams to transform + // encoded frames. + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; + }; + + virtual ~AudioSendStream() = default; + + virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0; + + // Reconfigure the stream according to the Configuration. + virtual void Reconfigure(const Config& config, + SetParametersCallback callback) = 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; + + // 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_ |