| /* |
| * 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 WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ |
| #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/api/audio_codecs/audio_decoder_factory.h" |
| #include "webrtc/api/call/transport.h" |
| #include "webrtc/api/rtpparameters.h" |
| #include "webrtc/api/rtpreceiverinterface.h" |
| #include "webrtc/call/rtp_config.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/rtc_base/optional.h" |
| #include "webrtc/rtc_base/scoped_ref_ptr.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| class AudioSinkInterface; |
| |
| // WORK IN PROGRESS |
| // This class is under development and is not yet intended for for use outside |
| // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
| |
| class AudioReceiveStream { |
| public: |
| struct Stats { |
| uint32_t remote_ssrc = 0; |
| int64_t bytes_rcvd = 0; |
| uint32_t packets_rcvd = 0; |
| uint32_t packets_lost = 0; |
| float fraction_lost = 0.0f; |
| std::string codec_name; |
| rtc::Optional<int> codec_payload_type; |
| uint32_t ext_seqnum = 0; |
| uint32_t jitter_ms = 0; |
| uint32_t jitter_buffer_ms = 0; |
| uint32_t jitter_buffer_preferred_ms = 0; |
| uint32_t delay_estimate_ms = 0; |
| int32_t audio_level = -1; |
| // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy |
| double total_output_energy = 0.0; |
| // See description of "totalSamplesReceived" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived |
| uint64_t total_samples_received = 0; |
| // See description of "totalSamplesDuration" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesduration |
| double total_output_duration = 0.0; |
| // See description of "concealedSamples" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples |
| uint64_t concealed_samples = 0; |
| float expand_rate = 0.0f; |
| float speech_expand_rate = 0.0f; |
| float secondary_decoded_rate = 0.0f; |
| float secondary_discarded_rate = 0.0f; |
| float accelerate_rate = 0.0f; |
| float preemptive_expand_rate = 0.0f; |
| int32_t decoding_calls_to_silence_generator = 0; |
| int32_t decoding_calls_to_neteq = 0; |
| int32_t decoding_normal = 0; |
| int32_t decoding_plc = 0; |
| int32_t decoding_cng = 0; |
| int32_t decoding_plc_cng = 0; |
| int32_t decoding_muted_output = 0; |
| int64_t capture_start_ntp_time_ms = 0; |
| }; |
| |
| struct Config { |
| std::string ToString() const; |
| |
| // Receive-stream specific RTP settings. |
| struct Rtp { |
| std::string ToString() const; |
| |
| // Synchronization source (stream identifier) to be received. |
| uint32_t remote_ssrc = 0; |
| |
| // Sender SSRC used for sending RTCP (such as receiver reports). |
| uint32_t local_ssrc = 0; |
| |
| // Enable feedback for send side bandwidth estimation. |
| // See |
| // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| // for details. |
| bool transport_cc = false; |
| |
| // See NackConfig for description. |
| NackConfig nack; |
| |
| // RTP header extensions used for the received stream. |
| std::vector<RtpExtension> extensions; |
| } rtp; |
| |
| Transport* rtcp_send_transport = nullptr; |
| |
| // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- |
| // level components. |
| // TODO(solenberg): Remove when VoiceEngine channels are created outside |
| // of Call. |
| int voe_channel_id = -1; |
| |
| // Identifier for an A/V synchronization group. Empty string to disable. |
| // TODO(pbos): Synchronize streams in a sync group, not just one video |
| // stream to one audio stream. Tracked by issue webrtc:4762. |
| std::string sync_group; |
| |
| // Decoder specifications for every payload type that we can receive. |
| std::map<int, SdpAudioFormat> decoder_map; |
| |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; |
| }; |
| |
| // 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; |
| |
| virtual Stats GetStats() const = 0; |
| // TODO(solenberg): Remove, once AudioMonitor is gone. |
| virtual int GetOutputLevel() const = 0; |
| |
| // Sets an audio sink that receives unmixed audio from the receive stream. |
| // Ownership of the sink is passed to the stream and can be used by the |
| // caller to do lifetime management (i.e. when the sink's dtor is called). |
| // Only one sink can be set and passing a null sink clears an existing one. |
| // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
| // to stream through this sink. In practice, this happens if mixed audio |
| // is being pulled+rendered and/or if audio is being pulled for the purposes |
| // of feeding to the AEC. |
| virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; |
| |
| // Sets playback gain of the stream, applied when mixing, and thus after it |
| // is potentially forwarded to any attached AudioSinkInterface implementation. |
| virtual void SetGain(float gain) = 0; |
| |
| virtual std::vector<RtpSource> GetSources() const = 0; |
| |
| protected: |
| virtual ~AudioReceiveStream() {} |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ |