Move webrtc/audio_*.h to webrtc/api/call

webrtc/audio_receive_stream.h

-/*
- *  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_AUDIO_RECEIVE_STREAM_H_
-#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
-
-#include <map>
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/config.h"
-#include "webrtc/transport.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;
-    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;
-    float expand_rate = 0.0f;
-    float speech_expand_rate = 0.0f;
-    float secondary_decoded_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;
-    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;
-
-    // Decoders for every payload that we can receive. Call owns the
-    // AudioDecoder instances once the Config is submitted to
-    // Call::CreateReceiveStream().
-    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
-    std::map<uint8_t, AudioDecoder*> 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;
-
-  // 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;
-
- protected:
-  virtual ~AudioReceiveStream() {}
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_