140 lines
4.7 KiB
C++
140 lines
4.7 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
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#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
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#include "webrtc/common_types.h"
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#include "webrtc/config.h"
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#include "webrtc/transport.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioSinkInterface;
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
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class AudioReceiveStream {
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public:
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struct Stats {
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uint32_t remote_ssrc = 0;
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int64_t bytes_rcvd = 0;
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uint32_t packets_rcvd = 0;
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uint32_t packets_lost = 0;
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float fraction_lost = 0.0f;
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std::string codec_name;
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uint32_t ext_seqnum = 0;
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uint32_t jitter_ms = 0;
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uint32_t jitter_buffer_ms = 0;
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uint32_t jitter_buffer_preferred_ms = 0;
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uint32_t delay_estimate_ms = 0;
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int32_t audio_level = -1;
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float expand_rate = 0.0f;
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float speech_expand_rate = 0.0f;
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float secondary_decoded_rate = 0.0f;
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float accelerate_rate = 0.0f;
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float preemptive_expand_rate = 0.0f;
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int32_t decoding_calls_to_silence_generator = 0;
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int32_t decoding_calls_to_neteq = 0;
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int32_t decoding_normal = 0;
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int32_t decoding_plc = 0;
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int32_t decoding_cng = 0;
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int32_t decoding_plc_cng = 0;
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int64_t capture_start_ntp_time_ms = 0;
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};
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struct Config {
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std::string ToString() const;
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// Receive-stream specific RTP settings.
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struct Rtp {
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std::string ToString() const;
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// Synchronization source (stream identifier) to be received.
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uint32_t remote_ssrc = 0;
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// Sender SSRC used for sending RTCP (such as receiver reports).
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uint32_t local_ssrc = 0;
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// Enable feedback for send side bandwidth estimation.
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// See
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
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// for details.
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bool transport_cc = false;
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// See NackConfig for description.
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NackConfig nack;
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// RTP header extensions used for the received stream.
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std::vector<RtpExtension> extensions;
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} rtp;
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Transport* rtcp_send_transport = nullptr;
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// Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
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// level components.
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// TODO(solenberg): Remove when VoiceEngine channels are created outside
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// of Call.
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int voe_channel_id = -1;
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// Identifier for an A/V synchronization group. Empty string to disable.
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// TODO(pbos): Synchronize streams in a sync group, not just one video
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// stream to one audio stream. Tracked by issue webrtc:4762.
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std::string sync_group;
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// Decoders for every payload that we can receive. Call owns the
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// AudioDecoder instances once the Config is submitted to
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// Call::CreateReceiveStream().
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// TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
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std::map<uint8_t, AudioDecoder*> decoder_map;
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
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};
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// Starts stream activity.
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// When a stream is active, it can receive, process and deliver packets.
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virtual void Start() = 0;
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// Stops stream activity.
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// When a stream is stopped, it can't receive, process or deliver packets.
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virtual void Stop() = 0;
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virtual Stats GetStats() const = 0;
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// Sets an audio sink that receives unmixed audio from the receive stream.
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// Ownership of the sink is passed to the stream and can be used by the
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// caller to do lifetime management (i.e. when the sink's dtor is called).
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// Only one sink can be set and passing a null sink clears an existing one.
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// NOTE: Audio must still somehow be pulled through AudioTransport for audio
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// to stream through this sink. In practice, this happens if mixed audio
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// is being pulled+rendered and/or if audio is being pulled for the purposes
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// of feeding to the AEC.
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virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
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// Sets playback gain of the stream, applied when mixing, and thus after it
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// is potentially forwarded to any attached AudioSinkInterface implementation.
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virtual void SetGain(float gain) = 0;
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protected:
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virtual ~AudioReceiveStream() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
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