3d7db263b9
VoENetwork is kept for now, but is not really used anylonger. webrtcvoiceengine is changed to have the same behavior for unsignaled ssrc as video has, which is reflected by disabling one test case and this will be discussed and followed up. BUG=webrtc:5079 TBR=tommi Review-Url: https://codereview.webrtc.org/1909333002 Cr-Commit-Position: refs/heads/master@{#12555}
121 lines
4.0 KiB
C++
121 lines
4.0 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/config.h"
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#include "webrtc/stream.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 AudioDecoder;
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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 : public ReceiveStream {
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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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// 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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};
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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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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
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