4f4ec0a927
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: a457752f4a
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
102 lines
3.1 KiB
C++
102 lines
3.1 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 <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 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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// 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* receive_transport = nullptr;
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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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// TODO(pbos): Remove config option once combined A/V BWE is always on.
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bool combined_audio_video_bwe = false;
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};
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virtual Stats GetStats() const = 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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