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Fredrik Solenberg23fba1f2015-04-29 15:24:01 +02001/*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
kjellandera69d9732016-08-31 07:33:05 -070011#ifndef WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
12#define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020013
Fredrik Solenberg04f49312015-06-08 13:04:56 +020014#include <map>
kwibergfffa42b2016-02-23 10:46:32 -080015#include <memory>
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020016#include <string>
17#include <vector>
18
ossu29b1a8d2016-06-13 07:34:51 -070019#include "webrtc/base/scoped_ref_ptr.h"
20#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
pbos1ba8d392016-05-01 20:18:34 -070021#include "webrtc/common_types.h"
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020022#include "webrtc/config.h"
solenbergcf18b342015-10-01 08:13:42 -070023#include "webrtc/transport.h"
Fredrik Solenberg04f49312015-06-08 13:04:56 +020024#include "webrtc/typedefs.h"
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020025
26namespace webrtc {
Tommif888bb52015-12-12 01:37:01 +010027class AudioSinkInterface;
Fredrik Solenberg04f49312015-06-08 13:04:56 +020028
Fredrik Solenberga4527c82015-12-03 13:06:20 +010029// WORK IN PROGRESS
30// This class is under development and is not yet intended for for use outside
31// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
32// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
33
pbos1ba8d392016-05-01 20:18:34 -070034class AudioReceiveStream {
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020035 public:
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020036 struct Stats {
37 uint32_t remote_ssrc = 0;
38 int64_t bytes_rcvd = 0;
39 uint32_t packets_rcvd = 0;
40 uint32_t packets_lost = 0;
41 float fraction_lost = 0.0f;
42 std::string codec_name;
43 uint32_t ext_seqnum = 0;
44 uint32_t jitter_ms = 0;
45 uint32_t jitter_buffer_ms = 0;
46 uint32_t jitter_buffer_preferred_ms = 0;
47 uint32_t delay_estimate_ms = 0;
48 int32_t audio_level = -1;
49 float expand_rate = 0.0f;
50 float speech_expand_rate = 0.0f;
51 float secondary_decoded_rate = 0.0f;
52 float accelerate_rate = 0.0f;
53 float preemptive_expand_rate = 0.0f;
54 int32_t decoding_calls_to_silence_generator = 0;
55 int32_t decoding_calls_to_neteq = 0;
56 int32_t decoding_normal = 0;
57 int32_t decoding_plc = 0;
58 int32_t decoding_cng = 0;
59 int32_t decoding_plc_cng = 0;
henrik.lundin63489782016-09-20 01:47:12 -070060 int32_t decoding_muted_output = 0;
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020061 int64_t capture_start_ntp_time_ms = 0;
62 };
Fredrik Solenberg04f49312015-06-08 13:04:56 +020063
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020064 struct Config {
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020065 std::string ToString() const;
66
67 // Receive-stream specific RTP settings.
68 struct Rtp {
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020069 std::string ToString() const;
70
71 // Synchronization source (stream identifier) to be received.
Fredrik Solenberg04f49312015-06-08 13:04:56 +020072 uint32_t remote_ssrc = 0;
73
74 // Sender SSRC used for sending RTCP (such as receiver reports).
75 uint32_t local_ssrc = 0;
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020076
Stefan Holmer3842c5c2016-01-12 13:55:00 +010077 // Enable feedback for send side bandwidth estimation.
78 // See
79 // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
80 // for details.
81 bool transport_cc = false;
82
solenberg8189b022016-06-14 12:13:00 -070083 // See NackConfig for description.
84 NackConfig nack;
85
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +020086 // RTP header extensions used for the received stream.
87 std::vector<RtpExtension> extensions;
88 } rtp;
Fredrik Solenberg04f49312015-06-08 13:04:56 +020089
solenbergcf18b342015-10-01 08:13:42 -070090 Transport* rtcp_send_transport = nullptr;
91
92 // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
93 // level components.
94 // TODO(solenberg): Remove when VoiceEngine channels are created outside
95 // of Call.
pbos8fc7fa72015-07-15 08:02:58 -070096 int voe_channel_id = -1;
97
98 // Identifier for an A/V synchronization group. Empty string to disable.
99 // TODO(pbos): Synchronize streams in a sync group, not just one video
100 // stream to one audio stream. Tracked by issue webrtc:4762.
101 std::string sync_group;
102
Fredrik Solenberg04f49312015-06-08 13:04:56 +0200103 // Decoders for every payload that we can receive. Call owns the
104 // AudioDecoder instances once the Config is submitted to
105 // Call::CreateReceiveStream().
106 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
107 std::map<uint8_t, AudioDecoder*> decoder_map;
ossu29b1a8d2016-06-13 07:34:51 -0700108
109 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +0200110 };
111
pbos1ba8d392016-05-01 20:18:34 -0700112 // Starts stream activity.
113 // When a stream is active, it can receive, process and deliver packets.
114 virtual void Start() = 0;
115 // Stops stream activity.
116 // When a stream is stopped, it can't receive, process or deliver packets.
117 virtual void Stop() = 0;
118
Fredrik Solenberg04f49312015-06-08 13:04:56 +0200119 virtual Stats GetStats() const = 0;
Tommif888bb52015-12-12 01:37:01 +0100120
121 // Sets an audio sink that receives unmixed audio from the receive stream.
122 // Ownership of the sink is passed to the stream and can be used by the
123 // caller to do lifetime management (i.e. when the sink's dtor is called).
deadbeef884f5852016-01-15 09:20:04 -0800124 // Only one sink can be set and passing a null sink clears an existing one.
Tommif888bb52015-12-12 01:37:01 +0100125 // NOTE: Audio must still somehow be pulled through AudioTransport for audio
126 // to stream through this sink. In practice, this happens if mixed audio
127 // is being pulled+rendered and/or if audio is being pulled for the purposes
128 // of feeding to the AEC.
kwibergfffa42b2016-02-23 10:46:32 -0800129 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
pbos1ba8d392016-05-01 20:18:34 -0700130
solenberg217fb662016-06-17 08:30:54 -0700131 // Sets playback gain of the stream, applied when mixing, and thus after it
132 // is potentially forwarded to any attached AudioSinkInterface implementation.
133 virtual void SetGain(float gain) = 0;
134
pbos1ba8d392016-05-01 20:18:34 -0700135 protected:
136 virtual ~AudioReceiveStream() {}
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +0200137};
Fredrik Solenberg23fba1f2015-04-29 15:24:01 +0200138} // namespace webrtc
139
kjellandera69d9732016-08-31 07:33:05 -0700140#endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_