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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/audio/audio_receive_stream.h" | 11 #include "webrtc/audio/audio_receive_stream.h" |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 | 14 |
| 15 #include "webrtc/audio/audio_state.h" |
15 #include "webrtc/audio/conversion.h" | 16 #include "webrtc/audio/conversion.h" |
16 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/logging.h" | 18 #include "webrtc/base/logging.h" |
18 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" | 19 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" |
19 #include "webrtc/system_wrappers/include/tick_util.h" | 20 #include "webrtc/system_wrappers/include/tick_util.h" |
20 #include "webrtc/voice_engine/include/voe_base.h" | 21 #include "webrtc/voice_engine/include/voe_base.h" |
21 #include "webrtc/voice_engine/include/voe_codec.h" | 22 #include "webrtc/voice_engine/include/voe_codec.h" |
22 #include "webrtc/voice_engine/include/voe_neteq_stats.h" | 23 #include "webrtc/voice_engine/include/voe_neteq_stats.h" |
23 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 24 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
24 #include "webrtc/voice_engine/include/voe_video_sync.h" | 25 #include "webrtc/voice_engine/include/voe_video_sync.h" |
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53 ss << ", sync_group: " << sync_group; | 54 ss << ", sync_group: " << sync_group; |
54 } | 55 } |
55 ss << ", combined_audio_video_bwe: " | 56 ss << ", combined_audio_video_bwe: " |
56 << (combined_audio_video_bwe ? "true" : "false"); | 57 << (combined_audio_video_bwe ? "true" : "false"); |
57 ss << '}'; | 58 ss << '}'; |
58 return ss.str(); | 59 return ss.str(); |
59 } | 60 } |
60 | 61 |
61 namespace internal { | 62 namespace internal { |
62 AudioReceiveStream::AudioReceiveStream( | 63 AudioReceiveStream::AudioReceiveStream( |
63 RemoteBitrateEstimator* remote_bitrate_estimator, | 64 RemoteBitrateEstimator* remote_bitrate_estimator, |
64 const webrtc::AudioReceiveStream::Config& config, | 65 const webrtc::AudioReceiveStream::Config& config, |
65 VoiceEngine* voice_engine) | 66 const rtc::scoped_refptr<webrtc::AudioState>& audio_state) |
66 : remote_bitrate_estimator_(remote_bitrate_estimator), | 67 : remote_bitrate_estimator_(remote_bitrate_estimator), |
67 config_(config), | 68 config_(config), |
68 voice_engine_(voice_engine), | 69 audio_state_(audio_state), |
69 voe_base_(voice_engine), | |
70 rtp_header_parser_(RtpHeaderParser::Create()) { | 70 rtp_header_parser_(RtpHeaderParser::Create()) { |
71 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); | 71 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
72 RTC_DCHECK(config.voe_channel_id != -1); | 72 RTC_DCHECK_NE(config_.voe_channel_id, -1); |
73 RTC_DCHECK(remote_bitrate_estimator_ != nullptr); | 73 RTC_DCHECK(remote_bitrate_estimator_); |
74 RTC_DCHECK(voice_engine_ != nullptr); | 74 RTC_DCHECK(audio_state_.get()); |
75 RTC_DCHECK(rtp_header_parser_ != nullptr); | 75 RTC_DCHECK(rtp_header_parser_); |
76 for (const auto& ext : config.rtp.extensions) { | 76 for (const auto& ext : config.rtp.extensions) { |
77 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 77 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
78 RTC_DCHECK_GE(ext.id, 1); | 78 RTC_DCHECK_GE(ext.id, 1); |
79 RTC_DCHECK_LE(ext.id, 14); | 79 RTC_DCHECK_LE(ext.id, 14); |
80 if (ext.name == RtpExtension::kAudioLevel) { | 80 if (ext.name == RtpExtension::kAudioLevel) { |
81 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | 81 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
82 kRtpExtensionAudioLevel, ext.id)); | 82 kRtpExtensionAudioLevel, ext.id)); |
83 } else if (ext.name == RtpExtension::kAbsSendTime) { | 83 } else if (ext.name == RtpExtension::kAbsSendTime) { |
84 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | 84 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
85 kRtpExtensionAbsoluteSendTime, ext.id)); | 85 kRtpExtensionAbsoluteSendTime, ext.id)); |
86 } else if (ext.name == RtpExtension::kTransportSequenceNumber) { | 86 } else if (ext.name == RtpExtension::kTransportSequenceNumber) { |
87 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | 87 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
88 kRtpExtensionTransportSequenceNumber, ext.id)); | 88 kRtpExtensionTransportSequenceNumber, ext.id)); |
89 } else { | 89 } else { |
90 RTC_NOTREACHED() << "Unsupported RTP extension."; | 90 RTC_NOTREACHED() << "Unsupported RTP extension."; |
91 } | 91 } |
92 } | 92 } |
93 } | 93 } |
94 | 94 |
95 AudioReceiveStream::~AudioReceiveStream() { | 95 AudioReceiveStream::~AudioReceiveStream() { |
96 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 96 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
97 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); | 97 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
98 } | 98 } |
99 | 99 |
100 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { | 100 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { |
101 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 101 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
102 webrtc::AudioReceiveStream::Stats stats; | 102 webrtc::AudioReceiveStream::Stats stats; |
103 stats.remote_ssrc = config_.rtp.remote_ssrc; | 103 stats.remote_ssrc = config_.rtp.remote_ssrc; |
104 ScopedVoEInterface<VoECodec> codec(voice_engine_); | 104 internal::AudioState* audio_state = |
105 ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_); | 105 static_cast<internal::AudioState*>(audio_state_.get()); |
106 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_); | 106 VoiceEngine* voice_engine = audio_state->voice_engine(); |
107 ScopedVoEInterface<VoEVideoSync> sync(voice_engine_); | 107 ScopedVoEInterface<VoECodec> codec(voice_engine); |
108 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_); | 108 ScopedVoEInterface<VoENetEqStats> neteq(voice_engine); |
| 109 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine); |
| 110 ScopedVoEInterface<VoEVideoSync> sync(voice_engine); |
| 111 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine); |
109 unsigned int ssrc = 0; | 112 unsigned int ssrc = 0; |
110 webrtc::CallStatistics call_stats = {0}; | 113 webrtc::CallStatistics call_stats = {0}; |
111 webrtc::CodecInst codec_inst = {0}; | 114 webrtc::CodecInst codec_inst = {0}; |
112 // Only collect stats if we have seen some traffic with the SSRC. | 115 // Only collect stats if we have seen some traffic with the SSRC. |
113 if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 || | 116 if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 || |
114 rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 || | 117 rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 || |
115 codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { | 118 codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { |
116 return stats; | 119 return stats; |
117 } | 120 } |
118 | 121 |
119 stats.bytes_rcvd = call_stats.bytesReceived; | 122 stats.bytes_rcvd = call_stats.bytesReceived; |
120 stats.packets_rcvd = call_stats.packetsReceived; | 123 stats.packets_rcvd = call_stats.packetsReceived; |
121 stats.packets_lost = call_stats.cumulativeLost; | 124 stats.packets_lost = call_stats.cumulativeLost; |
122 stats.fraction_lost = Q8ToFloat(call_stats.fractionLost); | 125 stats.fraction_lost = Q8ToFloat(call_stats.fractionLost); |
123 if (codec_inst.pltype != -1) { | 126 if (codec_inst.pltype != -1) { |
124 stats.codec_name = codec_inst.plname; | 127 stats.codec_name = codec_inst.plname; |
125 } | 128 } |
126 stats.ext_seqnum = call_stats.extendedMax; | 129 stats.ext_seqnum = call_stats.extendedMax; |
127 if (codec_inst.plfreq / 1000 > 0) { | 130 if (codec_inst.plfreq / 1000 > 0) { |
128 stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000); | 131 stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000); |
129 } | 132 } |
130 { | 133 { |
131 int jitter_buffer_delay_ms = 0; | 134 int jitter_buffer_delay_ms = 0; |
132 int playout_buffer_delay_ms = 0; | 135 int playout_buffer_delay_ms = 0; |
133 sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms, | 136 sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms, |
134 &playout_buffer_delay_ms); | 137 &playout_buffer_delay_ms); |
135 stats.delay_estimate_ms = | 138 stats.delay_estimate_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms; |
136 jitter_buffer_delay_ms + playout_buffer_delay_ms; | |
137 } | 139 } |
138 { | 140 { |
139 unsigned int level = 0; | 141 unsigned int level = 0; |
140 if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level) | 142 if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level) != |
141 != -1) { | 143 -1) { |
142 stats.audio_level = static_cast<int32_t>(level); | 144 stats.audio_level = static_cast<int32_t>(level); |
143 } | 145 } |
144 } | 146 } |
145 | 147 |
146 webrtc::NetworkStatistics ns = {0}; | 148 webrtc::NetworkStatistics ns = {0}; |
147 if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) { | 149 if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) { |
148 // Get jitter buffer and total delay (alg + jitter + playout) stats. | 150 // Get jitter buffer and total delay (alg + jitter + playout) stats. |
149 stats.jitter_buffer_ms = ns.currentBufferSize; | 151 stats.jitter_buffer_ms = ns.currentBufferSize; |
150 stats.jitter_buffer_preferred_ms = ns.preferredBufferSize; | 152 stats.jitter_buffer_preferred_ms = ns.preferredBufferSize; |
151 stats.expand_rate = Q14ToFloat(ns.currentExpandRate); | 153 stats.expand_rate = Q14ToFloat(ns.currentExpandRate); |
152 stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate); | 154 stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate); |
153 stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate); | 155 stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate); |
154 stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); | 156 stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); |
155 stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); | 157 stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); |
156 } | 158 } |
157 | 159 |
158 webrtc::AudioDecodingCallStats ds; | 160 webrtc::AudioDecodingCallStats ds; |
159 if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) { | 161 if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) { |
160 stats.decoding_calls_to_silence_generator = | 162 stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator; |
161 ds.calls_to_silence_generator; | |
162 stats.decoding_calls_to_neteq = ds.calls_to_neteq; | 163 stats.decoding_calls_to_neteq = ds.calls_to_neteq; |
163 stats.decoding_normal = ds.decoded_normal; | 164 stats.decoding_normal = ds.decoded_normal; |
164 stats.decoding_plc = ds.decoded_plc; | 165 stats.decoding_plc = ds.decoded_plc; |
165 stats.decoding_cng = ds.decoded_cng; | 166 stats.decoding_cng = ds.decoded_cng; |
166 stats.decoding_plc_cng = ds.decoded_plc_cng; | 167 stats.decoding_plc_cng = ds.decoded_plc_cng; |
167 } | 168 } |
168 | 169 |
169 stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; | 170 stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; |
170 | 171 |
171 return stats; | 172 return stats; |
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217 if (packet_time.timestamp >= 0) | 218 if (packet_time.timestamp >= 0) |
218 arrival_time_ms = (packet_time.timestamp + 500) / 1000; | 219 arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
219 size_t payload_size = length - header.headerLength; | 220 size_t payload_size = length - header.headerLength; |
220 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, | 221 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, |
221 header, false); | 222 header, false); |
222 } | 223 } |
223 return true; | 224 return true; |
224 } | 225 } |
225 } // namespace internal | 226 } // namespace internal |
226 } // namespace webrtc | 227 } // namespace webrtc |
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