Line data Source code
1 : /*
2 : * Copyright (c) 2012 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 :
11 : #ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
12 : #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
13 :
14 : #include "webrtc/base/buffer.h"
15 : #include "webrtc/base/task_queue.h"
16 : #include "webrtc/base/thread_annotations.h"
17 : #include "webrtc/base/thread_checker.h"
18 : #include "webrtc/modules/audio_device/include/audio_device.h"
19 : #include "webrtc/system_wrappers/include/file_wrapper.h"
20 : #include "webrtc/typedefs.h"
21 :
22 : namespace webrtc {
23 : // Delta times between two successive playout callbacks are limited to this
24 : // value before added to an internal array.
25 : const size_t kMaxDeltaTimeInMs = 500;
26 : // TODO(henrika): remove when no longer used by external client.
27 : const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
28 :
29 : class AudioDeviceObserver;
30 :
31 : class AudioDeviceBuffer {
32 : public:
33 : enum LogState {
34 : LOG_START = 0,
35 : LOG_STOP,
36 : LOG_ACTIVE,
37 : };
38 :
39 : AudioDeviceBuffer();
40 : virtual ~AudioDeviceBuffer();
41 :
42 0 : void SetId(uint32_t id) {};
43 : int32_t RegisterAudioCallback(AudioTransport* audio_callback);
44 :
45 : void StartPlayout();
46 : void StartRecording();
47 : void StopPlayout();
48 : void StopRecording();
49 :
50 : int32_t SetRecordingSampleRate(uint32_t fsHz);
51 : int32_t SetPlayoutSampleRate(uint32_t fsHz);
52 : int32_t RecordingSampleRate() const;
53 : int32_t PlayoutSampleRate() const;
54 :
55 : int32_t SetRecordingChannels(size_t channels);
56 : int32_t SetPlayoutChannels(size_t channels);
57 : size_t RecordingChannels() const;
58 : size_t PlayoutChannels() const;
59 : int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
60 : int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
61 :
62 : virtual int32_t SetRecordedBuffer(const void* audio_buffer,
63 : size_t samples_per_channel);
64 : int32_t SetCurrentMicLevel(uint32_t level);
65 : virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
66 : virtual int32_t DeliverRecordedData();
67 : uint32_t NewMicLevel() const;
68 :
69 : virtual int32_t RequestPlayoutData(size_t samples_per_channel);
70 : virtual int32_t GetPlayoutData(void* audio_buffer);
71 :
72 : // TODO(henrika): these methods should not be used and does not contain any
73 : // valid implementation. Investigate the possibility to either remove them
74 : // or add a proper implementation if needed.
75 : int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
76 : int32_t StopInputFileRecording();
77 : int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
78 : int32_t StopOutputFileRecording();
79 :
80 : int32_t SetTypingStatus(bool typing_status);
81 :
82 : private:
83 : // Starts/stops periodic logging of audio stats.
84 : void StartPeriodicLogging();
85 : void StopPeriodicLogging();
86 :
87 : // Called periodically on the internal thread created by the TaskQueue.
88 : // Updates some stats but dooes it on the task queue to ensure that access of
89 : // members is serialized hence avoiding usage of locks.
90 : // state = LOG_START => members are initialized and the timer starts.
91 : // state = LOG_STOP => no logs are printed and the timer stops.
92 : // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
93 : void LogStats(LogState state);
94 :
95 : // Updates counters in each play/record callback but does it on the task
96 : // queue to ensure that they can be read by LogStats() without any locks since
97 : // each task is serialized by the task queue.
98 : void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
99 : void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
100 :
101 : // Clears all members tracking stats for recording and playout.
102 : // These methods both run on the task queue.
103 : void ResetRecStats();
104 : void ResetPlayStats();
105 :
106 : // This object lives on the main (creating) thread and most methods are
107 : // called on that same thread. When audio has started some methods will be
108 : // called on either a native audio thread for playout or a native thread for
109 : // recording. Some members are not annotated since they are "protected by
110 : // design" and adding e.g. a race checker can cause failuries for very few
111 : // edge cases and it is IMHO not worth the risk to use them in this class.
112 : // TODO(henrika): see if it is possible to refactor and annotate all members.
113 :
114 : // Main thread on which this object is created.
115 : rtc::ThreadChecker main_thread_checker_;
116 :
117 : // Native (platform specific) audio thread driving the playout side.
118 : rtc::ThreadChecker playout_thread_checker_;
119 :
120 : // Native (platform specific) audio thread driving the recording side.
121 : rtc::ThreadChecker recording_thread_checker_;
122 :
123 : // Task queue used to invoke LogStats() periodically. Tasks are executed on a
124 : // worker thread but it does not necessarily have to be the same thread for
125 : // each task.
126 : rtc::TaskQueue task_queue_;
127 :
128 : // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
129 : // and it must outlive this object. It is not possible to change this member
130 : // while any media is active. It is possible to start media without calling
131 : // RegisterAudioCallback() but that will lead to ignored audio callbacks in
132 : // both directions where native audio will be acive but no audio samples will
133 : // be transported.
134 : AudioTransport* audio_transport_cb_;
135 :
136 : // The members below that are not annotated are protected by design. They are
137 : // all set on the main thread (verified by |main_thread_checker_|) and then
138 : // read on either the playout or recording audio thread. But, media will never
139 : // be active when the member is set; hence no conflict exists. It is too
140 : // complex to ensure and verify that this is actually the case.
141 :
142 : // Sample rate in Hertz.
143 : uint32_t rec_sample_rate_;
144 : uint32_t play_sample_rate_;
145 :
146 : // Number of audio channels.
147 : size_t rec_channels_;
148 : size_t play_channels_;
149 :
150 : // Keeps track of if playout/recording are active or not. A combination
151 : // of these states are used to determine when to start and stop the timer.
152 : // Only used on the creating thread and not used to control any media flow.
153 : bool playing_ ACCESS_ON(main_thread_checker_);
154 : bool recording_ ACCESS_ON(main_thread_checker_);
155 :
156 : // Buffer used for audio samples to be played out. Size can be changed
157 : // dynamically. The 16-bit samples are interleaved, hence the size is
158 : // proportional to the number of channels.
159 : rtc::BufferT<int16_t> play_buffer_ ACCESS_ON(playout_thread_checker_);
160 :
161 : // Byte buffer used for recorded audio samples. Size can be changed
162 : // dynamically.
163 : rtc::BufferT<int16_t> rec_buffer_ ACCESS_ON(recording_thread_checker_);
164 :
165 : // AGC parameters.
166 : #if !defined(WEBRTC_WIN)
167 : uint32_t current_mic_level_ ACCESS_ON(recording_thread_checker_);
168 : #else
169 : // Windows uses a dedicated thread for volume APIs.
170 : uint32_t current_mic_level_;
171 : #endif
172 : uint32_t new_mic_level_ ACCESS_ON(recording_thread_checker_);
173 :
174 : // Contains true of a key-press has been detected.
175 : bool typing_status_ ACCESS_ON(recording_thread_checker_);
176 :
177 : // Delay values used by the AEC.
178 : int play_delay_ms_ ACCESS_ON(recording_thread_checker_);
179 : int rec_delay_ms_ ACCESS_ON(recording_thread_checker_);
180 :
181 : // Contains a clock-drift measurement.
182 : int clock_drift_ ACCESS_ON(recording_thread_checker_);
183 :
184 : // Counts number of times LogStats() has been called.
185 : size_t num_stat_reports_ ACCESS_ON(task_queue_);
186 :
187 : // Total number of recording callbacks where the source provides 10ms audio
188 : // data each time.
189 : uint64_t rec_callbacks_ ACCESS_ON(task_queue_);
190 :
191 : // Total number of recording callbacks stored at the last timer task.
192 : uint64_t last_rec_callbacks_ ACCESS_ON(task_queue_);
193 :
194 : // Total number of playback callbacks where the sink asks for 10ms audio
195 : // data each time.
196 : uint64_t play_callbacks_ ACCESS_ON(task_queue_);
197 :
198 : // Total number of playout callbacks stored at the last timer task.
199 : uint64_t last_play_callbacks_ ACCESS_ON(task_queue_);
200 :
201 : // Total number of recorded audio samples.
202 : uint64_t rec_samples_ ACCESS_ON(task_queue_);
203 :
204 : // Total number of recorded samples stored at the previous timer task.
205 : uint64_t last_rec_samples_ ACCESS_ON(task_queue_);
206 :
207 : // Total number of played audio samples.
208 : uint64_t play_samples_ ACCESS_ON(task_queue_);
209 :
210 : // Total number of played samples stored at the previous timer task.
211 : uint64_t last_play_samples_ ACCESS_ON(task_queue_);
212 :
213 : // Contains max level (max(abs(x))) of recorded audio packets over the last
214 : // 10 seconds where a new measurement is done twice per second. The level
215 : // is reset to zero at each call to LogStats().
216 : int16_t max_rec_level_ ACCESS_ON(task_queue_);
217 :
218 : // Contains max level of recorded audio packets over the last 10 seconds
219 : // where a new measurement is done twice per second.
220 : int16_t max_play_level_ ACCESS_ON(task_queue_);
221 :
222 : // Time stamp of last timer task (drives logging).
223 : int64_t last_timer_task_time_ ACCESS_ON(task_queue_);
224 :
225 : // Counts number of audio callbacks modulo 50 to create a signal when
226 : // a new storage of audio stats shall be done.
227 : int16_t rec_stat_count_ ACCESS_ON(recording_thread_checker_);
228 : int16_t play_stat_count_ ACCESS_ON(playout_thread_checker_);
229 :
230 : // Time stamps of when playout and recording starts.
231 : int64_t play_start_time_ ACCESS_ON(main_thread_checker_);
232 : int64_t rec_start_time_ ACCESS_ON(main_thread_checker_);
233 :
234 : // Set to true at construction and modified to false as soon as one audio-
235 : // level estimate larger than zero is detected.
236 : bool only_silence_recorded_;
237 :
238 : // Set to true when logging of audio stats is enabled for the first time in
239 : // StartPeriodicLogging() and set to false by StopPeriodicLogging().
240 : // Setting this member to false prevents (possiby invalid) log messages from
241 : // being printed in the LogStats() task.
242 : bool log_stats_ ACCESS_ON(task_queue_);
243 : };
244 :
245 : } // namespace webrtc
246 :
247 : #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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