Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(855)

Unified Diff: webrtc/modules/audio_device/audio_device_buffer.h

Issue 2466613002: Adds thread safety annotations to the AudioDeviceBuffer class (Closed)
Patch Set: Minor thread change for Windows Created 4 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/modules/audio_device/audio_device_buffer.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/audio_device/audio_device_buffer.h
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index 7e9f3e3eec30105a5971727002fe9b8bb39d87eb..f5f8f1d2207d0be59cb85a761081a70f10850b75 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -12,8 +12,8 @@
#define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#include "webrtc/base/buffer.h"
-#include "webrtc/base/criticalsection.h"
#include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
@@ -103,29 +103,41 @@ class AudioDeviceBuffer {
void ResetRecStats();
void ResetPlayStats();
- // Ensures that methods are called on the same thread as the thread that
- // creates this object.
- rtc::ThreadChecker thread_checker_;
+ // This object lives on the main (creating) thread and most methods are
+ // called on that same thread. When audio has started some methods will be
+ // called on either a native audio thread for playout or a native thread for
+ // recording. Some members are not annotated since they are "protected by
+ // design" and adding e.g. a race checker can cause failuries for very few
+ // edge cases and it is IMHO not worth the risk to use them in this class.
+ // TODO(henrika): see if it is possible to refactor and annotate all members.
- // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
- // and it must outlive this object.
- AudioTransport* audio_transport_cb_;
+ // Main thread on which this object is created.
+ rtc::ThreadChecker main_thread_checker_;
+
+ // Native (platform specific) audio thread driving the playout side.
+ rtc::ThreadChecker playout_thread_checker_;
- // TODO(henrika): given usage of thread checker, it should be possible to
- // remove all locks in this class.
- rtc::CriticalSection lock_;
- rtc::CriticalSection lock_cb_;
+ // Native (platform specific) audio thread driving the recording side.
+ rtc::ThreadChecker recording_thread_checker_;
// Task queue used to invoke LogStats() periodically. Tasks are executed on a
// worker thread but it does not necessarily have to be the same thread for
// each task.
rtc::TaskQueue task_queue_;
- // Keeps track of if playout/recording are active or not. A combination
- // of these states are used to determine when to start and stop the timer.
- // Only used on the creating thread and not used to control any media flow.
- bool playing_;
- bool recording_;
+ // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+ // and it must outlive this object. It is not possible to change this member
+ // while any media is active. It is possible to start media without calling
+ // RegisterAudioCallback() but that will lead to ignored audio callbacks in
+ // both directions where native audio will be acive but no audio samples will
+ // be transported.
+ AudioTransport* audio_transport_cb_;
+
+ // The members below that are not annotated are protected by design. They are
+ // all set on the main thread (verified by |main_thread_checker_|) and then
+ // read on either the playout or recording audio thread. But, media will never
+ // be active when the member is set; hence no conflict exists. It is too
+ // complex to ensure and verify that this is actually the case.
// Sample rate in Hertz.
uint32_t rec_sample_rate_;
@@ -135,93 +147,88 @@ class AudioDeviceBuffer {
size_t rec_channels_;
size_t play_channels_;
- // Number of bytes per audio sample (2 or 4).
- size_t rec_bytes_per_sample_;
- size_t play_bytes_per_sample_;
+ // Keeps track of if playout/recording are active or not. A combination
+ // of these states are used to determine when to start and stop the timer.
+ // Only used on the creating thread and not used to control any media flow.
+ bool playing_ ACCESS_ON(main_thread_checker_);
+ bool recording_ ACCESS_ON(main_thread_checker_);
- // Byte buffer used for recorded audio samples. Size can be changed
+ // Buffer used for audio samples to be played out. Size can be changed
// dynamically.
- rtc::Buffer rec_buffer_;
+ rtc::Buffer play_buffer_ ACCESS_ON(playout_thread_checker_);
- // Buffer used for audio samples to be played out. Size can be changed
+ // Byte buffer used for recorded audio samples. Size can be changed
// dynamically.
- rtc::Buffer play_buffer_;
+ rtc::Buffer rec_buffer_ ACCESS_ON(recording_thread_checker_);
// AGC parameters.
+#if !defined(WEBRTC_WIN)
+ uint32_t current_mic_level_ ACCESS_ON(recording_thread_checker_);
+#else
+ // Windows uses a dedicated thread for volume APIs.
uint32_t current_mic_level_;
- uint32_t new_mic_level_;
+#endif
+ uint32_t new_mic_level_ ACCESS_ON(recording_thread_checker_);
// Contains true of a key-press has been detected.
- bool typing_status_;
+ bool typing_status_ ACCESS_ON(recording_thread_checker_);
// Delay values used by the AEC.
- int play_delay_ms_;
- int rec_delay_ms_;
+ int play_delay_ms_ ACCESS_ON(recording_thread_checker_);
+ int rec_delay_ms_ ACCESS_ON(recording_thread_checker_);
// Contains a clock-drift measurement.
- int clock_drift_;
+ int clock_drift_ ACCESS_ON(recording_thread_checker_);
// Counts number of times LogStats() has been called.
- size_t num_stat_reports_;
+ size_t num_stat_reports_ ACCESS_ON(task_queue_);
// Total number of recording callbacks where the source provides 10ms audio
// data each time.
- uint64_t rec_callbacks_;
+ uint64_t rec_callbacks_ ACCESS_ON(task_queue_);
// Total number of recording callbacks stored at the last timer task.
- uint64_t last_rec_callbacks_;
+ uint64_t last_rec_callbacks_ ACCESS_ON(task_queue_);
// Total number of playback callbacks where the sink asks for 10ms audio
// data each time.
- uint64_t play_callbacks_;
+ uint64_t play_callbacks_ ACCESS_ON(task_queue_);
// Total number of playout callbacks stored at the last timer task.
- uint64_t last_play_callbacks_;
+ uint64_t last_play_callbacks_ ACCESS_ON(task_queue_);
// Total number of recorded audio samples.
- uint64_t rec_samples_;
+ uint64_t rec_samples_ ACCESS_ON(task_queue_);
// Total number of recorded samples stored at the previous timer task.
- uint64_t last_rec_samples_;
+ uint64_t last_rec_samples_ ACCESS_ON(task_queue_);
// Total number of played audio samples.
- uint64_t play_samples_;
+ uint64_t play_samples_ ACCESS_ON(task_queue_);
// Total number of played samples stored at the previous timer task.
- uint64_t last_play_samples_;
-
- // Time stamp of last timer task (drives logging).
- uint64_t last_timer_task_time_;
-
- // Time stamp of last playout callback.
- uint64_t last_playout_time_;
-
- // An array where the position corresponds to time differences (in
- // milliseconds) between two successive playout callbacks, and the stored
- // value is the number of times a given time difference was found.
- // Writing to the array is done without a lock since it is only read once at
- // destruction when no audio is running.
- uint32_t playout_diff_times_[kMaxDeltaTimeInMs + 1] = {0};
+ uint64_t last_play_samples_ ACCESS_ON(task_queue_);
// Contains max level (max(abs(x))) of recorded audio packets over the last
// 10 seconds where a new measurement is done twice per second. The level
- // is reset to zero at each call to LogStats(). Only modified on the task
- // queue thread.
- int16_t max_rec_level_;
+ // is reset to zero at each call to LogStats().
+ int16_t max_rec_level_ ACCESS_ON(task_queue_);
// Contains max level of recorded audio packets over the last 10 seconds
// where a new measurement is done twice per second.
- int16_t max_play_level_;
+ int16_t max_play_level_ ACCESS_ON(task_queue_);
+
+ // Time stamp of last timer task (drives logging).
+ uint64_t last_timer_task_time_ ACCESS_ON(task_queue_);
// Counts number of audio callbacks modulo 50 to create a signal when
// a new storage of audio stats shall be done.
- // Only updated on the OS-specific audio thread that drives audio.
- int16_t rec_stat_count_;
- int16_t play_stat_count_;
+ int16_t rec_stat_count_ ACCESS_ON(recording_thread_checker_);
+ int16_t play_stat_count_ ACCESS_ON(playout_thread_checker_);
// Time stamps of when playout and recording starts.
- uint64_t play_start_time_;
- uint64_t rec_start_time_;
+ uint64_t play_start_time_ ACCESS_ON(main_thread_checker_);
+ uint64_t rec_start_time_ ACCESS_ON(main_thread_checker_);
// Set to true at construction and modified to false as soon as one audio-
// level estimate larger than zero is detected.
« no previous file with comments | « no previous file | webrtc/modules/audio_device/audio_device_buffer.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698