Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_device/audio_device_buffer.h

@@ -0,0 +1,263 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/rtc_base/buffer.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/task_queue.h"
+#include "webrtc/rtc_base/thread_annotations.h"
+#include "webrtc/rtc_base/thread_checker.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Delta times between two successive playout callbacks are limited to this
+// value before added to an internal array.
+const size_t kMaxDeltaTimeInMs = 500;
+// TODO(henrika): remove when no longer used by external client.
+const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
+
+class AudioDeviceObserver;
+
+class AudioDeviceBuffer {
+ public:
+  enum LogState {
+    LOG_START = 0,
+    LOG_STOP,
+    LOG_ACTIVE,
+  };
+
+  struct Stats {
+    void ResetRecStats() {
+      rec_callbacks = 0;
+      rec_samples = 0;
+      max_rec_level = 0;
+    }
+
+    void ResetPlayStats() {
+      play_callbacks = 0;
+      play_samples = 0;
+      max_play_level = 0;
+    }
+
+    // Total number of recording callbacks where the source provides 10ms audio
+    // data each time.
+    uint64_t rec_callbacks = 0;
+
+    // Total number of playback callbacks where the sink asks for 10ms audio
+    // data each time.
+    uint64_t play_callbacks = 0;
+
+    // Total number of recorded audio samples.
+    uint64_t rec_samples = 0;
+
+    // Total number of played audio samples.
+    uint64_t play_samples = 0;
+
+    // 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().
+    int16_t max_rec_level = 0;
+
+    // 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 = 0;
+  };
+
+  AudioDeviceBuffer();
+  virtual ~AudioDeviceBuffer();
+
+  void SetId(uint32_t id) {};
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback);
+
+  void StartPlayout();
+  void StartRecording();
+  void StopPlayout();
+  void StopRecording();
+
+  int32_t SetRecordingSampleRate(uint32_t fsHz);
+  int32_t SetPlayoutSampleRate(uint32_t fsHz);
+  int32_t RecordingSampleRate() const;
+  int32_t PlayoutSampleRate() const;
+
+  int32_t SetRecordingChannels(size_t channels);
+  int32_t SetPlayoutChannels(size_t channels);
+  size_t RecordingChannels() const;
+  size_t PlayoutChannels() const;
+  int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
+  int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
+
+  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
+                                    size_t samples_per_channel);
+  int32_t SetCurrentMicLevel(uint32_t level);
+  virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
+  virtual int32_t DeliverRecordedData();
+  uint32_t NewMicLevel() const;
+
+  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
+  virtual int32_t GetPlayoutData(void* audio_buffer);
+
+  int32_t SetTypingStatus(bool typing_status);
+
+  // Called on iOS where the native audio layer can be interrupted by other
+  // audio applications. This method can then be used to reset internal
+  // states and detach thread checkers to allow for a new audio session where
+  // native callbacks may come from a new set of I/O threads.
+  void NativeAudioInterrupted();
+
+ private:
+  // Starts/stops periodic logging of audio stats.
+  void StartPeriodicLogging();
+  void StopPeriodicLogging();
+
+  // Called periodically on the internal thread created by the TaskQueue.
+  // Updates some stats but dooes it on the task queue to ensure that access of
+  // members is serialized hence avoiding usage of locks.
+  // state = LOG_START => members are initialized and the timer starts.
+  // state = LOG_STOP => no logs are printed and the timer stops.
+  // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
+  void LogStats(LogState state);
+
+  // Updates counters in each play/record callback. These counters are later
+  // (periodically) read by LogStats() using a lock.
+  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
+  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
+
+  // Clears all members tracking stats for recording and playout.
+  // These methods both run on the task queue.
+  void ResetRecStats();
+  void ResetPlayStats();
+
+  // 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.
+
+  // 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_;
+
+  // Native (platform specific) audio thread driving the recording side.
+  rtc::ThreadChecker recording_thread_checker_;
+
+  rtc::CriticalSection lock_;
+
+  // 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_;
+
+  // 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_;
+  uint32_t play_sample_rate_;
+
+  // Number of audio channels.
+  size_t rec_channels_;
+  size_t play_channels_;
+
+  // 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_ RTC_ACCESS_ON(main_thread_checker_);
+  bool recording_ RTC_ACCESS_ON(main_thread_checker_);
+
+  // Buffer used for audio samples to be played out. Size can be changed
+  // dynamically. The 16-bit samples are interleaved, hence the size is
+  // proportional to the number of channels.
+  rtc::BufferT<int16_t> play_buffer_ RTC_ACCESS_ON(playout_thread_checker_);
+
+  // Byte buffer used for recorded audio samples. Size can be changed
+  // dynamically.
+  rtc::BufferT<int16_t> rec_buffer_ RTC_ACCESS_ON(recording_thread_checker_);
+
+  // AGC parameters.
+#if !defined(WEBRTC_WIN)
+  uint32_t current_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
+#else
+  // Windows uses a dedicated thread for volume APIs.
+  uint32_t current_mic_level_;
+#endif
+  uint32_t new_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
+
+  // Contains true of a key-press has been detected.
+  bool typing_status_ RTC_ACCESS_ON(recording_thread_checker_);
+
+  // Delay values used by the AEC.
+  int play_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
+  int rec_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
+
+  // Contains a clock-drift measurement.
+  int clock_drift_ RTC_ACCESS_ON(recording_thread_checker_);
+
+  // Counts number of times LogStats() has been called.
+  size_t num_stat_reports_ RTC_ACCESS_ON(task_queue_);
+
+  // Time stamp of last timer task (drives logging).
+  int64_t last_timer_task_time_ RTC_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.
+  int16_t rec_stat_count_ RTC_ACCESS_ON(recording_thread_checker_);
+  int16_t play_stat_count_ RTC_ACCESS_ON(playout_thread_checker_);
+
+  // Time stamps of when playout and recording starts.
+  int64_t play_start_time_ RTC_ACCESS_ON(main_thread_checker_);
+  int64_t rec_start_time_ RTC_ACCESS_ON(main_thread_checker_);
+
+  // Contains counters for playout and recording statistics.
+  Stats stats_ RTC_GUARDED_BY(lock_);
+
+  // Stores current stats at each timer task. Used to calculate differences
+  // between two successive timer events.
+  Stats last_stats_ RTC_ACCESS_ON(task_queue_);
+
+  // Set to true at construction and modified to false as soon as one audio-
+  // level estimate larger than zero is detected.
+  bool only_silence_recorded_;
+
+  // Set to true when logging of audio stats is enabled for the first time in
+  // StartPeriodicLogging() and set to false by StopPeriodicLogging().
+  // Setting this member to false prevents (possiby invalid) log messages from
+  // being printed in the LogStats() task.
+  bool log_stats_ RTC_ACCESS_ON(task_queue_);
+
+// Should *never* be defined in production builds. Only used for testing.
+// When defined, the output signal will be replaced by a sinus tone at 440Hz.
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  double phase_ RTC_ACCESS_ON(playout_thread_checker_);
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_