[AndroidWebViewShell] Replace rebaseline script with a new version using test_runner.py
[chromium-blink-merge.git] / media / filters / audio_clock.h
blob63159b59e5503cb8e176353803f12a4658b77b97
1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #ifndef MEDIA_FILTERS_AUDIO_CLOCK_H_
6 #define MEDIA_FILTERS_AUDIO_CLOCK_H_
8 #include <deque>
10 #include "base/time/time.h"
11 #include "media/base/media_export.h"
13 namespace media {
15 // Models a queue of buffered audio in a playback pipeline for use with
16 // estimating the amount of delay in wall clock time. Takes changes in playback
17 // rate into account to handle scenarios where multiple rates may be present in
18 // a playback pipeline with large delay.
21 // USAGE
23 // Prior to starting audio playback, construct an AudioClock with an initial
24 // media timestamp and a sample rate matching the sample rate the audio device
25 // was opened at.
27 // Each time the audio rendering callback is executed, call WroteAudio() once
28 // (and only once!) containing information on what was written:
29 // 1) How many frames of audio data requested
30 // 2) How many frames of audio data provided
31 // 3) The playback rate of the audio data provided
32 // 4) The current amount of delay
34 // After a call to WroteAudio(), clients can inspect the resulting media
35 // timestamp. This can be used for UI purposes, synchronizing video, etc...
38 // DETAILS
40 // Silence (whether caused by the initial audio delay or failing to write the
41 // amount of requested frames due to underflow) is also modeled and will cause
42 // the media timestamp to stop increasing until all known silence has been
43 // played. AudioClock's model is initialized with silence during the first call
44 // to WroteAudio() using the delay value.
46 // Playback rates are tracked for translating frame durations into media
47 // durations. Since silence doesn't affect media timestamps, it also isn't
48 // affected by playback rates.
49 class MEDIA_EXPORT AudioClock {
50 public:
51 AudioClock(base::TimeDelta start_timestamp, int sample_rate);
52 ~AudioClock();
54 // |frames_written| amount of audio data scaled to |playback_rate| written.
55 // |frames_requested| amount of audio data requested by hardware.
56 // |delay_frames| is the current amount of hardware delay.
57 void WroteAudio(int frames_written,
58 int frames_requested,
59 int delay_frames,
60 double playback_rate);
62 // If WroteAudio() calls are suspended (i.e. due to playback being paused) the
63 // AudioClock will not properly advance time (even though all data up until
64 // back_timestamp() will playout on the physical device).
66 // To compensate for this, when calls resume, before the next WroteAudio(),
67 // callers should call CompensateForSuspendedWrites() to advance the clock for
68 // audio which continued playing out while WroteAudio() calls were suspended.
70 // |delay_frames| must be provided to properly prime the clock to compensate
71 // for a new initial delay.
72 void CompensateForSuspendedWrites(base::TimeDelta elapsed, int delay_frames);
74 // Returns the bounds of media data currently buffered by the audio hardware,
75 // taking silence and changes in playback rate into account. Buffered audio
76 // structure and timestamps are updated with every call to WroteAudio().
78 // start_timestamp = 1000 ms sample_rate = 40 Hz
79 // +-----------------------+-----------------------+-----------------------+
80 // | 10 frames silence | 20 frames @ 1.0x | 20 frames @ 0.5x |
81 // | = 250 ms (wall) | = 500 ms (wall) | = 500 ms (wall) |
82 // | = 0 ms (media) | = 500 ms (media) | = 250 ms (media) |
83 // +-----------------------+-----------------------+-----------------------+
84 // ^ ^
85 // front_timestamp() is equal to back_timestamp() is equal to
86 // |start_timestamp| since no amount of media frames tracked
87 // media data has been played yet. by AudioClock, which would be
88 // 1000 + 500 + 250 = 1750 ms.
89 base::TimeDelta front_timestamp() const { return front_timestamp_; }
90 base::TimeDelta back_timestamp() const { return back_timestamp_; }
92 // Returns the amount of wall time until |timestamp| will be played by the
93 // audio hardware.
95 // |timestamp| must be within front_timestamp() and back_timestamp().
96 base::TimeDelta TimeUntilPlayback(base::TimeDelta timestamp) const;
98 void ContiguousAudioDataBufferedForTesting(
99 base::TimeDelta* total,
100 base::TimeDelta* same_rate_total) const;
102 private:
103 // Even with a ridiculously high sample rate of 256kHz, using 64 bits will
104 // permit tracking up to 416999965 days worth of time (that's 1141 millenia).
106 // 32 bits on the other hand would top out at measly 2 hours and 20 minutes.
107 struct AudioData {
108 AudioData(int64_t frames, double playback_rate);
110 int64_t frames;
111 double playback_rate;
114 // Helpers for operating on |buffered_|.
115 void PushBufferedAudioData(int64_t frames, double playback_rate);
116 void PopBufferedAudioData(int64_t frames);
117 base::TimeDelta ComputeBufferedMediaDuration() const;
119 const base::TimeDelta start_timestamp_;
120 const double microseconds_per_frame_;
122 std::deque<AudioData> buffered_;
123 int64_t total_buffered_frames_;
125 base::TimeDelta front_timestamp_;
126 base::TimeDelta back_timestamp_;
128 DISALLOW_COPY_AND_ASSIGN(AudioClock);
131 } // namespace media
133 #endif // MEDIA_FILTERS_AUDIO_CLOCK_H_