| // Copyright 2016 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #ifndef MEDIA_BASE_AUDIO_LATENCY_H_ |
| #define MEDIA_BASE_AUDIO_LATENCY_H_ |
| |
| #include "media/base/media_shmem_export.h" |
| |
| namespace base { |
| class TimeDelta; |
| } |
| |
| namespace media { |
| |
| class MEDIA_SHMEM_EXPORT AudioLatency { |
| public: |
| // Categories of expected latencies for input/output audio. Do not change |
| // existing values, they are used for UMA histogram reporting. |
| enum LatencyType { |
| // Specific latency in milliseconds. |
| LATENCY_EXACT_MS = 0, |
| // Lowest possible latency which does not cause glitches. |
| LATENCY_INTERACTIVE = 1, |
| // Latency optimized for real time communication. |
| LATENCY_RTC = 2, |
| // Latency optimized for continuous playback and power saving. |
| LATENCY_PLAYBACK = 3, |
| // For validation only. |
| LATENCY_LAST = LATENCY_PLAYBACK, |
| LATENCY_COUNT = LATENCY_LAST + 1 |
| }; |
| |
| // Indicates if the OS does not require resampling for playback. |
| static bool IsResamplingPassthroughSupported(LatencyType type); |
| |
| // |preferred_buffer_size| should be set to 0 if a client has no preference. |
| static int GetHighLatencyBufferSize(int sample_rate, |
| int preferred_buffer_size); |
| |
| // |hardware_buffer_size| should be set to 0 if unknown/invalid/not preferred. |
| static int GetRtcBufferSize(int sample_rate, int hardware_buffer_size); |
| |
| static int GetInteractiveBufferSize(int hardware_buffer_size); |
| |
| // Return the closest buffer size for this platform that will result in a |
| // latency not less than |duration| for the given |sample_rate|. |
| // |
| // Requirements: |
| // - |hardware_buffer_size| > 0 and |max_allowed_buffer_size| > 0. |
| // - If |min_hardware_buffer_size| and |max_hardware_buffer_size| are > 0 then |
| // the following must be true: |min_hardware_buffer_size| <= |
| // |hardware_buffer_size| <= |max_hardware_buffer_size| <= |
| // |max_allowed_buffer_size|. |
| // |
| // The returned buffer size is guaranteed to be between |
| // |min_hardware_buffer_size| and |max_allowed_buffer_size|. |
| // |max_hardware_buffer_size| is used to help determine a buffer size that |
| // won't cause the caller and the hardware to run at unsynchronized buffer |
| // sizes (e.g. hardware running at 4096 and caller running at 4224). |
| // |hardware_buffer_size| is the platform's preferred buffer size. |
| // |
| // It is valid for both the min and max to be zero in which case only |
| // |hardware_buffer_size| and multiples of it will be used. |
| static int GetExactBufferSize(base::TimeDelta duration, |
| int sample_rate, |
| int hardware_buffer_size, |
| int min_hardware_buffer_size, |
| int max_hardware_buffer_size, |
| int max_allowed_buffer_size); |
| }; |
| |
| } // namespace media |
| |
| #endif // MEDIA_BASE_AUDIO_LATENCY_H_ |