b26198972c
BUG=4573,2982,2175,3590 TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo Summary: - Removes dependency of the 'enable_android_opensl' compiler flag. Instead, OpenSL ES is always supported, and will enabled for devices that supports low-latency output. - WebRTC no longer supports OpenSL ES for the input/recording side. - Removes old code and demos using OpenSL ES for audio input. - Improves accuracy of total delay estimates (better AEC performance). - Reduces roundtrip audio latency; especially when OpenSL can be used. Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6. Android One device. R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51759004 Cr-Commit-Position: refs/heads/master@{#9208}
70 lines
2.6 KiB
C++
70 lines
2.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioDeviceBuffer;
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// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
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// corresponding to 10ms of data. It then allows for this data to be pulled in
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// a finer or coarser granularity. I.e. interacting with this class instead of
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// directly with the AudioDeviceBuffer one can ask for any number of audio data
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// samples.
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class FineAudioBuffer {
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public:
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// |device_buffer| is a buffer that provides 10ms of audio data.
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// |desired_frame_size_bytes| is the number of bytes of audio data
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// (not samples) |GetBufferData| should return on success.
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// |sample_rate| is the sample rate of the audio data. This is needed because
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// |device_buffer| delivers 10ms of data. Given the sample rate the number
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// of samples can be calculated.
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FineAudioBuffer(AudioDeviceBuffer* device_buffer,
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int desired_frame_size_bytes,
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int sample_rate);
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~FineAudioBuffer();
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// Returns the required size of |buffer| when calling GetBufferData. If the
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// buffer is smaller memory trampling will happen.
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// |desired_frame_size_bytes| and |samples_rate| are as described in the
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// constructor.
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int RequiredBufferSizeBytes();
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// |buffer| must be of equal or greater size than what is returned by
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// RequiredBufferSize. This is to avoid unnecessary memcpy.
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void GetBufferData(int8_t* buffer);
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private:
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// Device buffer that provides 10ms chunks of data.
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AudioDeviceBuffer* device_buffer_;
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// Number of bytes delivered per GetBufferData
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int desired_frame_size_bytes_;
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int sample_rate_;
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int samples_per_10_ms_;
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// Convenience parameter to avoid converting from samples
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int bytes_per_10_ms_;
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// Storage for samples that are not yet asked for.
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rtc::scoped_ptr<int8_t[]> cache_buffer_;
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// Location of first unread sample.
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int cached_buffer_start_;
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// Number of bytes stored in cache.
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int cached_bytes_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
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