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Cleanup of iOS AudioDevice implementation

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TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo

Review URL: https://codereview.webrtc.org/1206783002 .

Cr-Commit-Position: refs/heads/master@{#9578}
This commit is contained in:
henrika
2015-07-14 17:04:08 +02:00
parent d6f1a38165
commit ba35d05a49
18 changed files with 2790 additions and 2388 deletions
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55
webrtc/modules/audio_device/android/audio_manager.h
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@ -16,6 +16,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_device/android/audio_common.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/utility/interface/helpers_android.h"
@ -23,60 +24,6 @@
namespace webrtc {
class AudioParameters {
public:
enum { kBitsPerSample = 16 };
AudioParameters()
: sample_rate_(0),
channels_(0),
frames_per_buffer_(0),
frames_per_10ms_buffer_(0),
bits_per_sample_(kBitsPerSample) {}
AudioParameters(int sample_rate, int channels, int frames_per_buffer)
: sample_rate_(sample_rate),
channels_(channels),
frames_per_buffer_(frames_per_buffer),
frames_per_10ms_buffer_(sample_rate / 100),
bits_per_sample_(kBitsPerSample) {}
void reset(int sample_rate, int channels, int frames_per_buffer) {
sample_rate_ = sample_rate;
channels_ = channels;
frames_per_buffer_ = frames_per_buffer;
frames_per_10ms_buffer_ = (sample_rate / 100);
}
int sample_rate() const { return sample_rate_; }
int channels() const { return channels_; }
int frames_per_buffer() const { return frames_per_buffer_; }
int frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
int bits_per_sample() const { return bits_per_sample_; }
bool is_valid() const {
return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
}
int GetBytesPerFrame() const { return channels_ * bits_per_sample_ / 8; }
int GetBytesPerBuffer() const {
return frames_per_buffer_ * GetBytesPerFrame();
}
int GetBytesPer10msBuffer() const {
return frames_per_10ms_buffer_ * GetBytesPerFrame();
}
float GetBufferSizeInMilliseconds() const {
if (sample_rate_ == 0)
return 0.0f;
return frames_per_buffer_ / (sample_rate_ / 1000.0f);
}
private:
int sample_rate_;
int channels_;
// Lowest possible size of native audio buffer. Measured in number of frames.
// This size is injected into the OpenSL ES output (since it does not "talk
// Java") implementation but is currently not utilized by the Java
// implementation since it aquires the same value internally.
int frames_per_buffer_;
int frames_per_10ms_buffer_;
int bits_per_sample_;
};
// Implements support for functions in the WebRTC audio stack for Android that
// relies on the AudioManager in android.media. It also populates an
// AudioParameter structure with native audio parameters detected at