Index: webrtc/modules/audio_device/android/audio_manager.h |
diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h |
index 8d96d27e3913662c538f99788aba1fb026edfd48..0bc82508f44654084895ec50950bc55f8bd5b1cc 100644 |
--- a/webrtc/modules/audio_device/android/audio_manager.h |
+++ b/webrtc/modules/audio_device/android/audio_manager.h |
@@ -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 |