Index: webrtc/common_audio/resampler/include/resampler.h |
diff --git a/webrtc/common_audio/resampler/include/resampler.h b/webrtc/common_audio/resampler/include/resampler.h |
index 259349b670b2d7b01125e8bbabe9a65f5d9bf7d0..e26ac904c0b9186e5e919690a30e1c5d7c86bb37 100644 |
--- a/webrtc/common_audio/resampler/include/resampler.h |
+++ b/webrtc/common_audio/resampler/include/resampler.h |
@@ -13,8 +13,8 @@ |
* A wrapper for resampling a numerous amount of sampling combinations. |
*/ |
-#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_ |
-#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_ |
+#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_ |
+#define WEBRTC_RESAMPLER_RESAMPLER_H_ |
#include <stddef.h> |
@@ -23,70 +23,73 @@ |
namespace webrtc { |
// All methods return 0 on success and -1 on failure. |
-class Resampler { |
- public: |
- Resampler(); |
- Resampler(int inFreq, int outFreq, size_t num_channels); |
- ~Resampler(); |
+class Resampler |
+{ |
- // Reset all states |
- int Reset(int inFreq, int outFreq, size_t num_channels); |
+public: |
+ Resampler(); |
+ Resampler(int inFreq, int outFreq, size_t num_channels); |
+ ~Resampler(); |
- // Reset all states if any parameter has changed |
- int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels); |
+ // Reset all states |
+ int Reset(int inFreq, int outFreq, size_t num_channels); |
- // Resample samplesIn to samplesOut. |
- int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut, |
- size_t maxLen, size_t& outLen); // NOLINT: to avoid changing APIs |
+ // Reset all states if any parameter has changed |
+ int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels); |
- private: |
- enum ResamplerMode { |
- kResamplerMode1To1, |
- kResamplerMode1To2, |
- kResamplerMode1To3, |
- kResamplerMode1To4, |
- kResamplerMode1To6, |
- kResamplerMode1To12, |
- kResamplerMode2To3, |
- kResamplerMode2To11, |
- kResamplerMode4To11, |
- kResamplerMode8To11, |
- kResamplerMode11To16, |
- kResamplerMode11To32, |
- kResamplerMode2To1, |
- kResamplerMode3To1, |
- kResamplerMode4To1, |
- kResamplerMode6To1, |
- kResamplerMode12To1, |
- kResamplerMode3To2, |
- kResamplerMode11To2, |
- kResamplerMode11To4, |
- kResamplerMode11To8 |
- }; |
+ // Resample samplesIn to samplesOut. |
+ int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut, |
+ size_t maxLen, size_t &outLen); |
- // Generic pointers since we don't know what states we'll need |
- void* state1_; |
- void* state2_; |
- void* state3_; |
+private: |
+ enum ResamplerMode |
+ { |
+ kResamplerMode1To1, |
+ kResamplerMode1To2, |
+ kResamplerMode1To3, |
+ kResamplerMode1To4, |
+ kResamplerMode1To6, |
+ kResamplerMode1To12, |
+ kResamplerMode2To3, |
+ kResamplerMode2To11, |
+ kResamplerMode4To11, |
+ kResamplerMode8To11, |
+ kResamplerMode11To16, |
+ kResamplerMode11To32, |
+ kResamplerMode2To1, |
+ kResamplerMode3To1, |
+ kResamplerMode4To1, |
+ kResamplerMode6To1, |
+ kResamplerMode12To1, |
+ kResamplerMode3To2, |
+ kResamplerMode11To2, |
+ kResamplerMode11To4, |
+ kResamplerMode11To8 |
+ }; |
- // Storage if needed |
- int16_t* in_buffer_; |
- int16_t* out_buffer_; |
- size_t in_buffer_size_; |
- size_t out_buffer_size_; |
- size_t in_buffer_size_max_; |
- size_t out_buffer_size_max_; |
+ // Generic pointers since we don't know what states we'll need |
+ void* state1_; |
+ void* state2_; |
+ void* state3_; |
- int my_in_frequency_khz_; |
- int my_out_frequency_khz_; |
- ResamplerMode my_mode_; |
- size_t num_channels_; |
+ // Storage if needed |
+ int16_t* in_buffer_; |
+ int16_t* out_buffer_; |
+ size_t in_buffer_size_; |
+ size_t out_buffer_size_; |
+ size_t in_buffer_size_max_; |
+ size_t out_buffer_size_max_; |
- // Extra instance for stereo |
- Resampler* slave_left_; |
- Resampler* slave_right_; |
+ int my_in_frequency_khz_; |
+ int my_out_frequency_khz_; |
+ ResamplerMode my_mode_; |
+ size_t num_channels_; |
+ |
+ // Extra instance for stereo |
+ Resampler* slave_left_; |
+ Resampler* slave_right_; |
}; |
} // namespace webrtc |
-#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_ |
+#endif // WEBRTC_RESAMPLER_RESAMPLER_H_ |