#include <simulator_buffers.h>
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| SimulatorBuffers (int render_input_sample_rate_hz, int capture_input_sample_rate_hz, int render_output_sample_rate_hz, int capture_output_sample_rate_hz, size_t num_render_input_channels, size_t num_capture_input_channels, size_t num_render_output_channels, size_t num_capture_output_channels) |
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| ~SimulatorBuffers () |
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void | CreateConfigAndBuffer (int sample_rate_hz, size_t num_channels, Random *rand_gen, std::unique_ptr< AudioBuffer > *buffer, StreamConfig *config, std::vector< float *> *buffer_data, std::vector< float > *buffer_data_samples) |
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void | UpdateInputBuffers () |
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| SimulatorBuffers (int render_input_sample_rate_hz, int capture_input_sample_rate_hz, int render_output_sample_rate_hz, int capture_output_sample_rate_hz, size_t num_render_input_channels, size_t num_capture_input_channels, size_t num_render_output_channels, size_t num_capture_output_channels) |
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| ~SimulatorBuffers () |
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void | CreateConfigAndBuffer (int sample_rate_hz, size_t num_channels, Random *rand_gen, std::unique_ptr< AudioBuffer > *buffer, StreamConfig *config, std::vector< float *> *buffer_data, std::vector< float > *buffer_data_samples) |
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void | UpdateInputBuffers () |
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◆ SimulatorBuffers() [1/2]
webrtc::test::SimulatorBuffers::SimulatorBuffers |
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int |
render_input_sample_rate_hz, |
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int |
capture_input_sample_rate_hz, |
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int |
render_output_sample_rate_hz, |
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int |
capture_output_sample_rate_hz, |
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size_t |
num_render_input_channels, |
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size_t |
num_capture_input_channels, |
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size_t |
num_render_output_channels, |
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size_t |
num_capture_output_channels |
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) |
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◆ ~SimulatorBuffers() [1/2]
webrtc::test::SimulatorBuffers::~SimulatorBuffers |
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default |
◆ SimulatorBuffers() [2/2]
webrtc::test::SimulatorBuffers::SimulatorBuffers |
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int |
render_input_sample_rate_hz, |
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int |
capture_input_sample_rate_hz, |
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int |
render_output_sample_rate_hz, |
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int |
capture_output_sample_rate_hz, |
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size_t |
num_render_input_channels, |
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size_t |
num_capture_input_channels, |
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size_t |
num_render_output_channels, |
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size_t |
num_capture_output_channels |
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) |
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◆ ~SimulatorBuffers() [2/2]
webrtc::test::SimulatorBuffers::~SimulatorBuffers |
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◆ CreateConfigAndBuffer() [1/2]
void webrtc::test::SimulatorBuffers::CreateConfigAndBuffer |
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int |
sample_rate_hz, |
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size_t |
num_channels, |
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Random * |
rand_gen, |
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std::unique_ptr< AudioBuffer > * |
buffer, |
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StreamConfig * |
config, |
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std::vector< float *> * |
buffer_data, |
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std::vector< float > * |
buffer_data_samples |
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◆ CreateConfigAndBuffer() [2/2]
void webrtc::test::SimulatorBuffers::CreateConfigAndBuffer |
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int |
sample_rate_hz, |
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size_t |
num_channels, |
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Random * |
rand_gen, |
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std::unique_ptr< AudioBuffer > * |
buffer, |
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StreamConfig * |
config, |
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std::vector< float *> * |
buffer_data, |
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std::vector< float > * |
buffer_data_samples |
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) |
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◆ UpdateInputBuffers() [1/2]
void webrtc::test::SimulatorBuffers::UpdateInputBuffers |
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◆ UpdateInputBuffers() [2/2]
void webrtc::test::SimulatorBuffers::UpdateInputBuffers |
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◆ capture_input
std::vector< float * > webrtc::test::SimulatorBuffers::capture_input |
◆ capture_input_buffer
std::unique_ptr< AudioBuffer > webrtc::test::SimulatorBuffers::capture_input_buffer |
◆ capture_input_config
StreamConfig webrtc::test::SimulatorBuffers::capture_input_config |
◆ capture_input_samples
std::vector< float > webrtc::test::SimulatorBuffers::capture_input_samples |
◆ capture_output
std::vector< float * > webrtc::test::SimulatorBuffers::capture_output |
◆ capture_output_buffer
std::unique_ptr< AudioBuffer > webrtc::test::SimulatorBuffers::capture_output_buffer |
◆ capture_output_config
StreamConfig webrtc::test::SimulatorBuffers::capture_output_config |
◆ capture_output_samples
std::vector< float > webrtc::test::SimulatorBuffers::capture_output_samples |
◆ render_input
std::vector< float * > webrtc::test::SimulatorBuffers::render_input |
◆ render_input_buffer
std::unique_ptr< AudioBuffer > webrtc::test::SimulatorBuffers::render_input_buffer |
◆ render_input_config
StreamConfig webrtc::test::SimulatorBuffers::render_input_config |
◆ render_input_samples
std::vector< float > webrtc::test::SimulatorBuffers::render_input_samples |
◆ render_output
std::vector< float * > webrtc::test::SimulatorBuffers::render_output |
◆ render_output_buffer
std::unique_ptr< AudioBuffer > webrtc::test::SimulatorBuffers::render_output_buffer |
◆ render_output_config
StreamConfig webrtc::test::SimulatorBuffers::render_output_config |
◆ render_output_samples
std::vector< float > webrtc::test::SimulatorBuffers::render_output_samples |
The documentation for this struct was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_processing/test/simulator_buffers.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/simulator_buffers.cc