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| /* Copyright 2018 The TensorFlow Authors. All Rights Reserved. | ||
| Licensed under the Apache License, Version 2.0 (the "License"); | ||
| you may not use this file except in compliance with the License. | ||
| You may obtain a copy of the License at | ||
| http://www.apache.org/licenses/LICENSE-2.0 | ||
| Unless required by applicable law or agreed to in writing, software | ||
| distributed under the License is distributed on an "AS IS" BASIS, | ||
| WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| See the License for the specific language governing permissions and | ||
| limitations under the License. | ||
| ==============================================================================*/ | ||
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| #include <stdio.h> | ||
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| #include "audio_provider.h" | ||
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| extern "C" { | ||
| #include "pico/pdm_microphone.h" | ||
| } | ||
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| #include "micro_features/micro_model_settings.h" | ||
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| #define DEFAULT_PDM_BUFFER_SIZE 256 | ||
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| namespace { | ||
| bool g_is_audio_initialized = false; | ||
| // An internal buffer able to fit 16x our sample size | ||
| constexpr int kAudioCaptureBufferSize = DEFAULT_PDM_BUFFER_SIZE * 16; | ||
| int16_t g_audio_capture_buffer[kAudioCaptureBufferSize]; | ||
| // A buffer that holds our output | ||
| int16_t g_audio_output_buffer[kMaxAudioSampleSize]; | ||
| // Mark as volatile so we can check in a while loop to see if | ||
| // any samples have arrived yet. | ||
| volatile int32_t g_latest_audio_timestamp = 0; | ||
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| struct pdm_microphone_config pdm_config = { | ||
| .gpio_data = 2, | ||
| .gpio_clk = 3, | ||
| .pio = pio0, | ||
| .pio_sm = 0, | ||
| .sample_rate = 16000, | ||
| .sample_buffer_size = 256, | ||
| }; | ||
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| // struct analog_microphone_config config = { | ||
| // .gpio = 26, | ||
| // .bias_voltage = 1.25, | ||
| // .sample_rate = 16000, | ||
| // .sample_buffer_size = 256, | ||
| // }; | ||
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| } // namespace | ||
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| #include "pico/stdlib.h" | ||
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| void CaptureSamples() { | ||
| // This is how many bytes of new data we have each time this is called | ||
| const int number_of_samples = DEFAULT_PDM_BUFFER_SIZE; | ||
| // Calculate what timestamp the last audio sample represents | ||
| const int32_t time_in_ms = | ||
| g_latest_audio_timestamp + | ||
| (number_of_samples / (kAudioSampleFrequency / 1000)); | ||
| // Determine the index, in the history of all samples, of the last sample | ||
| const int32_t start_sample_offset = | ||
| g_latest_audio_timestamp * (kAudioSampleFrequency / 1000); | ||
| // Determine the index of this sample in our ring buffer | ||
| const int capture_index = start_sample_offset % kAudioCaptureBufferSize; | ||
| // Read the data to the correct place in our buffer | ||
| pdm_microphone_read(g_audio_capture_buffer + capture_index, DEFAULT_PDM_BUFFER_SIZE); | ||
| // analog_microphone_read(g_audio_capture_buffer + capture_index, DEFAULT_PDM_BUFFER_SIZE); | ||
| // This is how we let the outside world know that new audio data has arrived. | ||
| g_latest_audio_timestamp = time_in_ms; | ||
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| // uint32_t now = to_ms_since_boot(get_absolute_time()); | ||
| // printf("now = %u\n", now); | ||
| } | ||
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| TfLiteStatus InitAudioRecording(tflite::ErrorReporter* error_reporter) { | ||
| pdm_microphone_init(&pdm_config); | ||
| // pdm_microphone_set_filter_gain(20); | ||
| // pdm_microphone_set_filter_max_volume(128); | ||
| pdm_microphone_set_samples_ready_handler(CaptureSamples); | ||
| pdm_microphone_start(); | ||
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| // analog_microphone_init(&config); | ||
| // analog_microphone_set_samples_ready_handler(CaptureSamples); | ||
| // analog_microphone_start(); | ||
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| // Block until we have our first audio sample | ||
| while (!g_latest_audio_timestamp) { | ||
| } | ||
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| return kTfLiteOk; | ||
| } | ||
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| TfLiteStatus GetAudioSamples(tflite::ErrorReporter* error_reporter, | ||
| int start_ms, int duration_ms, | ||
| int* audio_samples_size, int16_t** audio_samples) { | ||
| // Set everything up to start receiving audio | ||
| if (!g_is_audio_initialized) { | ||
| TfLiteStatus init_status = InitAudioRecording(error_reporter); | ||
| if (init_status != kTfLiteOk) { | ||
| return init_status; | ||
| } | ||
| g_is_audio_initialized = true; | ||
| } | ||
| // This next part should only be called when the main thread notices that the | ||
| // latest audio sample data timestamp has changed, so that there's new data | ||
| // in the capture ring buffer. The ring buffer will eventually wrap around and | ||
| // overwrite the data, but the assumption is that the main thread is checking | ||
| // often enough and the buffer is large enough that this call will be made | ||
| // before that happens. | ||
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| // Determine the index, in the history of all samples, of the first | ||
| // sample we want | ||
| const int start_offset = start_ms * (kAudioSampleFrequency / 1000); | ||
| // Determine how many samples we want in total | ||
| const int duration_sample_count = | ||
| duration_ms * (kAudioSampleFrequency / 1000); | ||
| for (int i = 0; i < duration_sample_count; ++i) { | ||
| // For each sample, transform its index in the history of all samples into | ||
| // its index in g_audio_capture_buffer | ||
| const int capture_index = (start_offset + i) % kAudioCaptureBufferSize; | ||
| // Write the sample to the output buffer | ||
| g_audio_output_buffer[i] = g_audio_capture_buffer[capture_index]; | ||
| } | ||
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| // Set pointers to provide access to the audio | ||
| *audio_samples_size = kMaxAudioSampleSize; | ||
| *audio_samples = g_audio_output_buffer; | ||
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| return kTfLiteOk; | ||
| } | ||
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| int32_t LatestAudioTimestamp() { return g_latest_audio_timestamp; } |