media/libaaudio/examples/input_monitor/src/input_monitor.cpp

/*
 * Copyright (C) 2017 The Android Open Source Project
 *
 * 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.
 */

// Record input using AAudio and display the peak amplitudes.

#include <new>
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <aaudio/AAudio.h>
#include <aaudio/AAudioTesting.h>
#include "AAudioExampleUtils.h"
#include "AAudioSimpleRecorder.h"

// TODO support FLOAT
#define REQUIRED_FORMAT    AAUDIO_FORMAT_PCM_I16
#define MIN_FRAMES_TO_READ 48  /* arbitrary, 1 msec at 48000 Hz */

static const int FRAMES_PER_LINE = 20000;

int main(int argc, const char **argv)
{
    AAudioArgsParser   argParser;
    aaudio_result_t result;
    AAudioSimpleRecorder recorder;
    int actualSamplesPerFrame;
    int actualSampleRate;
    aaudio_format_t       actualDataFormat;

    AAudioStream *aaudioStream = nullptr;
    aaudio_stream_state_t state;
    int32_t framesPerBurst = 0;
    int32_t framesPerRead = 0;
    int32_t framesToRecord = 0;
    int32_t framesLeft = 0;
    int32_t nextFrameCount = 0;
    int32_t frameCount = 0;
    int32_t xRunCount = 0;
    int64_t previousFramePosition = -1;
    int16_t *data = nullptr;
    float peakLevel = 0.0;
    int32_t deviceId;

    // Make printf print immediately so that debug info is not stuck
    // in a buffer if we hang or crash.
    setvbuf(stdout, nullptr, _IONBF, (size_t) 0);

    printf("%s - Monitor input level using AAudio read, V0.1.2\n", argv[0]);

    argParser.setFormat(REQUIRED_FORMAT);
    if (argParser.parseArgs(argc, argv)) {
        return EXIT_FAILURE;
    }

    result = recorder.open(argParser);
    if (result != AAUDIO_OK) {
        fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
        goto finish;
    }
    aaudioStream = recorder.getStream();
    argParser.compareWithStream(aaudioStream);

    deviceId = AAudioStream_getDeviceId(aaudioStream);
    printf("deviceId = %d\n", deviceId);

    actualSamplesPerFrame = AAudioStream_getChannelCount(aaudioStream);
    printf("SamplesPerFrame = %d\n", actualSamplesPerFrame);
    actualSampleRate = AAudioStream_getSampleRate(aaudioStream);
    printf("SamplesPerFrame = %d\n", actualSampleRate);

    // This is the number of frames that are written in one chunk by a DMA controller
    // or a DSP.
    framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream);
    printf("DataFormat: framesPerBurst = %d\n",framesPerBurst);

    // Some DMA might use very short bursts of 16 frames. We don't need to read such small
    // buffers. But it helps to use a multiple of the burst size for predictable scheduling.
    framesPerRead = framesPerBurst;
    while (framesPerRead < MIN_FRAMES_TO_READ) {
        framesPerRead *= 2;
    }
    printf("DataFormat: framesPerRead  = %d\n",framesPerRead);

    actualDataFormat = AAudioStream_getFormat(aaudioStream);
    printf("DataFormat: requested      = %d, actual = %d\n",
           REQUIRED_FORMAT, actualDataFormat);
    // TODO handle other data formats
    assert(actualDataFormat == REQUIRED_FORMAT);

    // Allocate a buffer for the PCM_16 audio data.
    data = new(std::nothrow) int16_t[framesPerRead * actualSamplesPerFrame];
    if (data == nullptr) {
        fprintf(stderr, "ERROR - could not allocate data buffer\n");
        result = AAUDIO_ERROR_NO_MEMORY;
        goto finish;
    }

    // Start the stream.
    result = recorder.start();
    if (result != AAUDIO_OK) {
        fprintf(stderr, "ERROR -  recorder.start() returned %d\n", result);
        goto finish;
    }

    state = AAudioStream_getState(aaudioStream);
    printf("after start, state = %s\n", AAudio_convertStreamStateToText(state));

    // Record for a while.
    framesToRecord = actualSampleRate * argParser.getDurationSeconds();
    framesLeft = framesToRecord;
    while (framesLeft > 0) {
        // Read audio data from the stream.
        const int64_t timeoutNanos = 1000 * NANOS_PER_MILLISECOND;
        int minFrames = (framesToRecord < framesPerRead) ? framesToRecord : framesPerRead;
        int actual = AAudioStream_read(aaudioStream, data, minFrames, timeoutNanos);
        if (actual < 0) {
            fprintf(stderr, "ERROR - AAudioStream_read() returned %d\n", actual);
            result = actual;
            goto finish;
        } else if (actual == 0) {
            fprintf(stderr, "WARNING - AAudioStream_read() returned %d\n", actual);
            goto finish;
        }
        framesLeft -= actual;
        frameCount += actual;

        // Peak finder.
        for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
            float sample = data[frameIndex * actualSamplesPerFrame] * (1.0/32768);
            if (sample > peakLevel) {
                peakLevel = sample;
            }
        }

        // Display level as stars, eg. "******".
        if (frameCount > nextFrameCount) {
            displayPeakLevel(peakLevel);
            peakLevel = 0.0;
            nextFrameCount += FRAMES_PER_LINE;
        }

        // Print timestamps.
        int64_t framePosition = 0;
        int64_t frameTime = 0;
        aaudio_result_t timeResult;
        timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
                                               &framePosition, &frameTime);

        if (timeResult == AAUDIO_OK) {
            if (framePosition > (previousFramePosition + FRAMES_PER_LINE)) {
                int64_t realTime = getNanoseconds();
                int64_t framesRead = AAudioStream_getFramesRead(aaudioStream);

                double latencyMillis = calculateLatencyMillis(framesRead, realTime,
                                                              framePosition, frameTime,
                                                              actualSampleRate);

                printf("--- timestamp: result = %4d, position = %lld, at %lld nanos"
                               ", latency = %7.2f msec\n",
                       timeResult,
                       (long long) framePosition,
                       (long long) frameTime,
                       latencyMillis);
                previousFramePosition = framePosition;
            }
        }
    }

    xRunCount = AAudioStream_getXRunCount(aaudioStream);
    printf("AAudioStream_getXRunCount %d\n", xRunCount);

    result = recorder.stop();
    if (result != AAUDIO_OK) {
        goto finish;
    }

    argParser.compareWithStream(aaudioStream);

finish:
    recorder.close();
    delete[] data;
    printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
    return (result != AAUDIO_OK) ? EXIT_FAILURE : EXIT_SUCCESS;
}