| /* |
| * 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. |
| */ |
| |
| // Play silence and recover from dead servers or disconnected devices. |
| |
| #include <stdio.h> |
| #include <stdlib.h> |
| #include <unistd.h> |
| |
| #include <aaudio/AAudio.h> |
| #include <aaudio/AAudioTesting.h> |
| #include "utils/AAudioExampleUtils.h" |
| #include "../examples/utils/AAudioExampleUtils.h" |
| |
| // Arbitrary period for glitches, once per second at 48000 Hz. |
| #define FORCED_UNDERRUN_PERIOD_FRAMES 48000 |
| // How long to sleep in a callback to cause an intentional glitch. For testing. |
| #define FORCED_UNDERRUN_SLEEP_MICROS (10 * 1000) |
| |
| #define MAX_TIMESTAMPS 1000 |
| |
| #define DEFAULT_TIMEOUT_NANOS ((int64_t)1000000000) |
| |
| #define NUM_SECONDS 1 |
| #define NUM_LOOPS 4 |
| |
| typedef struct TimestampInfo { |
| int64_t framesTotal; |
| int64_t appPosition; // frames |
| int64_t appNanoseconds; |
| int64_t timestampPosition; // frames |
| int64_t timestampNanos; |
| aaudio_result_t result; |
| } TimestampInfo; |
| |
| typedef struct TimestampCallbackData_s { |
| TimestampInfo timestamps[MAX_TIMESTAMPS]; |
| int64_t framesTotal = 0; |
| int64_t nextFrameToGlitch = FORCED_UNDERRUN_PERIOD_FRAMES; |
| int32_t timestampCount = 0; // in timestamps |
| bool forceUnderruns = false; |
| } TimestampCallbackData_t; |
| |
| // Callback function that fills the audio output buffer. |
| aaudio_data_callback_result_t timestampDataCallbackProc( |
| AAudioStream *stream, |
| void *userData, |
| void *audioData __unused, |
| int32_t numFrames |
| ) { |
| |
| // should not happen but just in case... |
| if (userData == nullptr) { |
| printf("ERROR - SimplePlayerDataCallbackProc needs userData\n"); |
| return AAUDIO_CALLBACK_RESULT_STOP; |
| } |
| TimestampCallbackData_t *timestampData = (TimestampCallbackData_t *) userData; |
| |
| aaudio_direction_t direction = AAudioStream_getDirection(stream); |
| if (direction == AAUDIO_DIRECTION_INPUT) { |
| timestampData->framesTotal += numFrames; |
| } |
| |
| if (timestampData->forceUnderruns) { |
| if (timestampData->framesTotal > timestampData->nextFrameToGlitch) { |
| usleep(FORCED_UNDERRUN_SLEEP_MICROS); |
| printf("Simulate glitch at %lld\n", (long long) timestampData->framesTotal); |
| timestampData->nextFrameToGlitch += FORCED_UNDERRUN_PERIOD_FRAMES; |
| } |
| } |
| |
| if (timestampData->timestampCount < MAX_TIMESTAMPS) { |
| TimestampInfo *timestamp = ×tampData->timestamps[timestampData->timestampCount]; |
| timestamp->result = AAudioStream_getTimestamp(stream, |
| CLOCK_MONOTONIC, |
| ×tamp->timestampPosition, |
| ×tamp->timestampNanos); |
| timestamp->framesTotal = timestampData->framesTotal; |
| timestamp->appPosition = (direction == AAUDIO_DIRECTION_OUTPUT) |
| ? AAudioStream_getFramesWritten(stream) |
| : AAudioStream_getFramesRead(stream); |
| timestamp->appNanoseconds = getNanoseconds(); |
| timestampData->timestampCount++; |
| } |
| |
| if (direction == AAUDIO_DIRECTION_OUTPUT) { |
| timestampData->framesTotal += numFrames; |
| } |
| return AAUDIO_CALLBACK_RESULT_CONTINUE; |
| } |
| |
| static TimestampCallbackData_t sTimestampData; |
| |
| static aaudio_result_t testTimeStamps(aaudio_policy_t mmapPolicy, |
| aaudio_sharing_mode_t sharingMode, |
| aaudio_performance_mode_t performanceMode, |
| aaudio_direction_t direction) { |
| aaudio_result_t result = AAUDIO_OK; |
| |
| int32_t framesPerBurst = 0; |
| float *buffer = nullptr; |
| |
| int32_t actualChannelCount = 0; |
| int32_t actualSampleRate = 0; |
| int32_t originalBufferSize = 0; |
| int32_t requestedBufferSize = 0; |
| int32_t finalBufferSize = 0; |
| aaudio_format_t actualDataFormat = AAUDIO_FORMAT_PCM_FLOAT; |
| aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED; |
| aaudio_sharing_mode_t actualPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE; |
| |
| AAudioStreamBuilder *aaudioBuilder = nullptr; |
| AAudioStream *aaudioStream = nullptr; |
| |
| memset(&sTimestampData, 0, sizeof(sTimestampData)); |
| |
| printf("------------ testTimeStamps(policy = %d, sharing = %s, perf = %s, dir = %s) -----------\n", |
| mmapPolicy, |
| getSharingModeText(sharingMode), |
| getPerformanceModeText(performanceMode), |
| getDirectionText(direction)); |
| |
| AAudio_setMMapPolicy(mmapPolicy); |
| |
| // Use an AAudioStreamBuilder to contain requested parameters. |
| result = AAudio_createStreamBuilder(&aaudioBuilder); |
| if (result != AAUDIO_OK) { |
| printf("AAudio_createStreamBuilder returned %s", |
| AAudio_convertResultToText(result)); |
| goto finish; |
| } |
| |
| // Request stream properties. |
| AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_I16); |
| AAudioStreamBuilder_setSharingMode(aaudioBuilder, sharingMode); |
| AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, performanceMode); |
| AAudioStreamBuilder_setDirection(aaudioBuilder, direction); |
| AAudioStreamBuilder_setDataCallback(aaudioBuilder, timestampDataCallbackProc, &sTimestampData); |
| |
| // Create an AAudioStream using the Builder. |
| result = AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream); |
| if (result != AAUDIO_OK) { |
| printf("AAudioStreamBuilder_openStream returned %s", |
| AAudio_convertResultToText(result)); |
| goto finish; |
| } |
| |
| // Check to see what kind of stream we actually got. |
| actualSampleRate = AAudioStream_getSampleRate(aaudioStream); |
| actualChannelCount = AAudioStream_getChannelCount(aaudioStream); |
| actualDataFormat = AAudioStream_getFormat(aaudioStream); |
| |
| actualSharingMode = AAudioStream_getSharingMode(aaudioStream); |
| if (actualSharingMode != sharingMode) { |
| printf("did not get expected sharingMode, got %3d, skipping test\n", |
| actualSharingMode); |
| result = AAUDIO_OK; |
| goto finish; |
| } |
| actualPerformanceMode = AAudioStream_getPerformanceMode(aaudioStream); |
| if (actualPerformanceMode != performanceMode) { |
| printf("did not get expected performanceMode, got %3d, skipping test\n", |
| actualPerformanceMode); |
| result = AAUDIO_OK; |
| goto finish; |
| } |
| |
| printf(" chans = %3d, rate = %6d format = %d\n", |
| actualChannelCount, actualSampleRate, actualDataFormat); |
| printf(" Is MMAP used? %s\n", AAudioStream_isMMapUsed(aaudioStream) |
| ? "yes" : "no"); |
| |
| // This is the number of frames that are read in one chunk by a DMA controller |
| // or a DSP or a mixer. |
| framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream); |
| printf(" framesPerBurst = %3d\n", framesPerBurst); |
| |
| originalBufferSize = AAudioStream_getBufferSizeInFrames(aaudioStream); |
| requestedBufferSize = 4 * framesPerBurst; |
| finalBufferSize = AAudioStream_setBufferSizeInFrames(aaudioStream, requestedBufferSize); |
| |
| printf(" BufferSize: original = %4d, requested = %4d, final = %4d\n", |
| originalBufferSize, requestedBufferSize, finalBufferSize); |
| |
| { |
| int64_t position; |
| int64_t nanoseconds; |
| result = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC, &position, &nanoseconds); |
| printf("before start, AAudioStream_getTimestamp() returns %s\n", |
| AAudio_convertResultToText(result)); |
| } |
| |
| for (int runs = 0; runs < NUM_LOOPS; runs++) { |
| printf("------------------ loop #%d\n", runs); |
| |
| int64_t temp = sTimestampData.framesTotal; |
| memset(&sTimestampData, 0, sizeof(sTimestampData)); |
| sTimestampData.framesTotal = temp; |
| |
| sTimestampData.forceUnderruns = false; |
| |
| result = AAudioStream_requestStart(aaudioStream); |
| if (result != AAUDIO_OK) { |
| printf("AAudioStream_requestStart returned %s", |
| AAudio_convertResultToText(result)); |
| goto finish; |
| } |
| |
| for (int second = 0; second < NUM_SECONDS; second++) { |
| // Give AAudio callback time to run in the background. |
| sleep(1); |
| |
| // Periodically print the progress so we know it hasn't died. |
| printf("framesWritten = %d, XRuns = %d\n", |
| (int) AAudioStream_getFramesWritten(aaudioStream), |
| (int) AAudioStream_getXRunCount(aaudioStream) |
| ); |
| } |
| |
| result = AAudioStream_requestStop(aaudioStream); |
| if (result != AAUDIO_OK) { |
| printf("AAudioStream_requestStop returned %s\n", |
| AAudio_convertResultToText(result)); |
| } |
| |
| printf("timestampCount = %d\n", sTimestampData.timestampCount); |
| int printed = 0; |
| for (int i = 0; i < sTimestampData.timestampCount; i++) { |
| TimestampInfo *timestamp = &sTimestampData.timestamps[i]; |
| bool posChanged = (timestamp->timestampPosition != (timestamp - 1)->timestampPosition); |
| bool timeChanged = (timestamp->timestampNanos != (timestamp - 1)->timestampNanos); |
| if ((printed < 20) && ((i < 10) || posChanged || timeChanged)) { |
| printf(" %3d : frames %8lld, xferd %8lld", i, |
| (long long) timestamp->framesTotal, |
| (long long) timestamp->appPosition); |
| if (timestamp->result != AAUDIO_OK) { |
| printf(", result = %s\n", AAudio_convertResultToText(timestamp->result)); |
| } else { |
| bool negative = timestamp->timestampPosition < 0; |
| bool retro = (i > 0 && (timestamp->timestampPosition < |
| (timestamp - 1)->timestampPosition)); |
| const char *message = negative ? " <=NEGATIVE!" |
| : (retro ? " <= RETROGRADE!" : ""); |
| |
| double latency = calculateLatencyMillis(timestamp->timestampPosition, |
| timestamp->timestampNanos, |
| timestamp->appPosition, |
| timestamp->appNanoseconds, |
| actualSampleRate); |
| printf(", STAMP: pos = %8lld, nanos = %8lld, lat = %7.1f msec %s\n", |
| (long long) timestamp->timestampPosition, |
| (long long) timestamp->timestampNanos, |
| latency, |
| message); |
| } |
| printed++; |
| } |
| } |
| |
| // Avoid race conditions in AudioFlinger. |
| // There is normally a delay between a real user stopping and restarting a stream. |
| sleep(1); |
| } |
| |
| finish: |
| if (aaudioStream != nullptr) { |
| AAudioStream_close(aaudioStream); |
| } |
| AAudioStreamBuilder_delete(aaudioBuilder); |
| printf("result = %d = %s\n", result, AAudio_convertResultToText(result)); |
| |
| return result; |
| } |
| |
| int main(int argc, char **argv) { |
| (void) argc; |
| (void *) argv; |
| |
| aaudio_result_t result = AAUDIO_OK; |
| |
| // 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("Test Timestamps V0.1.3\n"); |
| |
| // Legacy |
| aaudio_policy_t policy = AAUDIO_POLICY_NEVER; |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_NONE, |
| AAUDIO_DIRECTION_INPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_INPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_NONE, |
| AAUDIO_DIRECTION_OUTPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_OUTPUT); |
| |
| // MMAP |
| policy = AAUDIO_POLICY_ALWAYS; |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_EXCLUSIVE, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_INPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_EXCLUSIVE, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_OUTPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_INPUT); |
| result = testTimeStamps(policy, |
| AAUDIO_SHARING_MODE_SHARED, |
| AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, |
| AAUDIO_DIRECTION_OUTPUT); |
| |
| return (result == AAUDIO_OK) ? EXIT_SUCCESS : EXIT_FAILURE; |
| } |