Merge "aaudio: remove old createThread API" into oc-dev (44d34932) · Commits · e / os / android_frameworks_av

media/libaaudio/examples/write_sine/jni/Android.mk

+2 −2

Original line number	Diff line number	Diff line
		@@ -18,9 +18,9 @@ LOCAL_C_INCLUDES := \
		$(call include-path-for, audio-utils) \
		frameworks/av/media/libaaudio/include

		LOCAL_SRC_FILES:= ../src/write_sine_threaded.cpp
		LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
		LOCAL_SHARED_LIBRARIES := libaaudio
		LOCAL_MODULE := write_sine_threaded_ndk
		LOCAL_MODULE := write_sine_callback_ndk
		include $(BUILD_EXECUTABLE)

		include $(CLEAR_VARS)

media/libaaudio/examples/write_sine/src/write_sine_threaded.cpp

deleted100644 → 0

+0 −386

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2016 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 sine waves using an AAudio background thread.

		//#include <assert.h>
		#include <atomic>
		#include <unistd.h>
		#include <stdlib.h>
		#include <stdio.h>
		#include <math.h>
		#include <time.h>
		#include <aaudio/AAudio.h>
		#include "SineGenerator.h"

		#define NUM_SECONDS 5
		#define NANOS_PER_MICROSECOND ((int64_t)1000)
		#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
		#define MILLIS_PER_SECOND 1000
		#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * MILLIS_PER_SECOND)

		#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
		//#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED

		// Prototype for a callback.
		typedef int audio_callback_proc_t(float *outputBuffer,
		int32_t numFrames,
		void *userContext);

		static void SimpleAAudioPlayerThreadProc(void arg);

		// TODO merge into common code
		static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
		struct timespec time;
		int result = clock_gettime(clockId, &time);
		if (result < 0) {
		return -errno; // TODO standardize return value
		}
		return (time.tv_sec * NANOS_PER_SECOND) + time.tv_nsec;
		}

		/**
		* Simple wrapper for AAudio that opens a default stream and then calls
		* a callback function to fill the output buffers.
		*/
		class SimpleAAudioPlayer {
		public:
		SimpleAAudioPlayer() {}
		~SimpleAAudioPlayer() {
		close();
		};

		void setSharingMode(aaudio_sharing_mode_t requestedSharingMode) {
		mRequestedSharingMode = requestedSharingMode;
		}

		/** Also known as "sample rate"
		*/
		int32_t getFramesPerSecond() {
		return mFramesPerSecond;
		}

		int32_t getSamplesPerFrame() {
		return mSamplesPerFrame;
		}

		/**
		* Open a stream
		*/
		aaudio_result_t open(audio_callback_proc_t proc, void userContext) {
		mCallbackProc = proc;
		mUserContext = userContext;
		aaudio_result_t result = AAUDIO_OK;

		// Use an AAudioStreamBuilder to contain requested parameters.
		result = AAudio_createStreamBuilder(&mBuilder);
		if (result != AAUDIO_OK) return result;

		AAudioStreamBuilder_setSharingMode(mBuilder, mRequestedSharingMode);
		AAudioStreamBuilder_setSampleRate(mBuilder, 48000);

		// Open an AAudioStream using the Builder.
		result = AAudioStreamBuilder_openStream(mBuilder, &mStream);
		if (result != AAUDIO_OK) goto error;

		printf("Requested sharing mode = %d\n", mRequestedSharingMode);
		printf("Actual sharing mode = %d\n", AAudioStream_getSharingMode(mStream));

		// Check to see what kind of stream we actually got.
		mFramesPerSecond = AAudioStream_getSampleRate(mStream);
		printf("Actual framesPerSecond = %d\n", mFramesPerSecond);

		mSamplesPerFrame = AAudioStream_getSamplesPerFrame(mStream);
		printf("Actual samplesPerFrame = %d\n", mSamplesPerFrame);

		{
		int32_t bufferCapacity = AAudioStream_getBufferCapacityInFrames(mStream);
		printf("Actual bufferCapacity = %d\n", bufferCapacity);
		}

		// This is the number of frames that are read in one chunk by a DMA controller
		// or a DSP or a mixer.
		mFramesPerBurst = AAudioStream_getFramesPerBurst(mStream);
		// Some DMA might use very short bursts. We don't need to write such small
		// buffers. But it helps to use a multiple of the burst size for predictable scheduling.
		while (mFramesPerBurst < 48) {
		mFramesPerBurst *= 2;
		}
		printf("Actual framesPerBurst = %d\n",mFramesPerBurst);

		mDataFormat = AAudioStream_getFormat(mStream);
		printf("Actual dataFormat = %d\n", mDataFormat);

		// Allocate a buffer for the audio data.
		mOutputBuffer = new float[mFramesPerBurst * mSamplesPerFrame];
		if (mOutputBuffer == nullptr) {
		fprintf(stderr, "ERROR - could not allocate data buffer\n");
		result = AAUDIO_ERROR_NO_MEMORY;
		}

		// If needed allocate a buffer for converting float to int16_t.
		if (mDataFormat == AAUDIO_FORMAT_PCM_I16) {
		printf("Allocate data conversion buffer for float=>pcm16\n");
		mConversionBuffer = new int16_t[mFramesPerBurst * mSamplesPerFrame];
		if (mConversionBuffer == nullptr) {
		fprintf(stderr, "ERROR - could not allocate conversion buffer\n");
		result = AAUDIO_ERROR_NO_MEMORY;
		}
		}
		return result;

		error:
		AAudioStreamBuilder_delete(mBuilder);
		mBuilder = nullptr;
		return result;
		}

		aaudio_result_t close() {
		if (mStream != nullptr) {
		stop();
		printf("call AAudioStream_close(%p)\n", mStream); fflush(stdout);
		AAudioStream_close(mStream);
		mStream = nullptr;
		AAudioStreamBuilder_delete(mBuilder);
		mBuilder = nullptr;
		delete mOutputBuffer;
		mOutputBuffer = nullptr;
		delete mConversionBuffer;
		mConversionBuffer = nullptr;
		}
		return AAUDIO_OK;
		}

		// Start a thread that will call the callback proc.
		aaudio_result_t start() {
		mEnabled.store(true);
		int64_t nanosPerBurst = mFramesPerBurst * NANOS_PER_SECOND
		/ mFramesPerSecond;
		return AAudioStream_createThread(mStream, nanosPerBurst,
		SimpleAAudioPlayerThreadProc,
		this);
		}

		// Tell the thread to stop.
		aaudio_result_t stop() {
		mEnabled.store(false);
		return AAudioStream_joinThread(mStream, nullptr, 2 * NANOS_PER_SECOND);
		}

		bool isEnabled() const {
		return mEnabled.load();
		}

		aaudio_result_t callbackLoop() {
		aaudio_result_t result = 0;
		int64_t framesWritten = 0;
		int32_t xRunCount = 0;
		bool started = false;
		int64_t framesInBuffer =
		AAudioStream_getFramesWritten(mStream) -
		AAudioStream_getFramesRead(mStream);
		int64_t framesAvailable =
		AAudioStream_getBufferSizeInFrames(mStream) - framesInBuffer;

		int64_t startTime = 0;
		int64_t startPosition = 0;
		int32_t loopCount = 0;

		// Give up after several burst periods have passed.
		const int burstsPerTimeout = 8;
		int64_t nanosPerTimeout = 0;
		int64_t runningNanosPerTimeout = 500 * NANOS_PER_MILLISECOND;

		while (isEnabled() && result >= 0) {
		// Call application's callback function to fill the buffer.
		if (mCallbackProc(mOutputBuffer, mFramesPerBurst, mUserContext)) {
		mEnabled.store(false);
		}

		// if needed, convert from float to int16_t PCM
		//printf("app callbackLoop writing %d frames, state = %s\n", mFramesPerBurst,
		// AAudio_convertStreamStateToText(AAudioStream_getState(mStream)));
		if (mConversionBuffer != nullptr) {
		int32_t numSamples = mFramesPerBurst * mSamplesPerFrame;
		for (int i = 0; i < numSamples; i++) {
		mConversionBuffer[i] = (int16_t)(32767.0 * mOutputBuffer[i]);
		}
		// Write the application data to stream.
		result = AAudioStream_write(mStream, mConversionBuffer,
		mFramesPerBurst, nanosPerTimeout);
		} else {
		// Write the application data to stream.
		result = AAudioStream_write(mStream, mOutputBuffer,
		mFramesPerBurst, nanosPerTimeout);
		}

		if (result < 0) {
		fprintf(stderr, "ERROR - AAudioStream_write() returned %d %s\n", result,
		AAudio_convertResultToText(result));
		break;
		} else if (started && result != mFramesPerBurst) {
		fprintf(stderr, "ERROR - AAudioStream_write() timed out! %d\n", result);
		break;
		} else {
		framesWritten += result;
		}

		if (startTime > 0 && ((loopCount & 0x01FF) == 0)) {
		double elapsedFrames = (double)(framesWritten - startPosition);
		int64_t elapsedTime = getNanoseconds() - startTime;
		double measuredRate = elapsedFrames * NANOS_PER_SECOND / elapsedTime;
		printf("app callbackLoop write() measured rate %f\n", measuredRate);
		}
		loopCount++;

		if (!started && framesWritten >= framesAvailable) {
		// Start buffer if fully primed.{
		result = AAudioStream_requestStart(mStream);
		printf("app callbackLoop requestStart returned %d\n", result);
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - AAudioStream_requestStart() returned %d %s\n", result,
		AAudio_convertResultToText(result));
		mEnabled.store(false);
		return result;
		}
		started = true;
		nanosPerTimeout = runningNanosPerTimeout;
		startPosition = framesWritten;
		startTime = getNanoseconds();
		}

		{
		int32_t tempXRunCount = AAudioStream_getXRunCount(mStream);
		if (tempXRunCount != xRunCount) {
		xRunCount = tempXRunCount;
		printf("AAudioStream_getXRunCount returns %d at frame %d\n",
		xRunCount, (int) framesWritten);
		}
		}
		}

		result = AAudioStream_requestStop(mStream);
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - AAudioStream_requestStop() returned %d %s\n", result,
		AAudio_convertResultToText(result));
		return result;
		}

		return result;
		}

		private:
		AAudioStreamBuilder *mBuilder = nullptr;
		AAudioStream *mStream = nullptr;
		float *mOutputBuffer = nullptr;
		int16_t *mConversionBuffer = nullptr;

		audio_callback_proc_t *mCallbackProc = nullptr;
		void *mUserContext = nullptr;
		aaudio_sharing_mode_t mRequestedSharingMode = SHARING_MODE;
		int32_t mSamplesPerFrame = 0;
		int32_t mFramesPerSecond = 0;
		int32_t mFramesPerBurst = 0;
		aaudio_audio_format_t mDataFormat = AAUDIO_FORMAT_PCM_I16;

		std::atomic<bool> mEnabled; // used to request that callback exit its loop
		};

		static void SimpleAAudioPlayerThreadProc(void arg) {
		SimpleAAudioPlayer player = (SimpleAAudioPlayer ) arg;
		player->callbackLoop();
		return nullptr;
		}

		// Application data that gets passed to the callback.
		typedef struct SineThreadedData_s {
		SineGenerator sineOsc1;
		SineGenerator sineOsc2;
		int32_t samplesPerFrame = 0;
		} SineThreadedData_t;

		// Callback function that fills the audio output buffer.
		int MyCallbackProc(float outputBuffer, int32_t numFrames, void userContext) {
		SineThreadedData_t data = (SineThreadedData_t ) userContext;
		// Render sine waves to left and right channels.
		data->sineOsc1.render(&outputBuffer[0], data->samplesPerFrame, numFrames);
		if (data->samplesPerFrame > 1) {
		data->sineOsc2.render(&outputBuffer[1], data->samplesPerFrame, numFrames);
		}
		return 0;
		}

		int main(int argc, char **argv)
		{
		(void)argc; // unused
		SimpleAAudioPlayer player;
		SineThreadedData_t myData;
		aaudio_result_t result;

		// 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 - Play a sine wave using an AAudio Thread\n", argv[0]);

		result = player.open(MyCallbackProc, &myData);
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - player.open() returned %d\n", result);
		goto error;
		}
		printf("player.getFramesPerSecond() = %d\n", player.getFramesPerSecond());
		printf("player.getSamplesPerFrame() = %d\n", player.getSamplesPerFrame());
		myData.sineOsc1.setup(440.0, 48000);
		myData.sineOsc1.setSweep(300.0, 600.0, 5.0);
		myData.sineOsc2.setup(660.0, 48000);
		myData.sineOsc2.setSweep(350.0, 900.0, 7.0);
		myData.samplesPerFrame = player.getSamplesPerFrame();

		result = player.start();
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - player.start() returned %d\n", result);
		goto error;
		}

		printf("Sleep for %d seconds while audio plays in a background thread.\n", NUM_SECONDS);
		for (int i = 0; i < NUM_SECONDS && player.isEnabled(); i++) {
		// FIXME sleep is not an NDK API
		// sleep(NUM_SECONDS);
		const struct timespec request = { .tv_sec = 1, .tv_nsec = 0 };
		(void) clock_nanosleep(CLOCK_MONOTONIC, 0 /flags/, &request, NULL /remain/);
		}
		printf("Woke up now!\n");

		result = player.stop();
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - player.stop() returned %d\n", result);
		goto error;
		}

		printf("Player stopped.\n");
		result = player.close();
		if (result != AAUDIO_OK) {
		fprintf(stderr, "ERROR - player.close() returned %d\n", result);
		goto error;
		}

		printf("SUCCESS\n");
		return EXIT_SUCCESS;
		error:
		player.close();
		printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
		return EXIT_FAILURE;
		}

media/libaaudio/examples/write_sine/static/Android.mk

+0 −19

Original line number	Diff line number	Diff line
		@@ -18,25 +18,6 @@ LOCAL_MODULE := write_sine
		include $(BUILD_EXECUTABLE)



		include $(CLEAR_VARS)
		LOCAL_MODULE_TAGS := tests
		LOCAL_C_INCLUDES := \
		$(call include-path-for, audio-utils) \
		frameworks/av/media/libaaudio/include

		LOCAL_SRC_FILES:= ../src/write_sine_threaded.cpp

		LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
		libbinder libcutils libutils \
		libaudioclient liblog
		LOCAL_STATIC_LIBRARIES := libaaudio

		LOCAL_MODULE := write_sine_threaded
		include $(BUILD_EXECUTABLE)



		include $(CLEAR_VARS)
		LOCAL_MODULE_TAGS := tests
		LOCAL_C_INCLUDES := \

media/libaaudio/include/aaudio/AAudio.h

+0 −51

Original line number	Diff line number	Diff line
		@@ -582,61 +582,10 @@ AAUDIO_API aaudio_result_t AAudioStream_write(AAudioStream* stream,
		int32_t numFrames,
		int64_t timeoutNanoseconds);


		// ============================================================
		// High priority audio threads
		// ============================================================

		/**
		* @deprecated Use AudioStreamBuilder_setCallback()
		*/
		typedef void (aaudio_audio_thread_proc_t)(void *);

		/**
		* @deprecated Use AudioStreamBuilder_setCallback()
		*
		* Create a thread associated with a stream. The thread has special properties for
		* low latency audio performance. This thread can be used to implement a callback API.
		*
		* Only one thread may be associated with a stream.
		*
		* If you are using multiple streams then we recommend that you only do
		* blocking reads or writes on one stream. You can do non-blocking I/O on the
		* other streams by setting the timeout to zero.
		* This thread should be created for the stream that you will block on.
		*
		* Note that this API is in flux.
		*
		* @param stream A stream created using AAudioStreamBuilder_openStream().
		* @param periodNanoseconds the estimated period at which the audio thread will need to wake up
		* @param threadProc your thread entry point
		* @param arg an argument that will be passed to your thread entry point
		* @return AAUDIO_OK or a negative error.
		*/
		AAUDIO_API aaudio_result_t AAudioStream_createThread(AAudioStream* stream,
		int64_t periodNanoseconds,
		aaudio_audio_thread_proc_t threadProc,
		void *arg);

		/**
		* @deprecated Use AudioStreamBuilder_setCallback()
		*
		* Wait until the thread exits or an error occurs.
		*
		* @param stream A stream created using AAudioStreamBuilder_openStream().
		* @param returnArg a pointer to a variable to receive the return value
		* @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion.
		* @return AAUDIO_OK or a negative error.
		*/
		AAUDIO_API aaudio_result_t AAudioStream_joinThread(AAudioStream* stream,
		void **returnArg,
		int64_t timeoutNanoseconds);

		// ============================================================
		// Stream - queries
		// ============================================================


		/**
		* This can be used to adjust the latency of the buffer by changing
		* the threshold where blocking will occur.

media/libaaudio/libaaudio.map.txt

+0 −2

Original line number	Diff line number	Diff line
		@@ -24,8 +24,6 @@ LIBAAUDIO {
		AAudioStream_waitForStateChange;
		AAudioStream_read;
		AAudioStream_write;
		AAudioStream_createThread;
		AAudioStream_joinThread;
		AAudioStream_setBufferSizeInFrames;
		AAudioStream_getBufferSizeInFrames;
		AAudioStream_getFramesPerDataCallback;