Merge "Codec2: add C2PlatformComponentStore"
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 2450920..fc5830a 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -25,6 +25,7 @@
#include "aaudio/AAudio.h"
#include <aaudio/AAudioTesting.h>
+#include <math.h>
#include "utility/AAudioUtilities.h"
@@ -50,44 +51,10 @@
return size;
}
-
// TODO expose and call clamp16_from_float function in primitives.h
static inline int16_t clamp16_from_float(float f) {
- /* Offset is used to expand the valid range of [-1.0, 1.0) into the 16 lsbs of the
- * floating point significand. The normal shift is 3<<22, but the -15 offset
- * is used to multiply by 32768.
- */
- static const float offset = (float)(3 << (22 - 15));
- /* zero = (0x10f << 22) = 0x43c00000 (not directly used) */
- static const int32_t limneg = (0x10f << 22) /*zero*/ - 32768; /* 0x43bf8000 */
- static const int32_t limpos = (0x10f << 22) /*zero*/ + 32767; /* 0x43c07fff */
-
- union {
- float f;
- int32_t i;
- } u;
-
- u.f = f + offset; /* recenter valid range */
- /* Now the valid range is represented as integers between [limneg, limpos].
- * Clamp using the fact that float representation (as an integer) is an ordered set.
- */
- if (u.i < limneg)
- u.i = -32768;
- else if (u.i > limpos)
- u.i = 32767;
- return u.i; /* Return lower 16 bits, the part of interest in the significand. */
-}
-
-// Same but without clipping.
-// Convert -1.0f to +1.0f to -32768 to +32767
-static inline int16_t floatToInt16(float f) {
- static const float offset = (float)(3 << (22 - 15));
- union {
- float f;
- int32_t i;
- } u;
- u.f = f + offset; /* recenter valid range */
- return u.i; /* Return lower 16 bits, the part of interest in the significand. */
+ static const float scale = 1 << 15;
+ return (int16_t) roundf(fmaxf(fminf(f * scale, scale - 1.f), -scale));
}
static float clipAndClampFloatToPcm16(float sample, float scaler) {
@@ -188,13 +155,14 @@
int32_t samplesPerFrame,
float amplitude1,
float amplitude2) {
- float scaler = amplitude1 / SHORT_SCALE;
- float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+ // Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
+ float scaler = amplitude1;
+ float delta = (amplitude2 - amplitude1) / numFrames;
for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
// No need to clip because int16_t range is inherently limited.
float sample = *source++ * scaler;
- *destination++ = floatToInt16(sample);
+ *destination++ = (int16_t) roundf(sample);
}
scaler += delta;
}
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 884a2b3..9f80695 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -111,3 +111,16 @@
"libutils",
],
}
+
+cc_test {
+ name: "test_aaudio_monkey",
+ defaults: ["libaaudio_tests_defaults"],
+ srcs: ["test_aaudio_monkey.cpp"],
+ header_libs: ["libaaudio_example_utils"],
+ shared_libs: [
+ "libaaudio",
+ "libbinder",
+ "libcutils",
+ "libutils",
+ ],
+}
diff --git a/media/libaaudio/tests/test_aaudio_monkey.cpp b/media/libaaudio/tests/test_aaudio_monkey.cpp
new file mode 100644
index 0000000..be54835
--- /dev/null
+++ b/media/libaaudio/tests/test_aaudio_monkey.cpp
@@ -0,0 +1,307 @@
+/*
+ * 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.
+ */
+
+// Try to trigger bugs by playing randomly on multiple streams.
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <vector>
+
+#include <aaudio/AAudio.h>
+#include "AAudioArgsParser.h"
+#include "AAudioExampleUtils.h"
+#include "AAudioSimplePlayer.h"
+#include "SineGenerator.h"
+
+#define DEFAULT_TIMEOUT_NANOS (1 * NANOS_PER_SECOND)
+
+#define NUM_LOOPS 1000
+#define MAX_MICROS_DELAY (2 * 1000 * 1000)
+
+// TODO Consider adding an input stream.
+#define PROB_START (0.20)
+#define PROB_PAUSE (PROB_START + 0.10)
+#define PROB_FLUSH (PROB_PAUSE + 0.10)
+#define PROB_STOP (PROB_FLUSH + 0.10)
+#define PROB_CLOSE (PROB_STOP + 0.10)
+static_assert(PROB_CLOSE < 0.9, "Probability sum too high.");
+
+aaudio_data_callback_result_t AAudioMonkeyDataCallback(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames);
+
+void AAudioMonkeyErrorCallbackProc(
+ AAudioStream *stream __unused,
+ void *userData __unused,
+ aaudio_result_t error) {
+ printf("Error Callback, error: %d\n",(int)error);
+}
+
+// This function is not thread safe. Only use this from a single thread.
+double nextRandomDouble() {
+ return drand48();
+}
+
+class AAudioMonkey : public AAudioSimplePlayer {
+public:
+
+ AAudioMonkey(int index, AAudioArgsParser *argParser)
+ : mArgParser(argParser)
+ , mIndex(index) {}
+
+ aaudio_result_t open() {
+ printf("Monkey # %d ---------------------------------------------- OPEN\n", mIndex);
+ double offset = mIndex * 50;
+ mSine1.setup(440.0, 48000);
+ mSine1.setSweep(300.0 + offset, 600.0 + offset, 5.0);
+ mSine2.setup(660.0, 48000);
+ mSine2.setSweep(350.0 + offset, 900.0 + offset, 7.0);
+
+ aaudio_result_t result = AAudioSimplePlayer::open(*mArgParser,
+ AAudioMonkeyDataCallback,
+ AAudioMonkeyErrorCallbackProc,
+ this);
+ if (result != AAUDIO_OK) {
+ printf("ERROR - player.open() returned %d\n", result);
+ }
+
+ mArgParser->compareWithStream(getStream());
+ return result;
+ }
+
+ bool isOpen() {
+ return (getStream() != nullptr);
+
+ }
+ /**
+ *
+ * @return true if stream passes tests
+ */
+ bool validate() {
+ if (!isOpen()) return true; // closed is OK
+
+ // update and query stream state
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ aaudio_result_t result = AAudioStream_waitForStateChange(getStream(),
+ AAUDIO_STREAM_STATE_UNKNOWN, &state, 0);
+ if (result != AAUDIO_OK) {
+ printf("ERROR - AAudioStream_waitForStateChange returned %d\n", result);
+ return false;
+ }
+
+ int64_t framesRead = AAudioStream_getFramesRead(getStream());
+ int64_t framesWritten = AAudioStream_getFramesWritten(getStream());
+ int32_t xRuns = AAudioStream_getXRunCount(getStream());
+ // Print status
+ printf("%30s, framesWritten = %8lld, framesRead = %8lld, xRuns = %d\n",
+ AAudio_convertStreamStateToText(state),
+ (unsigned long long) framesWritten,
+ (unsigned long long) framesRead,
+ xRuns);
+
+ if (framesWritten < framesRead) {
+ printf("WARNING - UNDERFLOW - diff = %d !!!!!!!!!!!!\n",
+ (int) (framesWritten - framesRead));
+ }
+ return true;
+ }
+
+ aaudio_result_t invoke() {
+ aaudio_result_t result = AAUDIO_OK;
+ if (!isOpen()) {
+ result = open();
+ if (result != AAUDIO_OK) return result;
+ }
+
+ if (!validate()) {
+ return -1;
+ }
+
+ double dice = nextRandomDouble();
+ // Select an action based on a weighted probability.
+ if (dice < PROB_START) {
+ printf("start\n");
+ result = AAudioStream_requestStart(getStream());
+ } else if (dice < PROB_PAUSE) {
+ printf("pause\n");
+ result = AAudioStream_requestPause(getStream());
+ } else if (dice < PROB_FLUSH) {
+ printf("flush\n");
+ result = AAudioStream_requestFlush(getStream());
+ } else if (dice < PROB_STOP) {
+ printf("stop\n");
+ result = AAudioStream_requestStop(getStream());
+ } else if (dice < PROB_CLOSE) {
+ printf("close\n");
+ result = close();
+ } else {
+ printf("do nothing\n");
+ }
+
+ if (result == AAUDIO_ERROR_INVALID_STATE) {
+ printf(" got AAUDIO_ERROR_INVALID_STATE - expected from a monkey\n");
+ result = AAUDIO_OK;
+ }
+ if (result == AAUDIO_OK && isOpen()) {
+ if (!validate()) {
+ result = -1;
+ }
+ }
+ return result;
+ }
+
+ aaudio_data_callback_result_t renderAudio(
+ AAudioStream *stream,
+ void *audioData,
+ int32_t numFrames) {
+
+ int32_t samplesPerFrame = AAudioStream_getChannelCount(stream);
+ // This code only plays on the first one or two channels.
+ // TODO Support arbitrary number of channels.
+ switch (AAudioStream_getFormat(stream)) {
+ case AAUDIO_FORMAT_PCM_I16: {
+ int16_t *audioBuffer = (int16_t *) audioData;
+ // Render sine waves as shorts to first channel.
+ mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+ // Render sine waves to second channel if there is one.
+ if (samplesPerFrame > 1) {
+ mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+ }
+ }
+ break;
+ case AAUDIO_FORMAT_PCM_FLOAT: {
+ float *audioBuffer = (float *) audioData;
+ // Render sine waves as floats to first channel.
+ mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+ // Render sine waves to second channel if there is one.
+ if (samplesPerFrame > 1) {
+ mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+ }
+ }
+ break;
+ default:
+ return AAUDIO_CALLBACK_RESULT_STOP;
+ }
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+ }
+
+private:
+ const AAudioArgsParser *mArgParser;
+ const int mIndex;
+ SineGenerator mSine1;
+ SineGenerator mSine2;
+};
+
+// Callback function that fills the audio output buffer.
+aaudio_data_callback_result_t AAudioMonkeyDataCallback(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames
+) {
+ // should not happen but just in case...
+ if (userData == nullptr) {
+ printf("ERROR - AAudioMonkeyDataCallback needs userData\n");
+ return AAUDIO_CALLBACK_RESULT_STOP;
+ }
+ AAudioMonkey *monkey = (AAudioMonkey *) userData;
+ return monkey->renderAudio(stream, audioData, numFrames);
+}
+
+
+static void usage() {
+ AAudioArgsParser::usage();
+ printf(" -i{seed} Initial random seed\n");
+ printf(" -t{count} number of monkeys in the Troop\n");
+}
+
+int main(int argc, const char **argv) {
+ AAudioArgsParser argParser;
+ std::vector<AAudioMonkey> monkeys;
+ aaudio_result_t result;
+ int numMonkeys = 1;
+
+ // 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 - Monkeys\n", argv[0]);
+
+ long int seed = (long int)getNanoseconds(); // different every time by default
+
+ for (int i = 1; i < argc; i++) {
+ const char *arg = argv[i];
+ if (argParser.parseArg(arg)) {
+ // Handle options that are not handled by the ArgParser
+ if (arg[0] == '-') {
+ char option = arg[1];
+ switch (option) {
+ case 'i':
+ seed = atol(&arg[2]);
+ break;
+ case 't':
+ numMonkeys = atoi(&arg[2]);
+ break;
+ default:
+ usage();
+ exit(EXIT_FAILURE);
+ break;
+ }
+ } else {
+ usage();
+ exit(EXIT_FAILURE);
+ break;
+ }
+ }
+ }
+
+ srand48(seed);
+ printf("seed = %ld, nextRandomDouble() = %f\n", seed, nextRandomDouble());
+
+ for (int m = 0; m < numMonkeys; m++) {
+ monkeys.emplace_back(m, &argParser);
+ }
+
+ for (int i = 0; i < NUM_LOOPS; i++) {
+ // pick a random monkey and invoke it
+ double dice = nextRandomDouble();
+ int monkeyIndex = floor(dice * numMonkeys);
+ printf("----------- Monkey #%d\n", monkeyIndex);
+ result = monkeys[monkeyIndex].invoke();
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
+
+ // sleep some random time
+ dice = nextRandomDouble();
+ dice = dice * dice * dice; // skew towards smaller delays
+ int micros = (int) (dice * MAX_MICROS_DELAY);
+ usleep(micros);
+
+ // TODO consider making this multi-threaded, one thread per monkey, to catch more bugs
+ }
+
+ printf("PASS\n");
+ return EXIT_SUCCESS;
+
+error:
+ printf("FAIL - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+ usleep(1000 * 1000); // give me time to stop the logcat
+ return EXIT_FAILURE;
+}
+
diff --git a/media/libaaudio/tests/test_linear_ramp.cpp b/media/libaaudio/tests/test_linear_ramp.cpp
index 5c53982..93226ba 100644
--- a/media/libaaudio/tests/test_linear_ramp.cpp
+++ b/media/libaaudio/tests/test_linear_ramp.cpp
@@ -15,13 +15,13 @@
*/
#include <iostream>
+#include <math.h>
#include <gtest/gtest.h>
#include "utility/AAudioUtilities.h"
#include "utility/LinearRamp.h"
-
TEST(test_linear_ramp, linear_ramp_segments) {
LinearRamp ramp;
const float source[4] = {1.0f, 1.0f, 1.0f, 1.0f };
@@ -32,40 +32,40 @@
ramp.setLengthInFrames(8);
ramp.setTarget(8.0f);
- ASSERT_EQ(8, ramp.getLengthInFrames());
+ EXPECT_EQ(8, ramp.getLengthInFrames());
bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
- ASSERT_EQ(1, ramping);
- ASSERT_EQ(0.0f, levelFrom);
- ASSERT_EQ(4.0f, levelTo);
+ EXPECT_EQ(1, ramping);
+ EXPECT_EQ(0.0f, levelFrom);
+ EXPECT_EQ(4.0f, levelTo);
AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
- ASSERT_EQ(0.0f, destination[0]);
- ASSERT_EQ(1.0f, destination[1]);
- ASSERT_EQ(2.0f, destination[2]);
- ASSERT_EQ(3.0f, destination[3]);
+ EXPECT_EQ(0.0f, destination[0]);
+ EXPECT_EQ(1.0f, destination[1]);
+ EXPECT_EQ(2.0f, destination[2]);
+ EXPECT_EQ(3.0f, destination[3]);
ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
- ASSERT_EQ(1, ramping);
- ASSERT_EQ(4.0f, levelFrom);
- ASSERT_EQ(8.0f, levelTo);
+ EXPECT_EQ(1, ramping);
+ EXPECT_EQ(4.0f, levelFrom);
+ EXPECT_EQ(8.0f, levelTo);
AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
- ASSERT_EQ(4.0f, destination[0]);
- ASSERT_EQ(5.0f, destination[1]);
- ASSERT_EQ(6.0f, destination[2]);
- ASSERT_EQ(7.0f, destination[3]);
+ EXPECT_EQ(4.0f, destination[0]);
+ EXPECT_EQ(5.0f, destination[1]);
+ EXPECT_EQ(6.0f, destination[2]);
+ EXPECT_EQ(7.0f, destination[3]);
ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
- ASSERT_EQ(0, ramping);
- ASSERT_EQ(8.0f, levelFrom);
- ASSERT_EQ(8.0f, levelTo);
+ EXPECT_EQ(0, ramping);
+ EXPECT_EQ(8.0f, levelFrom);
+ EXPECT_EQ(8.0f, levelTo);
AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
- ASSERT_EQ(8.0f, destination[0]);
- ASSERT_EQ(8.0f, destination[1]);
- ASSERT_EQ(8.0f, destination[2]);
- ASSERT_EQ(8.0f, destination[3]);
+ EXPECT_EQ(8.0f, destination[0]);
+ EXPECT_EQ(8.0f, destination[1]);
+ EXPECT_EQ(8.0f, destination[2]);
+ EXPECT_EQ(8.0f, destination[3]);
};
@@ -80,29 +80,101 @@
ramp.setLengthInFrames(4);
ramp.setTarget(8.0f);
ramp.forceCurrent(4.0f);
- ASSERT_EQ(4.0f, ramp.getCurrent());
+ EXPECT_EQ(4.0f, ramp.getCurrent());
bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
- ASSERT_EQ(1, ramping);
- ASSERT_EQ(4.0f, levelFrom);
- ASSERT_EQ(8.0f, levelTo);
+ EXPECT_EQ(1, ramping);
+ EXPECT_EQ(4.0f, levelFrom);
+ EXPECT_EQ(8.0f, levelTo);
AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
- ASSERT_EQ(4.0f, destination[0]);
- ASSERT_EQ(5.0f, destination[1]);
- ASSERT_EQ(6.0f, destination[2]);
- ASSERT_EQ(7.0f, destination[3]);
+ EXPECT_EQ(4.0f, destination[0]);
+ EXPECT_EQ(5.0f, destination[1]);
+ EXPECT_EQ(6.0f, destination[2]);
+ EXPECT_EQ(7.0f, destination[3]);
ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
- ASSERT_EQ(0, ramping);
- ASSERT_EQ(8.0f, levelFrom);
- ASSERT_EQ(8.0f, levelTo);
+ EXPECT_EQ(0, ramping);
+ EXPECT_EQ(8.0f, levelFrom);
+ EXPECT_EQ(8.0f, levelTo);
AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
- ASSERT_EQ(8.0f, destination[0]);
- ASSERT_EQ(8.0f, destination[1]);
- ASSERT_EQ(8.0f, destination[2]);
- ASSERT_EQ(8.0f, destination[3]);
+ EXPECT_EQ(8.0f, destination[0]);
+ EXPECT_EQ(8.0f, destination[1]);
+ EXPECT_EQ(8.0f, destination[2]);
+ EXPECT_EQ(8.0f, destination[3]);
};
+constexpr int16_t kMaxI16 = INT16_MAX;
+constexpr int16_t kMinI16 = INT16_MIN;
+constexpr int16_t kHalfI16 = 16384;
+constexpr int16_t kTenthI16 = 3277;
+
+//void AAudioConvert_floatToPcm16(const float *source,
+// int16_t *destination,
+// int32_t numSamples,
+// float amplitude);
+TEST(test_linear_ramp, float_to_i16) {
+ const float source[] = {12345.6f, 1.0f, 0.5f, 0.1f, 0.0f, -0.1f, -0.5f, -1.0f, -12345.6f};
+ constexpr size_t count = sizeof(source) / sizeof(source[0]);
+ int16_t destination[count];
+ const int16_t expected[count] = {kMaxI16, kMaxI16, kHalfI16, kTenthI16, 0,
+ -kTenthI16, -kHalfI16, kMinI16, kMinI16};
+
+ AAudioConvert_floatToPcm16(source, destination, count, 1.0f);
+ for (size_t i = 0; i < count; i++) {
+ EXPECT_EQ(expected[i], destination[i]);
+ }
+
+}
+
+//void AAudioConvert_pcm16ToFloat(const int16_t *source,
+// float *destination,
+// int32_t numSamples,
+// float amplitude);
+TEST(test_linear_ramp, i16_to_float) {
+ const int16_t source[] = {kMaxI16, kHalfI16, kTenthI16, 0,
+ -kTenthI16, -kHalfI16, kMinI16};
+ constexpr size_t count = sizeof(source) / sizeof(source[0]);
+ float destination[count];
+ const float expected[count] = {(32767.0f / 32768.0f), 0.5f, 0.1f, 0.0f, -0.1f, -0.5f, -1.0f};
+
+ AAudioConvert_pcm16ToFloat(source, destination, count, 1.0f);
+ for (size_t i = 0; i < count; i++) {
+ EXPECT_NEAR(expected[i], destination[i], 0.0001f);
+ }
+
+}
+
+//void AAudio_linearRamp(const int16_t *source,
+// int16_t *destination,
+// int32_t numFrames,
+// int32_t samplesPerFrame,
+// float amplitude1,
+// float amplitude2);
+TEST(test_linear_ramp, ramp_i16_to_i16) {
+ const int16_t source[] = {1, 1, 1, 1, 1, 1, 1, 1};
+ constexpr size_t count = sizeof(source) / sizeof(source[0]);
+ int16_t destination[count];
+ // Ramp will sweep from -1 to almost +1
+ const int16_t expected[count] = {
+ -1, // from -1.00
+ -1, // from -0.75
+ -1, // from -0.55, round away from zero
+ 0, // from -0.25, round up to zero
+ 0, // from 0.00
+ 0, // from 0.25, round down to zero
+ 1, // from 0.50, round away from zero
+ 1 // from 0.75
+ };
+
+ // sweep across zero to test symmetry
+ constexpr float amplitude1 = -1.0;
+ constexpr float amplitude2 = 1.0;
+ AAudio_linearRamp(source, destination, count, 1, amplitude1, amplitude2);
+ for (size_t i = 0; i < count; i++) {
+ EXPECT_EQ(expected[i], destination[i]);
+ }
+
+}
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 30f97ac..c8fa618 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -197,7 +197,7 @@
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
- mStatus = set(streamType, sampleRate, format, channelMask,
+ (void)set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
@@ -228,7 +228,7 @@
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
- mStatus = set(streamType, sampleRate, format, channelMask,
+ (void)set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
@@ -284,6 +284,11 @@
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
+ status_t status;
+ uint32_t channelCount;
+ pid_t callingPid;
+ pid_t myPid;
+
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
@@ -306,25 +311,29 @@
case TRANSFER_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
@@ -338,7 +347,8 @@
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
- return INVALID_OPERATION;
+ status = INVALID_OPERATION;
+ goto exit;
}
// handle default values first.
@@ -348,7 +358,8 @@
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("Invalid stream type %d", streamType);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mStreamType = streamType;
@@ -380,16 +391,18 @@
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mChannelMask = channelMask;
- uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
+ channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// force direct flag if format is not linear PCM
@@ -424,7 +437,8 @@
// sampling rate must be specified for direct outputs
if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
@@ -455,12 +469,14 @@
if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
ALOGE("notificationFrames=%d not permitted for non-fast track",
notificationFrames);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
if (frameCount > 0) {
ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu",
notificationFrames, frameCount);
- return BAD_VALUE;
+ status = BAD_VALUE;
+ goto exit;
}
mNotificationFramesReq = 0;
const uint32_t minNotificationsPerBuffer = 1;
@@ -472,15 +488,15 @@
notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
}
mNotificationFramesAct = 0;
- int callingpid = IPCThreadState::self()->getCallingPid();
- int mypid = getpid();
- if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) {
+ callingPid = IPCThreadState::self()->getCallingPid();
+ myPid = getpid();
+ if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
- if (pid == -1 || (callingpid != mypid)) {
- mClientPid = callingpid;
+ if (pid == -1 || (callingPid != myPid)) {
+ mClientPid = callingPid;
} else {
mClientPid = pid;
}
@@ -495,7 +511,7 @@
}
// create the IAudioTrack
- status_t status = createTrack_l();
+ status = createTrack_l();
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -503,10 +519,9 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
- return status;
+ goto exit;
}
- mStatus = NO_ERROR;
mUserData = user;
mLoopCount = 0;
mLoopStart = 0;
@@ -534,7 +549,10 @@
mFramesWrittenServerOffset = 0;
mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
mVolumeHandler = new media::VolumeHandler();
- return NO_ERROR;
+
+exit:
+ mStatus = status;
+ return status;
}
// -------------------------------------------------------------------------
@@ -1278,15 +1296,16 @@
status_t AudioTrack::createTrack_l()
{
+ status_t status;
+ bool callbackAdded = false;
+
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
- return NO_INIT;
+ status = NO_INIT;
+ goto exit;
}
- status_t status;
- bool callbackAdded = false;
-
{
// mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
// After fast request is denied, we will request again if IAudioTrack is re-created.
@@ -1355,7 +1374,10 @@
if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("AudioFlinger could not create track, status: %d output %d", status, output.outputId);
- goto error;
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ goto exit;
}
ALOG_ASSERT(track != 0);
@@ -1383,13 +1405,13 @@
if (iMem == 0) {
ALOGE("Could not get control block");
status = NO_INIT;
- goto error;
+ goto exit;
}
void *iMemPointer = iMem->pointer();
if (iMemPointer == NULL) {
ALOGE("Could not get control block pointer");
status = NO_INIT;
- goto error;
+ goto exit;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
@@ -1443,7 +1465,7 @@
if (buffers == NULL) {
ALOGE("Could not get buffer pointer");
status = NO_INIT;
- goto error;
+ goto exit;
}
}
@@ -1486,17 +1508,15 @@
mDeathNotifier = new DeathNotifier(this);
IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
- return NO_ERROR;
}
-error:
- if (callbackAdded) {
+exit:
+ if (status != NO_ERROR && callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput);
}
- if (status == NO_ERROR) {
- status = NO_INIT;
- }
+
+ mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;