aaudio: test 24 and 32 bit formats

Modify write sine tests.

Bug: 75038760
Test: adb shell write_sine_callback -f3
Test: adb shell write_sine_callback -f4
Change-Id: I61eccbffbf898e50ee241e135d21acebb6a63524
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index 4bba436..e670642 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -421,7 +421,9 @@
         printf("      -f{0|1|2} set format\n");
         printf("          0 = UNSPECIFIED\n");
         printf("          1 = PCM_I16\n");
-        printf("          2 = FLOAT\n");
+        printf("          2 = PCM_FLOAT\n");
+        printf("          3 = PCM_I24_PACKED\n");
+        printf("          4 = PCM_I32\n");
         printf("      -i{inputPreset} eg. 5 for AAUDIO_INPUT_PRESET_CAMCORDER\n");
         printf("      -m{0|1|2|3} set MMAP policy\n");
         printf("          0 = _UNSPECIFIED, use aaudio.mmap_policy system property, default\n");
diff --git a/media/libaaudio/examples/utils/AAudioExampleUtils.h b/media/libaaudio/examples/utils/AAudioExampleUtils.h
index 46b8895..5819dfd 100644
--- a/media/libaaudio/examples/utils/AAudioExampleUtils.h
+++ b/media/libaaudio/examples/utils/AAudioExampleUtils.h
@@ -32,6 +32,7 @@
 #define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
 #define NANOS_PER_SECOND      (NANOS_PER_MILLISECOND * 1000)
 
+// Use template functions to avoid warning of unused static functions.
 template <class T = aaudio_sharing_mode_t>
 const char *getSharingModeText(aaudio_sharing_mode_t mode) {
     const char *text = "unknown";
@@ -48,6 +49,7 @@
     return text;
 }
 
+template <class T = aaudio_performance_mode_t>
 const char *getPerformanceModeText(aaudio_performance_mode_t mode) {
     const char *text = "unknown";
     switch (mode) {
@@ -66,6 +68,7 @@
     return text;
 }
 
+template <class T = aaudio_direction_t>
 const char *getDirectionText(aaudio_direction_t direction) {
     const char *text = "unknown";
     switch (direction) {
@@ -81,6 +84,29 @@
     return text;
 }
 
+template <class T = aaudio_direction_t>
+constexpr int32_t getBytesPerSample(aaudio_format_t format) {
+    switch (format) {
+        case AAUDIO_FORMAT_PCM_I16:
+            return 2;
+        case AAUDIO_FORMAT_PCM_FLOAT:
+            return 4;
+        case AAUDIO_FORMAT_PCM_I24_PACKED:
+            return 3;
+        case AAUDIO_FORMAT_PCM_I32:
+            return 4;
+        default:
+            return -1;
+    }
+}
+
+// Return true if CPU is native Little Endian
+inline bool isNativeLittleEndian() {
+    // If the first byte of the data word in memory is 1 then Little Endian.
+    constexpr union { unsigned u; unsigned char c[sizeof(unsigned)]; } one = {1};
+    return one.c[0] != 0;
+}
+
 template <class T = int64_t>
 void convertNanosecondsToTimespec(int64_t nanoseconds, struct timespec *time) {
     time->tv_sec = nanoseconds / NANOS_PER_SECOND;
diff --git a/media/libaaudio/examples/utils/AAudioSimplePlayer.h b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
index fd1fc45..7daac20 100644
--- a/media/libaaudio/examples/utils/AAudioSimplePlayer.h
+++ b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
@@ -359,22 +359,38 @@
 
     int32_t samplesPerFrame = AAudioStream_getChannelCount(stream);
 
-
-    int numActiveOscilators = (samplesPerFrame > MAX_CHANNELS) ? MAX_CHANNELS : samplesPerFrame;
+    int numActiveOscillators = std::min(samplesPerFrame, MAX_CHANNELS);
     switch (AAudioStream_getFormat(stream)) {
         case AAUDIO_FORMAT_PCM_I16: {
             int16_t *audioBuffer = (int16_t *) audioData;
-            for (int i = 0; i < numActiveOscilators; ++i) {
-                sineData->sineOscillators[i].render(&audioBuffer[i], samplesPerFrame,
-                                                    numFrames);
+            for (int i = 0; i < numActiveOscillators; ++i) {
+                sineData->sineOscillators[i].render(&audioBuffer[i],
+                                                    samplesPerFrame, numFrames);
             }
         }
             break;
         case AAUDIO_FORMAT_PCM_FLOAT: {
             float *audioBuffer = (float *) audioData;
-            for (int i = 0; i < numActiveOscilators; ++i) {
-                sineData->sineOscillators[i].render(&audioBuffer[i], samplesPerFrame,
-                                                    numFrames);
+            for (int i = 0; i < numActiveOscillators; ++i) {
+                sineData->sineOscillators[i].render(&audioBuffer[i],
+                                                    samplesPerFrame, numFrames);
+            }
+        }
+            break;
+        case AAUDIO_FORMAT_PCM_I24_PACKED: {
+            uint8_t *audioBuffer = (uint8_t *) audioData;
+            for (int i = 0; i < numActiveOscillators; ++i) {
+                static const int bytesPerSample = getBytesPerSample(AAUDIO_FORMAT_PCM_I24_PACKED);
+                sineData->sineOscillators[i].render24(&audioBuffer[i * bytesPerSample],
+                                                      samplesPerFrame, numFrames);
+            }
+        }
+            break;
+        case AAUDIO_FORMAT_PCM_I32: {
+            int32_t *audioBuffer = (int32_t *) audioData;
+            for (int i = 0; i < numActiveOscillators; ++i) {
+                sineData->sineOscillators[i].render(&audioBuffer[i],
+                                                    samplesPerFrame, numFrames);
             }
         }
             break;
diff --git a/media/libaaudio/examples/utils/SineGenerator.h b/media/libaaudio/examples/utils/SineGenerator.h
index 9e6d46d..66a08fd 100644
--- a/media/libaaudio/examples/utils/SineGenerator.h
+++ b/media/libaaudio/examples/utils/SineGenerator.h
@@ -41,20 +41,54 @@
         }
     }
 
+    float next() {
+        float value = sinf(mPhase) * mAmplitude;
+        advancePhase();
+        return value;
+    }
+
     void render(int16_t *buffer, int32_t channelStride, int32_t numFrames) {
         int sampleIndex = 0;
         for (int i = 0; i < numFrames; i++) {
-            buffer[sampleIndex] = (int16_t) (INT16_MAX * sin(mPhase) * mAmplitude);
+            buffer[sampleIndex] = (int16_t) (INT16_MAX * next());
             sampleIndex += channelStride;
-            advancePhase();
         }
     }
+
     void render(float *buffer, int32_t channelStride, int32_t numFrames) {
         int sampleIndex = 0;
         for (int i = 0; i < numFrames; i++) {
-            buffer[sampleIndex] = sin(mPhase) * mAmplitude;
+            buffer[sampleIndex] = next();
             sampleIndex += channelStride;
-            advancePhase();
+        }
+    }
+
+    void render(int32_t *buffer, int32_t channelStride, int32_t numFrames) {
+        int sampleIndex = 0;
+        for (int i = 0; i < numFrames; i++) {
+            buffer[sampleIndex] = (int32_t) (INT32_MAX * next());
+            sampleIndex += channelStride;
+        }
+    }
+
+    void render24(uint8_t *buffer, int32_t channelStride, int32_t numFrames) {
+        int sampleIndex = 0;
+        constexpr int32_t INT24_MAX = (1 << 23) - 1;
+        constexpr int bytesPerSample = getBytesPerSample(AAUDIO_FORMAT_PCM_I24_PACKED);
+        const bool isLittleEndian = isNativeLittleEndian();
+        for (int i = 0; i < numFrames; i++) {
+            int32_t sample = (int32_t) (INT24_MAX * next());
+            uint32_t usample = (uint32_t) sample;
+            if (isLittleEndian) {
+                buffer[sampleIndex] = usample; // little end first
+                buffer[sampleIndex + 1] = usample >> 8;
+                buffer[sampleIndex + 2] = usample >> 16;
+            } else {
+                buffer[sampleIndex] = usample >> 16; // big end first
+                buffer[sampleIndex + 1] = usample >> 8;
+                buffer[sampleIndex + 2] = usample;
+            }
+            sampleIndex += channelStride * bytesPerSample;
         }
     }
 
@@ -100,4 +134,3 @@
 };
 
 #endif /* SINE_GENERATOR_H */
-