Add multichannel capability to AudioMixer

tests/test-mixer updated to handle multichannel.
New mixer path is still disabled (kUseNewMixer = false).
Subsequent CL will enable multichannel capability.

Change-Id: I71dafa94cc1d9af567c285e0ded7c5c94e3ff926

3 files changed, 30 insertions, 10 deletions

diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index 93bff47..9b39e77 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -72,9 +72,9 @@ function createwav() {
 # track__Resample / track__genericResample
 # track__NoResample / track__16BitsStereo / track__16BitsMono
 # Aux buffer
-    adb shell test-mixer $1 -s 9307 \
+    adb shell test-mixer $1 -c 5 -s 9307 \
         -a /sdcard/aux9307gra.wav -o /sdcard/tm9307gra.wav \
-        sine:2,1000,3000 sine:1,2000,9307 chirp:2,9307
+        sine:4,1000,3000 sine:1,2000,9307 chirp:3,9307
     adb pull /sdcard/tm9307gra.wav $2
     adb pull /sdcard/aux9307gra.wav $2
 
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 3940702..5a00f40 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -36,11 +36,12 @@
 using namespace android;
 
 static void usage(const char* name) {
-    fprintf(stderr, "Usage: %s [-f] [-m]"
+    fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]"
                     " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
                     " (<input-file> | <command>)+\n", name);
     fprintf(stderr, "    -f    enable floating point input track\n");
     fprintf(stderr, "    -m    enable floating point mixer output\n");
+    fprintf(stderr, "    -c    number of mixer output channels\n");
     fprintf(stderr, "    -s    mixer sample-rate\n");
     fprintf(stderr, "    -o    <output-file> WAV file, pcm16 (or float if -m specified)\n");
     fprintf(stderr, "    -a    <aux-buffer-file>\n");
@@ -90,7 +91,7 @@ int main(int argc, char* argv[]) {
     std::vector<int32_t> Names;
     std::vector<SignalProvider> Providers;
 
-    for (int ch; (ch = getopt(argc, argv, "fms:o:a:P:")) != -1;) {
+    for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
         switch (ch) {
         case 'f':
             useInputFloat = true;
@@ -98,6 +99,9 @@ int main(int argc, char* argv[]) {
         case 'm':
             useMixerFloat = true;
             break;
+        case 'c':
+            outputChannels = atoi(optarg);
+            break;
         case 's':
             outputSampleRate = atoi(optarg);
             break;
@@ -160,7 +164,7 @@ int main(int argc, char* argv[]) {
 
             parseCSV(argv[i] + strlen(sine), v);
             if (v.size() == 3) {
-                printf("creating sine(%d %d)\n", v[0], v[1]);
+                printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
                 if (useInputFloat) {
                     Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
                 } else {
@@ -191,6 +195,8 @@ int main(int argc, char* argv[]) {
     const size_t outputFrameSize = outputChannels
             * (useMixerFloat ? sizeof(float) : sizeof(int16_t));
     const size_t outputSize = outputFrames * outputFrameSize;
+    const audio_channel_mask_t outputChannelMask =
+            audio_channel_out_mask_from_count(outputChannels);
     void *outputAddr = NULL;
     (void) posix_memalign(&outputAddr, 32, outputSize);
     memset(outputAddr, 0, outputSize);
@@ -224,15 +230,29 @@ int main(int argc, char* argv[]) {
         Names.push_back(name);
         mixer->setBufferProvider(name, &Providers[i]);
         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
-                (void *) outputAddr);
+                (void *)outputAddr);
         mixer->setParameter(
                 name,
                 AudioMixer::TRACK,
-                AudioMixer::MIXER_FORMAT, (void *)mixerFormat);
-        mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+                AudioMixer::MIXER_FORMAT,
+                (void *)(uintptr_t)mixerFormat);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT,
                 (void *)(uintptr_t)inputFormat);
         mixer->setParameter(
                 name,
+                AudioMixer::TRACK,
+                AudioMixer::MIXER_CHANNEL_MASK,
+                (void *)(uintptr_t)outputChannelMask);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_MASK,
+                (void *)(uintptr_t)channelMask);
+        mixer->setParameter(
+                name,
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
                 (void *)(uintptr_t)Providers[i].getSampleRate());
diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h
index f954292..e446216 100644
--- a/services/audioflinger/tests/test_utils.h
+++ b/services/audioflinger/tests/test_utils.h
@@ -195,7 +195,7 @@ static void createSine(void *vbuffer, size_t frames,
         T yt = convertValue<T>(y);
 
         for (size_t j = 0; j < channels; ++j) {
-            buffer[i*channels + j] = yt / (j + 1);
+            buffer[i*channels + j] = yt / T(j + 1);
         }
     }
 }
@@ -221,7 +221,7 @@ static void createChirp(void *vbuffer, size_t frames,
         T yt = convertValue<T>(y);
 
         for (size_t j = 0; j < channels; ++j) {
-            buffer[i*channels + j] = yt / (j + 1);
+            buffer[i*channels + j] = yt / T(j + 1);
         }
     }
 }