Fix AudioMixer floating interaction with downmixer
Previously _if_ the full floating point mixer is enabled, a downmixer
would force the mixer input for a session submix to integer, breaking
other mixer inputs to the same submix that were in float.
Use another ReformatBufferProvider after the downmixer to solve
this issue.
Update the test-mixer app and the mixer_to_wave_tests shell
script to detect this issue.
Bug: 17363939
Change-Id: I74a56333f9ee75ddde39a75392c021c5eebddbef
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 9a4fad6..8da6245 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -39,7 +39,7 @@
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, " -f enable floating point input track by default\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");
@@ -47,8 +47,8 @@
fprintf(stderr, " -a <aux-buffer-file>\n");
fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n");
fprintf(stderr, " <input-file> is a WAV file\n");
- fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
- fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n");
+ fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
+ fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n");
}
static int writeFile(const char *filename, const void *buffer,
@@ -78,6 +78,18 @@
return EXIT_SUCCESS;
}
+const char *parseFormat(const char *s, bool *useFloat) {
+ if (!strncmp(s, "f,", 2)) {
+ *useFloat = true;
+ return s + 2;
+ }
+ if (!strncmp(s, "i,", 2)) {
+ *useFloat = false;
+ return s + 2;
+ }
+ return s;
+}
+
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
bool useInputFloat = false;
@@ -88,8 +100,9 @@
std::vector<int> Pvalues;
const char* outputFilename = NULL;
const char* auxFilename = NULL;
- std::vector<int32_t> Names;
- std::vector<SignalProvider> Providers;
+ std::vector<int32_t> names;
+ std::vector<SignalProvider> providers;
+ std::vector<audio_format_t> formats;
for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
switch (ch) {
@@ -138,54 +151,65 @@
size_t outputFrames = 0;
// create providers for each track
- Providers.resize(argc);
+ names.resize(argc);
+ providers.resize(argc);
+ formats.resize(argc);
for (int i = 0; i < argc; ++i) {
static const char chirp[] = "chirp:";
static const char sine[] = "sine:";
static const double kSeconds = 1;
+ bool useFloat = useInputFloat;
if (!strncmp(argv[i], chirp, strlen(chirp))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
- parseCSV(argv[i] + strlen(chirp), v);
+ parseCSV(s, v);
if (v.size() == 2) {
printf("creating chirp(%d %d)\n", v[0], v[1]);
- if (useInputFloat) {
- Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ if (useFloat) {
+ providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else if (!strncmp(argv[i], sine, strlen(sine))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
- parseCSV(argv[i] + strlen(sine), v);
+ parseCSV(s, v);
if (v.size() == 3) {
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);
+ if (useFloat) {
+ providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else {
printf("creating filename(%s)\n", argv[i]);
if (useInputFloat) {
- Providers[i].setFile<float>(argv[i]);
+ providers[i].setFile<float>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setFile<short>(argv[i]);
+ providers[i].setFile<short>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
}
// calculate the number of output frames
- size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
- / Providers[i].getSampleRate();
+ size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
+ / providers[i].getSampleRate();
if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
outputFrames = nframes;
}
@@ -213,22 +237,20 @@
// create the mixer.
const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
- audio_format_t inputFormat = useInputFloat
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
audio_format_t mixerFormat = useMixerFloat
? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
- float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+ float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
static float f0; // zero
// set up the tracks.
- for (size_t i = 0; i < Providers.size(); ++i) {
- //printf("track %d out of %d\n", i, Providers.size());
- uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+ for (size_t i = 0; i < providers.size(); ++i) {
+ //printf("track %d out of %d\n", i, providers.size());
+ uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
int32_t name = mixer->getTrackName(channelMask,
- inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+ formats[i], AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
- Names.push_back(name);
- mixer->setBufferProvider(name, &Providers[i]);
+ names[i] = name;
+ mixer->setBufferProvider(name, &providers[i]);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *)outputAddr);
mixer->setParameter(
@@ -240,7 +262,7 @@
name,
AudioMixer::TRACK,
AudioMixer::FORMAT,
- (void *)(uintptr_t)inputFormat);
+ (void *)(uintptr_t)formats[i]);
mixer->setParameter(
name,
AudioMixer::TRACK,
@@ -255,7 +277,7 @@
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
- (void *)(uintptr_t)Providers[i].getSampleRate());
+ (void *)(uintptr_t)providers[i].getSampleRate());
if (useRamp) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
@@ -277,11 +299,11 @@
// pump the mixer to process data.
size_t i;
for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
- for (size_t j = 0; j < Names.size(); ++j) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ for (size_t j = 0; j < names.size(); ++j) {
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(char *) outputAddr + i * outputFrameSize);
if (auxFilename) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
(char *) auxAddr + i * auxFrameSize);
}
}