Merge "Fix AudioMixer floating interaction with downmixer"

4 files changed, 136 insertions, 56 deletions

diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index beab7c2..0d4b358 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -430,6 +430,10 @@ void AudioMixer::setLog(NBLog::Writer *log)
     mState.mLog = log;
 }
 
+static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
+    return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+}
+
 int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
         audio_format_t format, int sessionId)
 {
@@ -492,10 +496,11 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
         t->mInputBufferProvider = NULL;
         t->mReformatBufferProvider = NULL;
         t->downmixerBufferProvider = NULL;
+        t->mPostDownmixReformatBufferProvider = NULL;
         t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
         t->mFormat = format;
-        t->mMixerInFormat = kUseFloat && kUseNewMixer
-                ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+        t->mMixerInFormat = selectMixerInFormat(format);
+        t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
         t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
                 AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
         t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
@@ -505,9 +510,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
             ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
             return -1;
         }
-        // prepareForDownmix() may change the input format requirement.
-        // If you desire floating point input to the mixer, it may change
-        // to integer because the downmixer requires integer to process.
+        // prepareForDownmix() may change mDownmixRequiresFormat
         ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
         t->prepareForReformat();
         mTrackNames |= 1 << n;
@@ -526,7 +529,7 @@ void AudioMixer::invalidateState(uint32_t mask)
  }
 
 // Called when channel masks have changed for a track name
-// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format,
+// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
 // which will simplify this logic.
 bool AudioMixer::setChannelMasks(int name,
         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
@@ -551,21 +554,18 @@ bool AudioMixer::setChannelMasks(int name,
 
     // channel masks have changed, does this track need a downmixer?
     // update to try using our desired format (if we aren't already using it)
-    const audio_format_t prevMixerInFormat = track.mMixerInFormat;
-    track.mMixerInFormat = kUseFloat && kUseNewMixer
-            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+    const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
     const status_t status = mState.tracks[name].prepareForDownmix();
     ALOGE_IF(status != OK,
             "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
             status, track.channelMask, track.mMixerChannelMask);
 
-    const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat;
-    if (mixerInFormatChanged) {
+    if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
         track.prepareForReformat(); // because of downmixer, track format may change!
     }
 
-    if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) {
-        // resampler input format or channels may have changed.
+    if (track.resampler && mixerChannelCountChanged) {
+        // resampler channels may have changed.
         const uint32_t resetToSampleRate = track.sampleRate;
         delete track.resampler;
         track.resampler = NULL;
@@ -579,6 +579,7 @@ bool AudioMixer::setChannelMasks(int name,
 void AudioMixer::track_t::unprepareForDownmix() {
     ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
 
+    mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
     if (downmixerBufferProvider != NULL) {
         // this track had previously been configured with a downmixer, delete it
         ALOGV(" deleting old downmixer");
@@ -611,7 +612,7 @@ status_t AudioMixer::track_t::prepareForDownmix()
                 sampleRate, sessionId, kCopyBufferFrameCount);
 
         if (pDbp->isValid()) { // if constructor completed properly
-            mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+            mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
             downmixerBufferProvider = pDbp;
             reconfigureBufferProviders();
             return NO_ERROR;
@@ -630,9 +631,18 @@ status_t AudioMixer::track_t::prepareForDownmix()
 
 void AudioMixer::track_t::unprepareForReformat() {
     ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+    bool requiresReconfigure = false;
     if (mReformatBufferProvider != NULL) {
         delete mReformatBufferProvider;
         mReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (mPostDownmixReformatBufferProvider != NULL) {
+        delete mPostDownmixReformatBufferProvider;
+        mPostDownmixReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
         reconfigureBufferProviders();
     }
 }
@@ -640,14 +650,29 @@ void AudioMixer::track_t::unprepareForReformat() {
 status_t AudioMixer::track_t::prepareForReformat()
 {
     ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
-    // discard the previous reformatter if there was one
+    // discard previous reformatters
     unprepareForReformat();
-    // only configure reformatter if needed
-    if (mFormat != mMixerInFormat) {
+    // only configure reformatters as needed
+    const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
+            ? mDownmixRequiresFormat : mMixerInFormat;
+    bool requiresReconfigure = false;
+    if (mFormat != targetFormat) {
         mReformatBufferProvider = new ReformatBufferProvider(
                 audio_channel_count_from_out_mask(channelMask),
-                mFormat, mMixerInFormat,
+                mFormat,
+                targetFormat,
                 kCopyBufferFrameCount);
+        requiresReconfigure = true;
+    }
+    if (targetFormat != mMixerInFormat) {
+        mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+                audio_channel_count_from_out_mask(mMixerChannelMask),
+                targetFormat,
+                mMixerInFormat,
+                kCopyBufferFrameCount);
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
         reconfigureBufferProviders();
     }
     return NO_ERROR;
@@ -664,6 +689,10 @@ void AudioMixer::track_t::reconfigureBufferProviders()
         downmixerBufferProvider->setBufferProvider(bufferProvider);
         bufferProvider = downmixerBufferProvider;
     }
+    if (mPostDownmixReformatBufferProvider) {
+        mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mPostDownmixReformatBufferProvider;
+    }
 }
 
 void AudioMixer::deleteTrackName(int name)
@@ -1026,6 +1055,9 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider
     if (mState.tracks[name].mReformatBufferProvider != NULL) {
         mState.tracks[name].mReformatBufferProvider->reset();
     } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+        mState.tracks[name].downmixerBufferProvider->reset();
+    } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
+        mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
     }
 
     mState.tracks[name].mInputBufferProvider = bufferProvider;
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 6ed6c1b..88e94c5 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -205,17 +205,34 @@ private:
         int32_t*           auxBuffer;
 
         // 16-byte boundary
+
+        /* Buffer providers are constructed to translate the track input data as needed.
+         *
+         * 1) mInputBufferProvider: The AudioTrack buffer provider.
+         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+         *    requires reformat. For example, it may convert floating point input to
+         *    PCM_16_bit if that's required by the downmixer.
+         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+         *    the number of channels required by the mixer sink.
+         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+         *    the downmixer requirements to the mixer engine input requirements.
+         */
         AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
         CopyBufferProvider*      mReformatBufferProvider; // provider wrapper for reformatting.
         CopyBufferProvider*      downmixerBufferProvider; // wrapper for channel conversion.
+        CopyBufferProvider*      mPostDownmixReformatBufferProvider;
 
+        // 16-byte boundary
         int32_t     sessionId;
 
-        // 16-byte boundary
         audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
         audio_format_t mFormat;          // input track format
         audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
                                          // each track must be converted to this format.
+        audio_format_t mDownmixRequiresFormat;  // required downmixer format
+                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+                                                // AUDIO_FORMAT_INVALID if no required format
 
         float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
         float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
@@ -225,7 +242,6 @@ private:
         float          mPrevAuxLevel;                 // floating point prev aux level
         float          mAuxInc;                       // floating point aux increment
 
-        // 16-byte boundary
         audio_channel_mask_t mMixerChannelMask;
         uint32_t             mMixerChannelCount;
 
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index 9b39e77..e60e6d5 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -63,8 +63,18 @@ function createwav() {
 # process__genericResampling
 # track__Resample / track__genericResample
     adb shell test-mixer $1 -s 48000 \
+        -o /sdcard/tm48000grif.wav \
+        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+        sine:f,6,6000,19000  chirp:i,4,30000
+    adb pull /sdcard/tm48000grif.wav $2
+
+# Test:
+# process__genericResampling
+# track__Resample / track__genericResample
+    adb shell test-mixer $1 -s 48000 \
         -o /sdcard/tm48000gr.wav \
-        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+        sine:6,6000,19000
     adb pull /sdcard/tm48000gr.wav $2
 
 # Test:
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 @@ static void usage(const char* name) {
     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 @@ static void usage(const char* name) {
     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 @@ static int writeFile(const char *filename, const void *buffer,
     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 @@ int main(int argc, char* argv[]) {
     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 @@ int main(int argc, char* argv[]) {
     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 @@ int main(int argc, char* argv[]) {
     // 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 @@ int main(int argc, char* argv[]) {
                 name,
                 AudioMixer::TRACK,
                 AudioMixer::FORMAT,
-                (void *)(uintptr_t)inputFormat);
+                (void *)(uintptr_t)formats[i]);
         mixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -255,7 +277,7 @@ int main(int argc, char* argv[]) {
                 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 @@ int main(int argc, char* argv[]) {
     // 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);
             }
         }