Add new AudioMixer processing hooks
This change adds new process and thread hooks for
multi-format handling.
It is enabled by setting kUseNewMixer = true.
Change-Id: I262a3d2e4411f8cef7370a497b77a34eb55f1f86
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index d73292e..af312c4 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -40,8 +40,36 @@
#include <media/EffectsFactoryApi.h>
+#include "AudioMixerOps.h"
#include "AudioMixer.h"
+// Use the FCC_2 macro for code assuming Fixed Channel Count of 2 and
+// whose stereo assumption may need to be revisited later.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// Set kUseNewMixer to true to use the new mixer engine. Otherwise the
+// original code will be used. This is false for now.
+static const bool kUseNewMixer = false;
+
+// Set kUseFloat to true to allow floating input into the mixer engine.
+// If kUseNewMixer is false, this is ignored or may be overridden internally
+// because of downmix/upmix support.
+static const bool kUseFloat = true;
+
namespace android {
// ----------------------------------------------------------------------------
@@ -300,15 +328,19 @@
t->downmixerBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
- t->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT;
- if (t->mFormat != t->mMixerInFormat) {
- prepareTrackForReformat(t, n);
- }
- status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
+ t->mMixerInFormat = kUseFloat && kUseNewMixer
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ // Check the downmixing (or upmixing) requirements.
+ status_t status = initTrackDownmix(t, n, channelMask);
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return -1;
}
+ // initTrackDownmix() 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.
+ ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+ prepareTrackForReformat(t, n);
mTrackNames |= 1 << n;
return TRACK0 + n;
}
@@ -443,6 +475,7 @@
}// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack"
// initialization successful:
+ pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // 16 bit input is required for downmix
pTrack->downmixerBufferProvider = pDbp;
reconfigureBufferProviders(pTrack);
return NO_ERROR;
@@ -467,12 +500,15 @@
{
ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
// discard the previous reformatter if there was one
- unprepareTrackForReformat(pTrack, trackName);
- pTrack->mReformatBufferProvider = new ReformatBufferProvider(
- audio_channel_count_from_out_mask(pTrack->channelMask),
- pTrack->mFormat, pTrack->mMixerInFormat);
- reconfigureBufferProviders(pTrack);
- return NO_ERROR;
+ unprepareTrackForReformat(pTrack, trackName);
+ // only configure reformatter if needed
+ if (pTrack->mFormat != pTrack->mMixerInFormat) {
+ pTrack->mReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(pTrack->channelMask),
+ pTrack->mFormat, pTrack->mMixerInFormat);
+ reconfigureBufferProviders(pTrack);
+ }
+ return NO_ERROR;
}
void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
@@ -596,8 +632,15 @@
track.channelMask = mask;
track.channelCount = channelCount;
// the mask has changed, does this track need a downmixer?
- initTrackDownmix(&mState.tracks[name], name, mask);
+ // update to try using our desired format (if we aren't already using it)
+ track.mMixerInFormat = kUseFloat && kUseNewMixer
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ status_t status = initTrackDownmix(&mState.tracks[name], name, mask);
+ ALOGE_IF(status != OK,
+ "Invalid channel mask %#x, initTrackDownmix returned %d",
+ mask, status);
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
+ prepareTrackForReformat(&track, name); // format may have changed
invalidateState(1 << name);
}
} break;
@@ -621,11 +664,7 @@
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track.mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
- //if (track.mFormat != track.mMixerInFormat)
- {
- ALOGD("Reformatting!");
- prepareTrackForReformat(&track, name);
- }
+ prepareTrackForReformat(&track, name);
invalidateState(1 << name);
}
} break;
@@ -723,7 +762,20 @@
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
- const int bits = mMixerInFormat == AUDIO_FORMAT_PCM_16_BIT ? 16 : /* FLOAT */ 32;
+
+ int bits;
+ switch (mMixerInFormat) {
+ case AUDIO_FORMAT_PCM_16_BIT:
+ bits = 16;
+ break;
+ case AUDIO_FORMAT_PCM_FLOAT:
+ bits = 32; // 32 bits to the AudioResampler::create() indicates float.
+ break;
+ default:
+ LOG_ALWAYS_FATAL("Invalid mMixerInFormat: %#x", mMixerInFormat);
+ break;
+ }
+ ALOGVV("Creating resampler with %d bits\n", bits);
resampler = AudioResampler::create(
bits,
// the resampler sees the number of channels after the downmixer, if any
@@ -848,16 +900,19 @@
if (n & NEEDS_RESAMPLE) {
all16BitsStereoNoResample = false;
resampling = true;
- t.hook = track__genericResample;
+ t.hook = getTrackHook(TRACKTYPE_RESAMPLE, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix + resample", i);
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
- t.hook = track__16BitsMono;
+ t.hook = getTrackHook(TRACKTYPE_NORESAMPLEMONO, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
- t.hook = track__16BitsStereo;
+ t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix", i);
}
@@ -888,7 +943,10 @@
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ track_t& t = state->tracks[i];
+ state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
}
}
}
@@ -931,6 +989,7 @@
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
int32_t* temp, int32_t* aux)
{
+ ALOGVV("track__genericResample\n");
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
@@ -1042,6 +1101,7 @@
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
+ ALOGVV("track__16BitsStereo\n");
const int16_t *in = static_cast<const int16_t *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
@@ -1133,6 +1193,7 @@
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
+ ALOGVV("track__16BitsMono\n");
const int16_t *in = static_cast<int16_t const *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
@@ -1220,6 +1281,7 @@
// no-op case
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
+ ALOGVV("process__nop\n");
uint32_t e0 = state->enabledTracks;
size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
while (e0) {
@@ -1267,6 +1329,7 @@
// generic code without resampling
void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
+ ALOGVV("process__genericNoResampling\n");
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
// acquire each track's buffer
@@ -1349,18 +1412,12 @@
}
}
}
- switch (t1.mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
- out += BLOCKSIZE * 2; // output is 2 floats/frame.
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- ditherAndClamp(out, outTemp, BLOCKSIZE);
- out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
- }
+
+ convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
+ BLOCKSIZE * FCC_2);
+ // TODO: fix ugly casting due to choice of out pointer type
+ out = reinterpret_cast<int32_t*>((uint8_t*)out
+ + BLOCKSIZE * FCC_2 * audio_bytes_per_sample(t1.mMixerFormat));
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
@@ -1379,6 +1436,7 @@
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
+ ALOGVV("process__genericResampling\n");
// this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1442,16 +1500,7 @@
}
}
}
- switch (t1.mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27(reinterpret_cast<float*>(out), outTemp, numFrames*2);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- ditherAndClamp(out, outTemp, numFrames);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
- }
+ convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat, numFrames * FCC_2);
}
}
@@ -1459,6 +1508,7 @@
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
+ ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
// This method is only called when state->enabledTracks has exactly
// one bit set. The asserts below would verify this, but are commented out
// since the whole point of this method is to optimize performance.
@@ -1683,5 +1733,275 @@
ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
}
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
+{
+ ALOGVV("process_NoResampleOneTrack\n");
+ // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+ track_t *t = &state->tracks[i];
+ TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+ TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+ const bool ramp = t->needsRamp();
+
+ for (size_t numFrames = state->frameCount; numFrames; ) {
+ AudioBufferProvider::Buffer& b(t->buffer);
+ // get input buffer
+ b.frameCount = numFrames;
+ const int64_t outputPTS = calculateOutputPTS(*t, pts, state->frameCount - numFrames);
+ t->bufferProvider->getNextBuffer(&b, outputPTS);
+ const TI *in = reinterpret_cast<TI*>(b.raw);
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ memset(out, 0, numFrames
+ * NCHAN * audio_bytes_per_sample(t->mMixerFormat));
+ ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
+ "buffer %p track %p, channels %d, needs %#x",
+ in, t, t->channelCount, t->needs);
+ return;
+ }
+
+ const size_t outFrames = b.frameCount;
+ if (ramp) {
+ volumeRampMulti<MIXTYPE_MULTI_SAVEONLY, NCHAN>(out, outFrames, in, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ } else {
+ volumeMulti<MIXTYPE_MULTI_SAVEONLY, NCHAN>(out, outFrames, in, aux,
+ t->volume, t->auxLevel);
+ }
+ out += outFrames * NCHAN;
+ if (aux != NULL) {
+ aux += NCHAN;
+ }
+ numFrames -= b.frameCount;
+
+ // release buffer
+ t->bufferProvider->releaseBuffer(&b);
+ }
+ if (ramp) {
+ t->adjustVolumeRamp(aux != NULL);
+ }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+ ALOGVV("track__Resample\n");
+ t->resampler->setSampleRate(t->sampleRate);
+
+ const bool ramp = t->needsRamp();
+ if (ramp || aux != NULL) {
+ // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
+ // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+ t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
+ memset(temp, 0, outFrameCount * NCHAN * sizeof(TO));
+ t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
+ if (ramp) {
+ volumeRampMulti<MIXTYPE_MULTI, NCHAN>(out, outFrameCount, temp, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ t->adjustVolumeRamp(aux != NULL);
+ } else {
+ volumeMulti<MIXTYPE_MULTI, NCHAN>(out, outFrameCount, temp, aux,
+ t->volume, t->auxLevel);
+ }
+ } else { // constant volume gain
+ t->resampler->setVolume(t->volume[0], t->volume[1]);
+ t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
+ }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in t->in.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux)
+{
+ ALOGVV("track__NoResample\n");
+ const TI *in = static_cast<const TI *>(t->in);
+
+ if (t->needsRamp()) {
+ volumeRampMulti<MIXTYPE, NCHAN>(out, frameCount, in, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ t->adjustVolumeRamp(aux != NULL);
+ } else {
+ volumeMulti<MIXTYPE, NCHAN>(out, frameCount, in, aux, t->volume, t->auxLevel);
+ }
+ // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+ // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+ in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * NCHAN;
+ t->in = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ // two int16_t are produced per iteration
+ ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+ if (!kUseNewMixer && channels == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return track__nop;
+ case TRACKTYPE_RESAMPLE:
+ return track__genericResample;
+ case TRACKTYPE_NORESAMPLEMONO:
+ return track__16BitsMono;
+ case TRACKTYPE_NORESAMPLE:
+ return track__16BitsStereo;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ }
+ LOG_ALWAYS_FATAL_IF(channels != FCC_2); // TODO: must be stereo right now
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return track__nop;
+ case TRACKTYPE_RESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__Resample<MIXTYPE_MULTI, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)\
+ track__Resample<MIXTYPE_MULTI, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLEMONO:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MONOEXPAND, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MONOEXPAND, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MULTI, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MULTI, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ */
+AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+ if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+ LOG_ALWAYS_FATAL("bad processType: %d", processType);
+ return NULL;
+ }
+ if (!kUseNewMixer && channels == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ return process__OneTrack16BitsStereoNoResampling;
+ }
+ LOG_ALWAYS_FATAL_IF(channels != FCC_2); // TODO: must be stereo right now
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ int16_t, float, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ float, int16_t, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ int16_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ return NULL;
+}
+
// ----------------------------------------------------------------------------
}; // namespace android
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 766ff60..e6de00c 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -221,6 +221,7 @@
// 16-byte boundary
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }
@@ -229,12 +230,14 @@
resampler->getUnreleasedFrames() : 0; };
};
+ typedef void (*process_hook_t)(state_t* state, int64_t pts);
+
// pad to 32-bytes to fill cache line
struct state_t {
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
- void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
+ process_hook_t hook; // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
NBLog::Writer* mLog;
@@ -345,6 +348,38 @@
static uint64_t sLocalTimeFreq;
static pthread_once_t sOnceControl;
static void sInitRoutine();
+
+ // multi-format process hooks
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void process_NoResampleOneTrack(state_t* state, int64_t pts);
+
+ // multi-format track hooks
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void track__Resample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK,
+ };
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // functions for determining the proper process and track hooks.
+ static process_hook_t getProcessHook(int processType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+ static hook_t getTrackHook(int trackType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixerOps.h b/services/audioflinger/AudioMixerOps.h
new file mode 100644
index 0000000..de92946
--- /dev/null
+++ b/services/audioflinger/AudioMixerOps.h
@@ -0,0 +1,361 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#ifndef ANDROID_AUDIO_MIXER_OPS_H
+#define ANDROID_AUDIO_MIXER_OPS_H
+
+namespace android {
+
+/* Behavior of is_same<>::value is true if the types are identical,
+ * false otherwise. Identical to the STL std::is_same.
+ */
+template<typename T, typename U>
+struct is_same
+{
+ static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T> // partial specialization
+{
+ static const bool value = true;
+};
+
+
+/* MixMul is a multiplication operator to scale an audio input signal
+ * by a volume gain, with the formula:
+ *
+ * O(utput) = I(nput) * V(olume)
+ *
+ * The output, input, and volume may have different types.
+ * There are 27 variants, of which 14 are actually defined in an
+ * explicitly templated class.
+ *
+ * The following type variables and the underlying meaning:
+ *
+ * Output type TO: int32_t (Q4.27) or int16_t (Q.15) or float [-1,1]
+ * Input signal type TI: int32_t (Q4.27) or int16_t (Q.15) or float [-1,1]
+ * Volume type TV: int32_t (U4.28) or int16_t (U4.12) or float [-1,1]
+ *
+ * For high precision audio, only the <TO, TI, TV> = <float, float, float>
+ * needs to be accelerated. This is perhaps the easiest form to do quickly as well.
+ */
+
+template <typename TO, typename TI, typename TV>
+inline TO MixMul(TI value, TV volume) {
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(false);
+ // should not be here :-).
+ // To avoid mistakes, this template is always specialized.
+ return value * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int16_t, int16_t>(int16_t value, int16_t volume) {
+ return value * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int32_t, int16_t>(int32_t value, int16_t volume) {
+ return (value >> 12) * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int16_t, int32_t>(int16_t value, int32_t volume) {
+ return value * (volume >> 16);
+}
+
+template <>
+inline int32_t MixMul<int32_t, int32_t, int32_t>(int32_t value, int32_t volume) {
+ return (value >> 12) * (volume >> 16);
+}
+
+template <>
+inline float MixMul<float, float, int16_t>(float value, int16_t volume) {
+ static const float norm = 1. / (1 << 12);
+ return value * volume * norm;
+}
+
+template <>
+inline float MixMul<float, float, int32_t>(float value, int32_t volume) {
+ static const float norm = 1. / (1 << 28);
+ return value * volume * norm;
+}
+
+template <>
+inline int16_t MixMul<int16_t, float, int16_t>(float value, int16_t volume) {
+ return clamp16_from_float(MixMul<float, float, int16_t>(value, volume));
+}
+
+template <>
+inline int16_t MixMul<int16_t, float, int32_t>(float value, int32_t volume) {
+ return clamp16_from_float(MixMul<float, float, int32_t>(value, volume));
+}
+
+template <>
+inline float MixMul<float, int16_t, int16_t>(int16_t value, int16_t volume) {
+ static const float norm = 1. / (1 << (15 + 12));
+ return static_cast<float>(value) * static_cast<float>(volume) * norm;
+}
+
+template <>
+inline float MixMul<float, int16_t, int32_t>(int16_t value, int32_t volume) {
+ static const float norm = 1. / (1ULL << (15 + 28));
+ return static_cast<float>(value) * static_cast<float>(volume) * norm;
+}
+
+template <>
+inline int16_t MixMul<int16_t, int16_t, int16_t>(int16_t value, int16_t volume) {
+ return clamp16(MixMul<int32_t, int16_t, int16_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int32_t, int16_t>(int32_t value, int16_t volume) {
+ return clamp16(MixMul<int32_t, int32_t, int16_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int16_t, int32_t>(int16_t value, int32_t volume) {
+ return clamp16(MixMul<int32_t, int16_t, int32_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int32_t, int32_t>(int32_t value, int32_t volume) {
+ return clamp16(MixMul<int32_t, int32_t, int32_t>(value, volume) >> 12);
+}
+
+/*
+ * MixAccum is used to add into an accumulator register of a possibly different
+ * type. The TO and TI types are the same as MixMul.
+ */
+
+template <typename TO, typename TI>
+inline void MixAccum(TO *auxaccum, TI value) {
+ if (!is_same<TO, TI>::value) {
+ LOG_ALWAYS_FATAL("MixAccum type not properly specialized: %d %d\n",
+ sizeof(TO), sizeof(TI));
+ }
+ *auxaccum += value;
+}
+
+template<>
+inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {
+ static const float norm = 1. / (1 << 15);
+ *auxaccum += norm * value;
+}
+
+template<>
+inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {
+ static const float norm = 1. / (1 << 27);
+ *auxaccum += norm * value;
+}
+
+template<>
+inline void MixAccum<int32_t, int16_t>(int32_t *auxaccum, int16_t value) {
+ *auxaccum += value << 12;
+}
+
+template<>
+inline void MixAccum<int32_t, float>(int32_t *auxaccum, float value) {
+ *auxaccum += clampq4_27_from_float(value);
+}
+
+/* MixMulAux is just like MixMul except it combines with
+ * an accumulator operation MixAccum.
+ */
+
+template <typename TO, typename TI, typename TV, typename TA>
+inline TO MixMulAux(TI value, TV volume, TA *auxaccum) {
+ MixAccum<TA, TI>(auxaccum, value);
+ return MixMul<TO, TI, TV>(value, volume);
+}
+
+/* MIXTYPE is used to determine how the samples in the input frame
+ * are mixed with volume gain into the output frame.
+ * See the volumeRampMulti functions below for more details.
+ */
+enum {
+ MIXTYPE_MULTI,
+ MIXTYPE_MONOEXPAND,
+ MIXTYPE_MULTI_SAVEONLY,
+};
+
+/*
+ * The volumeRampMulti and volumeRamp functions take a MIXTYPE
+ * which indicates the per-frame mixing and accumulation strategy.
+ *
+ * MIXTYPE_MULTI:
+ * NCHAN represents number of input and output channels.
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * vol: represents a volume array.
+ *
+ * This accumulates into the out pointer.
+ *
+ * MIXTYPE_MONOEXPAND:
+ * Single input channel. NCHAN represents number of output channels.
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * Input channel count is 1.
+ * vol: represents volume array.
+ *
+ * This accumulates into the out pointer.
+ *
+ * MIXTYPE_MULTI_SAVEONLY:
+ * NCHAN represents number of input and output channels.
+ * TO: int16_t (Q.15) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * vol: represents a volume array.
+ *
+ * MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.
+ */
+
+template <int MIXTYPE, int NCHAN,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+inline void volumeRampMulti(TO* out, size_t frameCount,
+ const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+#ifdef ALOGVV
+ ALOGVV("volumeRampMulti, MIXTYPE:%d\n", MIXTYPE);
+#endif
+ if (aux != NULL) {
+ do {
+ TA auxaccum = 0;
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ auxaccum /= NCHAN;
+ *aux++ += MixMul<TA, TA, TAV>(auxaccum, *vola);
+ vola[0] += volainc;
+ } while (--frameCount);
+ } else {
+ do {
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in++, vol[i]);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMul<TO, TI, TV>(*in++, vol[i]);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in, vol[i]);
+ vol[i] += volinc[i];
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ } while (--frameCount);
+ }
+}
+
+template <int MIXTYPE, int NCHAN,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+inline void volumeMulti(TO* out, size_t frameCount,
+ const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+#ifdef ALOGVV
+ ALOGVV("volumeMulti MIXTYPE:%d\n", MIXTYPE);
+#endif
+ if (aux != NULL) {
+ do {
+ TA auxaccum = 0;
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum);
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ auxaccum /= NCHAN;
+ *aux++ += MixMul<TA, TA, TAV>(auxaccum, vola);
+ } while (--frameCount);
+ } else {
+ do {
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in++, vol[i]);
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMul<TO, TI, TV>(*in++, vol[i]);
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in, vol[i]);
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ } while (--frameCount);
+ }
+}
+
+};
+
+#endif /* ANDROID_AUDIO_MIXER_OPS_H */