Add Track Sink Format to AudioMixer
Track Sink Format now allows both AUDIO_FORMAT_PCM_FLOAT and
AUDIO_FORMAT_PCM_16_BIT (default). The float case is not
enabled now.
Change-Id: Ibee70cab4725c8bc9905e49f1f9a9c2448f76e00
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 67d83b1..f00b82a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -193,6 +193,7 @@
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t->downmixerBufferProvider = NULL;
+ t->mSinkFormat = AUDIO_FORMAT_PCM_16_BIT;
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
if (status == OK) {
@@ -440,6 +441,13 @@
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
break */
+ case SINK_FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track.mSinkFormat != format) {
+ track.mSinkFormat = format;
+ ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);
+ }
+ } break;
default:
LOG_FATAL("bad param");
}
@@ -1043,7 +1051,7 @@
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
- size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
+ size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
while (e0) {
// process by group of tracks with same output buffer to
// avoid multiple memset() on same buffer
@@ -1062,7 +1070,8 @@
}
e0 &= ~(e1);
- memset(t1.mainBuffer, 0, bufSize);
+ memset(t1.mainBuffer, 0, sampleCount
+ * audio_bytes_per_sample(t1.mSinkFormat));
}
while (e1) {
@@ -1170,8 +1179,18 @@
}
}
}
- ditherAndClamp(out, outTemp, BLOCKSIZE);
- out += BLOCKSIZE;
+ switch (t1.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q19_12(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 sink format: %d", t1.mSinkFormat);
+ }
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
@@ -1253,7 +1272,16 @@
}
}
}
- ditherAndClamp(out, outTemp, numFrames);
+ switch (t1.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ ditherAndClamp(out, outTemp, numFrames);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
+ }
}
}
@@ -1294,27 +1322,45 @@
}
size_t outFrames = b.frameCount;
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
- // volume is boosted, so we might need to clamp even though
- // we process only one track.
+ switch (t.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT: {
+ float *fout = reinterpret_cast<float*>(out);
+ static float scale = 1. / (32768. * 4096.); // exact when inverted
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- // clamping...
- l = clamp16(l);
- r = clamp16(r);
- *out++ = (r<<16) | (l & 0xFFFF);
+ int32_t l = mulRL(1, rl, vrl);
+ int32_t r = mulRL(0, rl, vrl);
+ *fout++ = static_cast<float>(l) * scale;
+ *fout++ = static_cast<float>(r) * scale;
} while (--outFrames);
- } else {
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
+ } break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ // volume is boosted, so we might need to clamp even though
+ // we process only one track.
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ // clamping...
+ l = clamp16(l);
+ r = clamp16(r);
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ } else {
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);
}
numFrames -= b.frameCount;
t.bufferProvider->releaseBuffer(&b);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index d286986..3355db4 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -77,6 +77,7 @@
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
DOWNMIX_TYPE = 0X4004,
+ SINK_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
// for target RESAMPLE
SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
// parameter 'value' is the new sample rate in Hz.
@@ -193,7 +194,9 @@
int32_t sessionId;
- int32_t padding[2];
+ audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+
+ int32_t padding[1];
// 16-byte boundary