Using centralized isAudioPlaybackRateValid to validate parameters
Centralized validation code
bug: 20701446
Change-Id: I9d9941c7639c05b2afe069ff4f858c693c910bfe
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 6cf2ca9..055f724 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -108,6 +108,23 @@
pr2.mFallbackMode == pr2.mFallbackMode;
}
+static inline bool isAudioPlaybackRateValid(const AudioPlaybackRate &playbackRate) {
+ if (playbackRate.mFallbackMode == AUDIO_TIMESTRETCH_FALLBACK_FAIL &&
+ (playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_SPEECH ||
+ playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_DEFAULT)) {
+ //test sonic specific constraints
+ return playbackRate.mSpeed >= TIMESTRETCH_SONIC_SPEED_MIN &&
+ playbackRate.mSpeed <= TIMESTRETCH_SONIC_SPEED_MAX &&
+ playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
+ playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
+ } else {
+ return playbackRate.mSpeed >= AUDIO_TIMESTRETCH_SPEED_MIN &&
+ playbackRate.mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX &&
+ playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
+ playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
+ }
+}
+
// TODO: Consider putting these inlines into a class scope
// Returns the source frames needed to resample to destination frames. This is not a precise
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 81ae6d7..2649ade 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -78,7 +78,7 @@
static inline float adjustSpeed(float speed, float pitch)
{
- return kFixPitch ? (speed / pitch) : speed;
+ return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
}
static inline float adjustPitch(float pitch)
@@ -809,25 +809,33 @@
const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
const float effectivePitch = adjustPitch(playbackRate.mPitch);
- if (effectiveSpeed < AUDIO_TIMESTRETCH_SPEED_MIN
- || effectiveSpeed > AUDIO_TIMESTRETCH_SPEED_MAX
- || effectivePitch < AUDIO_TIMESTRETCH_PITCH_MIN
- || effectivePitch > AUDIO_TIMESTRETCH_PITCH_MAX) {
+ AudioPlaybackRate playbackRateTemp = playbackRate;
+ playbackRateTemp.mSpeed = effectiveSpeed;
+ playbackRateTemp.mPitch = effectivePitch;
+
+ if (!isAudioPlaybackRateValid(playbackRateTemp)) {
return BAD_VALUE;
- //TODO: add function in AudioResamplerPublic.h to check for validity.
}
// Check if the buffer size is compatible.
if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
- mPlaybackRate = playbackRate;
- mProxy->setPlaybackRate(playbackRate);
- //modify this
- AudioPlaybackRate playbackRateTemp = playbackRate;
- playbackRateTemp.mSpeed = effectiveSpeed;
- playbackRateTemp.mPitch = effectivePitch;
+ // Check resampler ratios are within bounds
+ if (effectiveRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+ ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
+ playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+
+ if (effectiveRate * AUDIO_RESAMPLER_UP_RATIO_MAX < mSampleRate) {
+ ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value",
+ playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+ mPlaybackRate = playbackRate;
+ //set effective rates
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
return NO_ERROR;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 01efc53..8a9a837 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -733,13 +733,9 @@
case PLAYBACK_RATE: {
const AudioPlaybackRate *playbackRate =
reinterpret_cast<AudioPlaybackRate*>(value);
- ALOG_ASSERT(AUDIO_TIMESTRETCH_SPEED_MIN <= playbackRate->mSpeed
- && playbackRate->mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX,
- "bad speed %f", playbackRate->mSpeed);
- ALOG_ASSERT(AUDIO_TIMESTRETCH_PITCH_MIN <= playbackRate->mPitch
- && playbackRate->mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX,
- "bad pitch %f", playbackRate->mPitch);
- //TODO: use function from AudioResamplerPublic.h to test validity.
+ ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+ "bad parameters speed %f, pitch %f",playbackRate->mSpeed,
+ playbackRate->mPitch);
if (track.setPlaybackRate(*playbackRate)) {
ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
"%f %f %d %d",
diff --git a/services/audioflinger/BufferProviders.cpp b/services/audioflinger/BufferProviders.cpp
index 8a580e8..3566ee2 100644
--- a/services/audioflinger/BufferProviders.cpp
+++ b/services/audioflinger/BufferProviders.cpp
@@ -332,7 +332,8 @@
mLocalBufferData(NULL),
mRemaining(0),
mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
- mFallbackFailErrorShown(false)
+ mFallbackFailErrorShown(false),
+ mAudioPlaybackRateValid(false)
{
LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
"TimestretchBufferProvider can't allocate Sonic stream");
@@ -460,6 +461,8 @@
sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
//TODO: pitch is ignored for now
//TODO: optimize: if parameters are the same, don't do any extra computation.
+
+ mAudioPlaybackRateValid = isAudioPlaybackRateValid(mPlaybackRate);
return OK;
}
@@ -479,8 +482,7 @@
*srcFrames = targetSrc + 1;
}
- if (mPlaybackRate.mSpeed< TIMESTRETCH_SONIC_SPEED_MIN ||
- mPlaybackRate.mSpeed > TIMESTRETCH_SONIC_SPEED_MAX ) {
+ if (!mAudioPlaybackRateValid) {
//fallback mode
if (*dstFrames > 0) {
switch(mPlaybackRate.mFallbackMode) {
diff --git a/services/audioflinger/BufferProviders.h b/services/audioflinger/BufferProviders.h
index 4970b6c..4bc895c 100644
--- a/services/audioflinger/BufferProviders.h
+++ b/services/audioflinger/BufferProviders.h
@@ -188,6 +188,7 @@
sonicStream mSonicStream; // handle to sonic timestretch object
//FIXME: this dependency should be abstracted out
bool mFallbackFailErrorShown; // log fallback error only once
+ bool mAudioPlaybackRateValid; // flag for current parameters validity
};
// ----------------------------------------------------------------------------