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Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

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Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
This commit is contained in:
Eric Laurent
2009-07-07 07:10:45 -07:00
parent 37ea711225
commit 88e209dcf8
8 changed files with 36 additions and 60 deletions
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7
core/jni/android_media_AudioTrack.cpp
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@@ -539,16 +539,17 @@ static jint android_media_AudioTrack_get_native_frame_count(JNIEnv *env, jobjec
// ----------------------------------------------------------------------------
static void android_media_AudioTrack_set_playback_rate(JNIEnv *env, jobject thiz,
static jint android_media_AudioTrack_set_playback_rate(JNIEnv *env, jobject thiz,
jint sampleRateInHz) {
AudioTrack *lpTrack = (AudioTrack *)env->GetIntField(
thiz, javaAudioTrackFields.nativeTrackInJavaObj);
if (lpTrack) {
lpTrack->setSampleRate(sampleRateInHz);
return android_media_translateErrorCode(lpTrack->setSampleRate(sampleRateInHz));
} else {
jniThrowException(env, "java/lang/IllegalStateException",
"Unable to retrieve AudioTrack pointer for setSampleRate()");
return AUDIOTRACK_ERROR;
}
}
@@ -788,7 +789,7 @@ static JNINativeMethod gMethods[] = {
{"native_get_native_frame_count",
"()I", (void *)android_media_AudioTrack_get_native_frame_count},
{"native_set_playback_rate",
"(I)V", (void *)android_media_AudioTrack_set_playback_rate},
"(I)I", (void *)android_media_AudioTrack_set_playback_rate},
{"native_get_playback_rate",
"()I", (void *)android_media_AudioTrack_get_playback_rate},
{"native_set_marker_pos","(I)I", (void *)android_media_AudioTrack_set_marker_pos},

4
include/media/AudioRecord.h
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@@ -197,7 +197,6 @@ public:
/* getters, see constructor */
uint32_t sampleRate() const;
int format() const;
int channelCount() const;
uint32_t frameCount() const;
@@ -217,7 +216,7 @@ public:
status_t stop();
bool stopped() const;
/* get sample rate for this track
/* get sample rate for this record track
*/
uint32_t getSampleRate();
@@ -323,7 +322,6 @@ private:
sp<ClientRecordThread> mClientRecordThread;
Mutex mRecordThreadLock;
uint32_t mSampleRate;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;

3
include/media/AudioTrack.h
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@@ -201,7 +201,6 @@ public:
/* getters, see constructor */
int streamType() const;
uint32_t sampleRate() const;
int format() const;
int channelCount() const;
uint32_t frameCount() const;
@@ -246,7 +245,7 @@ public:
/* set sample rate for this track, mostly used for games' sound effects
*/
void setSampleRate(int sampleRate);
status_t setSampleRate(int sampleRate);
uint32_t getSampleRate();
/* Enables looping and sets the start and end points of looping.

7
include/private/media/AudioTrackShared.h
View File

@@ -26,7 +26,6 @@ namespace android {
// ----------------------------------------------------------------------------
#define MAX_SAMPLE_RATE 65535
#define THREAD_PRIORITY_AUDIO_CLIENT (ANDROID_PRIORITY_AUDIO)
// Maximum cumulated timeout milliseconds before restarting audioflinger thread
#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP init time
@@ -55,9 +54,9 @@ struct audio_track_cblk_t
uint16_t volume[2];
uint32_t volumeLR;
};
uint16_t sampleRate;
uint16_t channels;
int16_t flowControlFlag; // underrun (out) or overrrun (in) indication
uint32_t sampleRate;
uint8_t channels;
uint8_t flowControlFlag; // underrun (out) or overrrun (in) indication
uint8_t out; // out equals 1 for AudioTrack and 0 for AudioRecord
uint8_t forceReady;
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger

18
libs/audioflinger/AudioFlinger.cpp
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@@ -1281,7 +1281,7 @@ sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
status_t lStatus;
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
@@ -1596,8 +1596,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = (uint16_t)sampleRate;
mCblk->channels = (uint16_t)channelCount;
mCblk->sampleRate = sampleRate;
mCblk->channels = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -1620,8 +1620,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = (uint16_t)sampleRate;
mCblk->channels = (uint16_t)channelCount;
mCblk->sampleRate = sampleRate;
mCblk->channels = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
@@ -1682,7 +1682,7 @@ int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
}
int AudioFlinger::MixerThread::TrackBase::channelCount() const {
return mCblk->channels;
return (int)mCblk->channels;
}
void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -2267,12 +2267,6 @@ sp<IAudioRecord> AudioFlinger::openRecord(
goto Exit;
}
if (sampleRate > MAX_SAMPLE_RATE) {
LOGE("Sample rate out of range");
lStatus = BAD_VALUE;
goto Exit;
}
if (mAudioRecordThread == 0) {
LOGE("Audio record thread not started");
lStatus = NO_INIT;

21
media/java/android/media/AudioTrack.java
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@@ -425,8 +425,7 @@ public class AudioTrack
}
/**
* Returns the current playback rate in Hz. Note that this rate may differ from the one set
* with {@link #setPlaybackRate(int)} as the value effectively used is implementation-dependent.
* Returns the current playback rate in Hz.
*/
public int getPlaybackRate() {
return native_get_playback_rate();
@@ -651,18 +650,13 @@ public class AudioTrack
* Sets the playback sample rate for this track. This sets the sampling rate at which
* the audio data will be consumed and played back, not the original sampling rate of the
* content. Setting it to half the sample rate of the content will cause the playback to
* last twice as long, but will also result result in a negative pitch shift.
* The current implementation supports a maximum sample rate of 64kHz.
* Use {@link #getSampleRate()} to check the rate actually used in hardware after
* potential clamping.
* last twice as long, but will also result in a negative pitch shift.
* The valid sample rate range if from 1Hz to twice the value returned by
* {@link #getNativeOutputSampleRate(int)}.
* @param sampleRateInHz the sample rate expressed in Hz
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
// FIXME: the implementation should support twice the hardware output sample rate
// (see {@link #getNativeOutputSampleRate(int)}), but currently
// due to the representation of the sample rate in the native layer, the sample rate
// is limited to 65535Hz
public int setPlaybackRate(int sampleRateInHz) {
if (mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
@@ -670,8 +664,7 @@ public class AudioTrack
if (sampleRateInHz <= 0) {
return ERROR_BAD_VALUE;
}
native_set_playback_rate(sampleRateInHz);
return SUCCESS;
return native_set_playback_rate(sampleRateInHz);
}
@@ -1031,8 +1024,8 @@ public class AudioTrack
private native final void native_setVolume(float leftVolume, float rightVolume);
private native final void native_set_playback_rate(int sampleRateInHz);
private native final int native_get_playback_rate();
private native final int native_set_playback_rate(int sampleRateInHz);
private native final int native_get_playback_rate();
private native final int native_set_marker_pos(int marker);
private native final int native_get_marker_pos();

13
media/libmedia/AudioRecord.cpp
View File

@@ -185,7 +185,6 @@ status_t AudioRecord::set(
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->out = 0;
mSampleRate = sampleRate;
mFormat = format;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
@@ -196,7 +195,7 @@ status_t AudioRecord::set(
mRemainingFrames = notificationFrames;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / mSampleRate;
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
@@ -218,11 +217,6 @@ uint32_t AudioRecord::latency() const
return mLatency;
}
uint32_t AudioRecord::sampleRate() const
{
return mSampleRate;
}
int AudioRecord::format() const
{
return mFormat;
@@ -321,6 +315,11 @@ bool AudioRecord::stopped() const
return !mActive;
}
uint32_t AudioRecord::getSampleRate()
{
return mCblk->sampleRate;
}
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
if (mCbf == 0) return INVALID_OPERATION;

23
media/libmedia/AudioTrack.cpp
View File

@@ -243,7 +243,6 @@ status_t AudioTrack::set(
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSampleRate = sampleRate;
mStreamType = streamType;
mFormat = format;
mChannelCount = channelCount;
@@ -254,7 +253,7 @@ status_t AudioTrack::set(
mNotificationFrames = notificationFrames;
mRemainingFrames = notificationFrames;
mUserData = user;
mLatency = afLatency + (1000*mFrameCount) / mSampleRate;
mLatency = afLatency + (1000*mFrameCount) / sampleRate;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
@@ -281,11 +280,6 @@ int AudioTrack::streamType() const
return mStreamType;
}
uint32_t AudioTrack::sampleRate() const
{
return mSampleRate;
}
int AudioTrack::format() const
{
return mFormat;
@@ -438,24 +432,23 @@ void AudioTrack::getVolume(float* left, float* right)
*right = mVolume[RIGHT];
}
void AudioTrack::setSampleRate(int rate)
status_t AudioTrack::setSampleRate(int rate)
{
int afSamplingRate;
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return;
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (rate <= 0) rate = 1;
if (rate > afSamplingRate*2) rate = afSamplingRate*2;
if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE;
if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
mCblk->sampleRate = (uint16_t)rate;
mCblk->sampleRate = rate;
return NO_ERROR;
}
uint32_t AudioTrack::getSampleRate()
{
return uint32_t(mCblk->sampleRate);
return mCblk->sampleRate;
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -866,7 +859,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
result.append(buffer);
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
result.append(buffer);