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-rw-r--r--services/audioflinger/AudioFlinger.cpp56
-rw-r--r--services/audioflinger/AudioResampler.cpp23
-rw-r--r--services/audioflinger/AudioResampler.h12
-rw-r--r--services/audioflinger/AudioResamplerCubic.cpp23
-rw-r--r--services/audioflinger/AudioResamplerCubic.h6
-rw-r--r--services/audioflinger/AudioResamplerDyn.cpp7
-rw-r--r--services/audioflinger/AudioResamplerDyn.h6
-rw-r--r--services/audioflinger/AudioResamplerSinc.cpp14
-rw-r--r--services/audioflinger/AudioResamplerSinc.h4
-rw-r--r--services/audioflinger/RecordTracks.h21
-rw-r--r--services/audioflinger/Threads.cpp402
-rw-r--r--services/audioflinger/Threads.h114
-rw-r--r--services/audioflinger/Tracks.cpp45
-rw-r--r--services/audioflinger/tests/resampler_tests.cpp5
14 files changed, 507 insertions, 231 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f3206cb..5002099 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -45,6 +45,8 @@
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
+#include <media/AudioResamplerPublic.h>
+
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
@@ -1140,19 +1142,46 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form
if (ret != NO_ERROR) {
return 0;
}
+ if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
+ return 0;
+ }
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
- audio_config_t config;
- memset(&config, 0, sizeof(config));
- config.sample_rate = sampleRate;
- config.channel_mask = channelMask;
- config.format = format;
+ audio_config_t config, proposed;
+ memset(&proposed, 0, sizeof(proposed));
+ proposed.sample_rate = sampleRate;
+ proposed.channel_mask = channelMask;
+ proposed.format = format;
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
- size_t size = dev->get_input_buffer_size(dev, &config);
+ size_t frames;
+ for (;;) {
+ // Note: config is currently a const parameter for get_input_buffer_size()
+ // but we use a copy from proposed in case config changes from the call.
+ config = proposed;
+ frames = dev->get_input_buffer_size(dev, &config);
+ if (frames != 0) {
+ break; // hal success, config is the result
+ }
+ // change one parameter of the configuration each iteration to a more "common" value
+ // to see if the device will support it.
+ if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
+ proposed.format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
+ proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw?
+ } else {
+ ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
+ "format %#x, channelMask 0x%X",
+ sampleRate, format, channelMask);
+ break; // retries failed, break out of loop with frames == 0.
+ }
+ }
mHardwareStatus = AUDIO_HW_IDLE;
- return size;
+ if (frames > 0 && config.sample_rate != sampleRate) {
+ frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
+ }
+ return frames; // may be converted to bytes at the Java level.
}
uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
@@ -1419,9 +1448,8 @@ sp<IAudioRecord> AudioFlinger::openRecord(
goto Exit;
}
- // we don't yet support anything other than 16-bit PCM
- if (!(audio_is_valid_format(format) &&
- audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
+ // we don't yet support anything other than linear PCM
+ if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("openRecord() invalid format %#x", format);
lStatus = BAD_VALUE;
goto Exit;
@@ -2002,11 +2030,11 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m
status, address.string());
// If the input could not be opened with the requested parameters and we can handle the
- // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
- // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
+ // conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
- config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT &&
- (halconfig.sample_rate <= 2 * config->sample_rate) &&
+ audio_is_linear_pcm(config->format) &&
+ audio_is_linear_pcm(halconfig.format) &&
+ (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 46e3d6c..e49b7b1 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -41,7 +41,7 @@ public:
AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
}
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
// number of bits used in interpolation multiply - 15 bits avoids overflow
@@ -51,9 +51,9 @@ private:
static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
void init() {}
- void resampleMono16(int32_t* out, size_t outFrameCount,
+ size_t resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
- void resampleStereo16(int32_t* out, size_t outFrameCount,
+ size_t resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
@@ -329,7 +329,7 @@ void AudioResampler::reset() {
// ----------------------------------------------------------------------------
-void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
// should never happen, but we overflow if it does
@@ -338,15 +338,16 @@ void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
// select the appropriate resampler
switch (mChannelCount) {
case 1:
- resampleMono16(out, outFrameCount, provider);
- break;
+ return resampleMono16(out, outFrameCount, provider);
case 2:
- resampleStereo16(out, outFrameCount, provider);
- break;
+ return resampleStereo16(out, outFrameCount, provider);
+ default:
+ LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+ return 0;
}
}
-void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -442,9 +443,10 @@ resampleStereo16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex / 2 /* channels for stereo */;
}
-void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -538,6 +540,7 @@ resampleMono16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex;
}
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 863614a..a8e3e6f 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -67,12 +67,18 @@ public:
// Resample int16_t samples from provider and accumulate into 'out'.
// A mono provider delivers a sequence of samples.
// A stereo provider delivers a sequence of interleaved pairs of samples.
- // Multi-channel providers are not supported.
+ //
// In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
// That is, for a mono provider, there is an implicit up-channeling.
// Since this method accumulates, the caller is responsible for clearing 'out' initially.
- // FIXME assumes provider is always successful; it should return the actual frame count.
- virtual void resample(int32_t* out, size_t outFrameCount,
+ //
+ // For a float resampler, 'out' holds interleaved pairs of float samples.
+ //
+ // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
+ // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
+ //
+ // Returns the number of frames resampled into the out buffer.
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
virtual void reset();
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index d3cbd1c..172c2a5 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_TAG "AudioSRC"
+#define LOG_TAG "AudioResamplerCubic"
#include <stdint.h>
#include <string.h>
@@ -32,7 +32,7 @@ void AudioResamplerCubic::init() {
memset(&right, 0, sizeof(state));
}
-void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
// should never happen, but we overflow if it does
@@ -41,15 +41,16 @@ void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
// select the appropriate resampler
switch (mChannelCount) {
case 1:
- resampleMono16(out, outFrameCount, provider);
- break;
+ return resampleMono16(out, outFrameCount, provider);
case 2:
- resampleStereo16(out, outFrameCount, provider);
- break;
+ return resampleStereo16(out, outFrameCount, provider);
+ default:
+ LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+ return 0;
}
}
-void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -67,7 +68,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL) {
- return;
+ return 0;
}
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
}
@@ -115,9 +116,10 @@ save_state:
// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex / 2 /* channels for stereo */;
}
-void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -135,7 +137,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL) {
- return;
+ return 0;
}
// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
@@ -182,6 +184,7 @@ save_state:
// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex;
}
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 1ddc5f9..4b45b0b 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -31,7 +31,7 @@ public:
AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
}
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
// number of bits used in interpolation multiply - 14 bits avoids overflow
@@ -43,9 +43,9 @@ private:
int32_t a, b, c, y0, y1, y2, y3;
} state;
void init();
- void resampleMono16(int32_t* out, size_t outFrameCount,
+ size_t resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
- void resampleStereo16(int32_t* out, size_t outFrameCount,
+ size_t resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
static inline int32_t interp(state* p, int32_t x) {
return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1;
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index c21d4ca..6481b85 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -477,15 +477,15 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate)
}
template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
- (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
+ return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
}
template<typename TC, typename TI, typename TO>
template<int CHANNELS, bool LOCKED, int STRIDE>
-void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
// TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
@@ -610,6 +610,7 @@ resample_exit:
ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
mInBuffer.setImpulse(impulse);
mPhaseFraction = phaseFraction;
+ return outputIndex / OUTPUT_CHANNELS;
}
/* instantiate templates used by AudioResampler::create */
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 238b163..3b1c381 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -52,7 +52,7 @@ public:
virtual void setVolume(float left, float right);
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
@@ -111,10 +111,10 @@ private:
int inSampleRate, int outSampleRate, double tbwCheat);
template<int CHANNELS, bool LOCKED, int STRIDE>
- void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+ size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
// define a pointer to member function type for resample
- typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+ typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
size_t outFrameCount, AudioBufferProvider* provider);
// data - the contiguous storage and layout of these is important.
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index ba9a356..41730ee 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -256,7 +256,7 @@ void AudioResamplerSinc::setVolume(float left, float right) {
mVolumeSIMD[1] = u4_28_from_float(clampFloatVol(right));
}
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
// FIXME store current state (up or down sample) and only load the coefs when the state
@@ -272,17 +272,18 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
// select the appropriate resampler
switch (mChannelCount) {
case 1:
- resample<1>(out, outFrameCount, provider);
- break;
+ return resample<1>(out, outFrameCount, provider);
case 2:
- resample<2>(out, outFrameCount, provider);
- break;
+ return resample<2>(out, outFrameCount, provider);
+ default:
+ LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+ return 0;
}
}
template<int CHANNELS>
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
const Constants& c(*mConstants);
@@ -357,6 +358,7 @@ resample_exit:
mImpulse = impulse;
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex / CHANNELS;
}
template<int CHANNELS>
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 6d8e85d..0fbeac8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -39,7 +39,7 @@ public:
virtual ~AudioResamplerSinc();
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
void init();
@@ -47,7 +47,7 @@ private:
virtual void setVolume(float left, float right);
template<int CHANNELS>
- void resample(int32_t* out, size_t outFrameCount,
+ size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
template<int CHANNELS>
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 204a9d6..25d6d95 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -34,6 +34,7 @@ public:
IAudioFlinger::track_flags_t flags,
track_type type);
virtual ~RecordTrack();
+ virtual status_t initCheck() const;
virtual status_t start(AudioSystem::sync_event_t event, int triggerSession);
virtual void stop();
@@ -66,21 +67,6 @@ private:
bool mOverflow; // overflow on most recent attempt to fill client buffer
- // updated by RecordThread::readInputParameters_l()
- AudioResampler *mResampler;
-
- // interleaved stereo pairs of fixed-point Q4.27
- int32_t *mRsmpOutBuffer;
- // current allocated frame count for the above, which may be larger than needed
- size_t mRsmpOutFrameCount;
-
- size_t mRsmpInUnrel; // unreleased frames remaining from
- // most recent getNextBuffer
- // for debug only
-
- // rolling counter that is never cleared
- int32_t mRsmpInFront; // next available frame
-
AudioBufferProvider::Buffer mSink; // references client's buffer sink in shared memory
// sync event triggering actual audio capture. Frames read before this event will
@@ -93,7 +79,10 @@ private:
ssize_t mFramesToDrop;
// used by resampler to find source frames
- ResamplerBufferProvider *mResamplerBufferProvider;
+ ResamplerBufferProvider *mResamplerBufferProvider;
+
+ // used by the record thread to convert frames to proper destination format
+ RecordBufferConverter *mRecordBufferConverter;
};
// playback track, used by PatchPanel
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 4efb3d7..1a20fae 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -86,7 +86,13 @@
#define ALOGVV(a...) do { } while(0)
#endif
+// TODO: Move these macro/inlines to a header file.
#define max(a, b) ((a) > (b) ? (a) : (b))
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+ return a < b ? a : b;
+}
namespace android {
@@ -5290,7 +5296,6 @@ failed: ;
// FIXME mNormalSource
}
-
AudioFlinger::RecordThread::~RecordThread()
{
if (mFastCapture != 0) {
@@ -5594,6 +5599,9 @@ reacquire_wakelock:
continue;
}
+ // TODO: This code probably should be moved to RecordTrack.
+ // TODO: Update the activeTrack buffer converter in case of reconfigure.
+
enum {
OVERRUN_UNKNOWN,
OVERRUN_TRUE,
@@ -5608,131 +5616,28 @@ reacquire_wakelock:
size_t framesOut = activeTrack->mSink.frameCount;
LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
- int32_t front = activeTrack->mRsmpInFront;
- ssize_t filled = rear - front;
+ // check available frames and handle overrun conditions
+ // if the record track isn't draining fast enough.
+ bool hasOverrun;
size_t framesIn;
-
- if (filled < 0) {
- // should not happen, but treat like a massive overrun and re-sync
- framesIn = 0;
- activeTrack->mRsmpInFront = rear;
- overrun = OVERRUN_TRUE;
- } else if ((size_t) filled <= mRsmpInFrames) {
- framesIn = (size_t) filled;
- } else {
- // client is not keeping up with server, but give it latest data
- framesIn = mRsmpInFrames;
- activeTrack->mRsmpInFront = front = rear - framesIn;
+ activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
+ if (hasOverrun) {
overrun = OVERRUN_TRUE;
}
-
if (framesOut == 0 || framesIn == 0) {
break;
}
- if (activeTrack->mResampler == NULL) {
- // no resampling
- if (framesIn > framesOut) {
- framesIn = framesOut;
- } else {
- framesOut = framesIn;
- }
- int8_t *dst = activeTrack->mSink.i8;
- while (framesIn > 0) {
- front &= mRsmpInFramesP2 - 1;
- size_t part1 = mRsmpInFramesP2 - front;
- if (part1 > framesIn) {
- part1 = framesIn;
- }
- int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
- if (mChannelCount == activeTrack->mChannelCount) {
- memcpy(dst, src, part1 * mFrameSize);
- } else if (mChannelCount == 1) {
- upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
- part1);
- } else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
- (const int16_t *)src, part1);
- }
- dst += part1 * activeTrack->mFrameSize;
- front += part1;
- framesIn -= part1;
- }
- activeTrack->mRsmpInFront += framesOut;
-
- } else {
- // resampling
- // FIXME framesInNeeded should really be part of resampler API, and should
- // depend on the SRC ratio
- // to keep mRsmpInBuffer full so resampler always has sufficient input
- size_t framesInNeeded;
- // FIXME only re-calculate when it changes, and optimize for common ratios
- // Do not precompute in/out because floating point is not associative
- // e.g. a*b/c != a*(b/c).
- const double in(mSampleRate);
- const double out(activeTrack->mSampleRate);
- framesInNeeded = ceil(framesOut * in / out) + 1;
- ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g",
- framesInNeeded, framesOut, in / out);
- // Although we theoretically have framesIn in circular buffer, some of those are
- // unreleased frames, and thus must be discounted for purpose of budgeting.
- size_t unreleased = activeTrack->mRsmpInUnrel;
- framesIn = framesIn > unreleased ? framesIn - unreleased : 0;
- if (framesIn < framesInNeeded) {
- ALOGV("not enough to resample: have %u frames in but need %u in to "
- "produce %u out given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0;
- LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut);
- if (newFramesOut == 0) {
- break;
- }
- framesInNeeded = ceil(newFramesOut * in / out) + 1;
- ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g",
- framesInNeeded, newFramesOut, out / in);
- LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded);
- ALOGV("success 2: have %u frames in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, newFramesOut, in / out);
- framesOut = newFramesOut;
- } else {
- ALOGV("success 1: have %u in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- }
-
- // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
- if (activeTrack->mRsmpOutFrameCount < framesOut) {
- // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
- delete[] activeTrack->mRsmpOutBuffer;
- // resampler always outputs stereo
- activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2];
- activeTrack->mRsmpOutFrameCount = framesOut;
- }
-
- // resampler accumulates, but we only have one source track
- memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
- activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut,
- // FIXME how about having activeTrack implement this interface itself?
- activeTrack->mResamplerBufferProvider
- /*this*/ /* AudioBufferProvider* */);
- // ditherAndClamp() works as long as all buffers returned by
- // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
- if (activeTrack->mChannelCount == 1) {
- // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t
- ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
- framesOut);
- // the resampler always outputs stereo samples:
- // do post stereo to mono conversion
- downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16,
- (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut);
- } else {
- ditherAndClamp((int32_t *)activeTrack->mSink.raw,
- activeTrack->mRsmpOutBuffer, framesOut);
- }
- // now done with mRsmpOutBuffer
-
- }
+ // Don't allow framesOut to be larger than what is possible with resampling
+ // from framesIn.
+ // This isn't strictly necessary but helps limit buffer resizing in
+ // RecordBufferConverter. TODO: remove when no longer needed.
+ framesOut = min(framesOut,
+ destinationFramesPossible(
+ framesIn, mSampleRate, activeTrack->mSampleRate));
+ // process frames from the RecordThread buffer provider to the RecordTrack buffer
+ framesOut = activeTrack->mRecordBufferConverter->convert(
+ activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
overrun = OVERRUN_FALSE;
@@ -6041,12 +5946,9 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac
// was initialized to some value closer to the thread's mRsmpInFront, then the track could
// see previously buffered data before it called start(), but with greater risk of overrun.
- recordTrack->mRsmpInFront = mRsmpInRear;
- recordTrack->mRsmpInUnrel = 0;
- // FIXME why reset?
- if (recordTrack->mResampler != NULL) {
- recordTrack->mResampler->reset();
- }
+ recordTrack->mResamplerBufferProvider->reset();
+ // clear any converter state as new data will be discontinuous
+ recordTrack->mRecordBufferConverter->reset();
recordTrack->mState = TrackBase::STARTING_2;
// signal thread to start
mWaitWorkCV.broadcast();
@@ -6222,12 +6124,52 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args
write(fd, result.string(), result.size());
}
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
+{
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+ RecordThread *recordThread = (RecordThread *) threadBase.get();
+ mRsmpInFront = recordThread->mRsmpInRear;
+ mRsmpInUnrel = 0;
+}
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
+ size_t *framesAvailable, bool *hasOverrun)
+{
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+ RecordThread *recordThread = (RecordThread *) threadBase.get();
+ const int32_t rear = recordThread->mRsmpInRear;
+ const int32_t front = mRsmpInFront;
+ const ssize_t filled = rear - front;
+
+ size_t framesIn;
+ bool overrun = false;
+ if (filled < 0) {
+ // should not happen, but treat like a massive overrun and re-sync
+ framesIn = 0;
+ mRsmpInFront = rear;
+ overrun = true;
+ } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
+ framesIn = (size_t) filled;
+ } else {
+ // client is not keeping up with server, but give it latest data
+ framesIn = recordThread->mRsmpInFrames;
+ mRsmpInFront = /* front = */ rear - framesIn;
+ overrun = true;
+ }
+ if (framesAvailable != NULL) {
+ *framesAvailable = framesIn;
+ }
+ if (hasOverrun != NULL) {
+ *hasOverrun = overrun;
+ }
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
- RecordTrack *activeTrack = mRecordTrack;
- sp<ThreadBase> threadBase = activeTrack->mThread.promote();
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
if (threadBase == 0) {
buffer->frameCount = 0;
buffer->raw = NULL;
@@ -6235,7 +6177,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
}
RecordThread *recordThread = (RecordThread *) threadBase.get();
int32_t rear = recordThread->mRsmpInRear;
- int32_t front = activeTrack->mRsmpInFront;
+ int32_t front = mRsmpInFront;
ssize_t filled = rear - front;
// FIXME should not be P2 (don't want to increase latency)
// FIXME if client not keeping up, discard
@@ -6252,17 +6194,16 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
part1 = ask;
}
if (part1 == 0) {
- // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
- LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved");
+ // out of data is fine since the resampler will return a short-count.
buffer->raw = NULL;
buffer->frameCount = 0;
- activeTrack->mRsmpInUnrel = 0;
+ mRsmpInUnrel = 0;
return NOT_ENOUGH_DATA;
}
buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
buffer->frameCount = part1;
- activeTrack->mRsmpInUnrel = part1;
+ mRsmpInUnrel = part1;
return NO_ERROR;
}
@@ -6270,18 +6211,197 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
AudioBufferProvider::Buffer* buffer)
{
- RecordTrack *activeTrack = mRecordTrack;
size_t stepCount = buffer->frameCount;
if (stepCount == 0) {
return;
}
- ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel);
- activeTrack->mRsmpInUnrel -= stepCount;
- activeTrack->mRsmpInFront += stepCount;
+ ALOG_ASSERT(stepCount <= mRsmpInUnrel);
+ mRsmpInUnrel -= stepCount;
+ mRsmpInFront += stepCount;
buffer->raw = NULL;
buffer->frameCount = 0;
}
+AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate) :
+ mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
+ // mSrcFormat
+ // mSrcSampleRate
+ // mDstChannelMask
+ // mDstFormat
+ // mDstSampleRate
+ // mSrcChannelCount
+ // mDstChannelCount
+ // mDstFrameSize
+ mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
+ mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0)
+{
+ (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
+ dstChannelMask, dstFormat, dstSampleRate);
+}
+
+AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
+ free(mBuf);
+ delete mResampler;
+ free(mRsmpOutBuffer);
+}
+
+size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
+ AudioBufferProvider *provider, size_t frames)
+{
+ if (mSrcSampleRate == mDstSampleRate) {
+ ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+ mSrcSampleRate, mSrcFormat, mDstFormat);
+
+ AudioBufferProvider::Buffer buffer;
+ for (size_t i = frames; i > 0; ) {
+ buffer.frameCount = i;
+ status_t status = provider->getNextBuffer(&buffer, 0);
+ if (status != OK || buffer.frameCount == 0) {
+ frames -= i; // cannot fill request.
+ break;
+ }
+ // convert to destination buffer
+ convert(dst, buffer.raw, buffer.frameCount);
+
+ dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
+ i -= buffer.frameCount;
+ provider->releaseBuffer(&buffer);
+ }
+ } else {
+ ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+ mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
+
+ // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
+ if (mRsmpOutFrameCount < frames) {
+ // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
+ free(mRsmpOutBuffer);
+ // resampler always outputs stereo (FOR NOW)
+ (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/);
+ mRsmpOutFrameCount = frames;
+ }
+ // resampler accumulates, but we only have one source track
+ memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t));
+ frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider);
+
+ // convert to destination buffer
+ convert(dst, mRsmpOutBuffer, frames);
+ }
+ return frames;
+}
+
+status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate)
+{
+ // quick evaluation if there is any change.
+ if (mSrcFormat == srcFormat
+ && mSrcChannelMask == srcChannelMask
+ && mSrcSampleRate == srcSampleRate
+ && mDstFormat == dstFormat
+ && mDstChannelMask == dstChannelMask
+ && mDstSampleRate == dstSampleRate) {
+ return NO_ERROR;
+ }
+
+ const bool valid =
+ audio_is_input_channel(srcChannelMask)
+ && audio_is_input_channel(dstChannelMask)
+ && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
+ && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
+ && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
+ ; // no upsampling checks for now
+ if (!valid) {
+ return BAD_VALUE;
+ }
+
+ mSrcFormat = srcFormat;
+ mSrcChannelMask = srcChannelMask;
+ mSrcSampleRate = srcSampleRate;
+ mDstFormat = dstFormat;
+ mDstChannelMask = dstChannelMask;
+ mDstSampleRate = dstSampleRate;
+
+ // compute derived parameters
+ mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
+ mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
+ mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
+
+ // do we need a format buffer?
+ if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) {
+ mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
+ } else {
+ mBufFrameSize = 0;
+ }
+ mBufFrames = 0; // force the buffer to be resized.
+
+ // do we need to resample?
+ if (mSrcSampleRate != mDstSampleRate) {
+ if (mResampler != NULL) {
+ delete mResampler;
+ }
+ mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+ mSrcChannelCount, mDstSampleRate); // may seem confusing...
+ mResampler->setSampleRate(mSrcSampleRate);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
+ }
+ return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::RecordBufferConverter::convert(
+ void *dst, /*const*/ void *src, size_t frames)
+{
+ // check if a memcpy will do
+ if (mResampler == NULL
+ && mSrcChannelCount == mDstChannelCount
+ && mSrcFormat == mDstFormat) {
+ memcpy(dst, src,
+ frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat));
+ return;
+ }
+ // reallocate buffer if needed
+ if (mBufFrameSize != 0 && mBufFrames < frames) {
+ free(mBuf);
+ mBufFrames = frames;
+ (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
+ }
+ // do processing
+ if (mResampler != NULL) {
+ // src channel count is always >= 2.
+ void *dstBuf = mBuf != NULL ? mBuf : dst;
+ // ditherAndClamp() works as long as all buffers returned by
+ // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
+ if (mDstChannelCount == 1) {
+ // the resampler always outputs stereo samples.
+ // FIXME: this rewrites back into src
+ ditherAndClamp((int32_t *)src, (const int32_t *)src, frames);
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+ (const int16_t *)src, frames);
+ } else {
+ ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames);
+ }
+ } else if (mSrcChannelCount != mDstChannelCount) {
+ void *dstBuf = mBuf != NULL ? mBuf : dst;
+ if (mSrcChannelCount == 1) {
+ upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src,
+ frames);
+ } else {
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+ (const int16_t *)src, frames);
+ }
+ }
+ if (mSrcFormat != mDstFormat) {
+ void *srcBuf = mBuf != NULL ? mBuf : src;
+ memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat,
+ frames * mDstChannelCount);
+ }
+}
+
bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
@@ -6303,7 +6423,7 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+ if (!audio_is_linear_pcm((audio_format_t) value)) {
status = BAD_VALUE;
} else {
reqFormat = (audio_format_t) value;
@@ -6377,10 +6497,10 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
}
if (reconfig) {
if (status == BAD_VALUE &&
- reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
- reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+ audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
+ audio_is_linear_pcm(reqFormat) &&
(mInput->stream->common.get_sample_rate(&mInput->stream->common)
- <= (2 * samplingRate)) &&
+ <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
audio_channel_count_from_in_mask(
mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
(channelMask == AUDIO_CHANNEL_IN_MONO ||
@@ -6451,6 +6571,8 @@ void AudioFlinger::RecordThread::readInputParameters_l()
// The value is somewhat arbitrary, and could probably be even larger.
// A larger value should allow more old data to be read after a track calls start(),
// without increasing latency.
+ //
+ // Note this is independent of the maximum downsampling ratio permitted for capture.
mRsmpInFrames = mFrameCount * 7;
mRsmpInFramesP2 = roundup(mRsmpInFrames);
delete[] mRsmpInBuffer;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index d600ea9..27bc56b 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1036,17 +1036,127 @@ class RecordThread : public ThreadBase
public:
class RecordTrack;
+
+ /* The ResamplerBufferProvider is used to retrieve recorded input data from the
+ * RecordThread. It maintains local state on the relative position of the read
+ * position of the RecordTrack compared with the RecordThread.
+ */
class ResamplerBufferProvider : public AudioBufferProvider
- // derives from AudioBufferProvider interface for use by resampler
{
public:
- ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { }
+ ResamplerBufferProvider(RecordTrack* recordTrack) :
+ mRecordTrack(recordTrack),
+ mRsmpInUnrel(0), mRsmpInFront(0) { }
virtual ~ResamplerBufferProvider() { }
+
+ // called to set the ResamplerBufferProvider to head of the RecordThread data buffer,
+ // skipping any previous data read from the hal.
+ virtual void reset();
+
+ /* Synchronizes RecordTrack position with the RecordThread.
+ * Calculates available frames and handle overruns if the RecordThread
+ * has advanced faster than the ResamplerBufferProvider has retrieved data.
+ * TODO: why not do this for every getNextBuffer?
+ *
+ * Parameters
+ * framesAvailable: pointer to optional output size_t to store record track
+ * frames available.
+ * hasOverrun: pointer to optional boolean, returns true if track has overrun.
+ */
+
+ virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL);
+
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
private:
RecordTrack * const mRecordTrack;
+ size_t mRsmpInUnrel; // unreleased frames remaining from
+ // most recent getNextBuffer
+ // for debug only
+ int32_t mRsmpInFront; // next available frame
+ // rolling counter that is never cleared
+ };
+
+ /* The RecordBufferConverter is used for format, channel, and sample rate
+ * conversion for a RecordTrack.
+ *
+ * TODO: Self contained, so move to a separate file later.
+ *
+ * RecordBufferConverter uses the convert() method rather than exposing a
+ * buffer provider interface; this is to save a memory copy.
+ */
+ class RecordBufferConverter
+ {
+ public:
+ RecordBufferConverter(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate);
+
+ ~RecordBufferConverter();
+
+ /* Converts input data from an AudioBufferProvider by format, channelMask,
+ * and sampleRate to a destination buffer.
+ *
+ * Parameters
+ * dst: buffer to place the converted data.
+ * provider: buffer provider to obtain source data.
+ * frames: number of frames to convert
+ *
+ * Returns the number of frames converted.
+ */
+ size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
+
+ // returns NO_ERROR if constructor was successful
+ status_t initCheck() const {
+ // mSrcChannelMask set on successful updateParameters
+ return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
+ }
+
+ // allows dynamic reconfigure of all parameters
+ status_t updateParameters(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate);
+
+ // called to reset resampler buffers on record track discontinuity
+ void reset() {
+ if (mResampler != NULL) {
+ mResampler->reset();
+ }
+ }
+
+ private:
+ // internal convert function for format and channel mask.
+ void convert(void *dst, /*const*/ void *src, size_t frames);
+
+ // user provided information
+ audio_channel_mask_t mSrcChannelMask;
+ audio_format_t mSrcFormat;
+ uint32_t mSrcSampleRate;
+ audio_channel_mask_t mDstChannelMask;
+ audio_format_t mDstFormat;
+ uint32_t mDstSampleRate;
+
+ // derived information
+ uint32_t mSrcChannelCount;
+ uint32_t mDstChannelCount;
+ size_t mDstFrameSize;
+
+ // format conversion buffer
+ void *mBuf;
+ size_t mBufFrames;
+ size_t mBufFrameSize;
+
+ // resampler info
+ AudioResampler *mResampler;
+ // interleaved stereo pairs of fixed-point Q4.27 or float depending on resampler
+ void *mRsmpOutBuffer;
+ // current allocated frame count for the above, which may be larger than needed
+ size_t mRsmpOutFrameCount;
};
#include "RecordTracks.h"
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 5625661..1566b1f 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1990,29 +1990,30 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
type),
- mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
- // See real initialization of mRsmpInFront at RecordThread::start()
- mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
+ mOverflow(false),
+ mFramesToDrop(0)
{
if (mCblk == NULL) {
return;
}
+ mRecordBufferConverter = new RecordBufferConverter(
+ thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+ channelMask, format, sampleRate);
+ // Check if the RecordBufferConverter construction was successful.
+ // If not, don't continue with construction.
+ //
+ // NOTE: It would be extremely rare that the record track cannot be created
+ // for the current device, but a pending or future device change would make
+ // the record track configuration valid.
+ if (mRecordBufferConverter->initCheck() != NO_ERROR) {
+ ALOGE("RecordTrack unable to create record buffer converter");
+ return;
+ }
+
mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack());
-
- uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
- // FIXME I don't understand either of the channel count checks
- if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
- channelCount <= FCC_2) {
- // sink SR
- mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
- thread->mChannelCount, sampleRate);
- // source SR
- mResampler->setSampleRate(thread->mSampleRate);
- mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
- mResamplerBufferProvider = new ResamplerBufferProvider(this);
- }
+ mResamplerBufferProvider = new ResamplerBufferProvider(this);
if (flags & IAudioFlinger::TRACK_FAST) {
ALOG_ASSERT(thread->mFastTrackAvail);
@@ -2023,11 +2024,19 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
ALOGV("%s", __func__);
- delete mResampler;
- delete[] mRsmpOutBuffer;
+ delete mRecordBufferConverter;
delete mResamplerBufferProvider;
}
+status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
+{
+ status_t status = TrackBase::initCheck();
+ if (status == NO_ERROR && mServerProxy == 0) {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts __unused)
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index d6217ba..9e375db 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -48,7 +48,10 @@ void resample(int channels, void *output,
if (thisFrames == 0 || thisFrames > outputFrames - i) {
thisFrames = outputFrames - i;
}
- resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
+ size_t framesResampled = resampler->resample(
+ (int32_t*) output + channels*i, thisFrames, provider);
+ // we should have enough buffer space, so there is no short count.
+ ASSERT_EQ(thisFrames, framesResampled);
i += thisFrames;
}
}
generated by cgit v1.1 at 2024-05-15 18:34:32 +0000