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멀티미디어 예제

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ioacademy-jikim
2015-08-04 19:14:13 +09:00
commit d3b375295e
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02_day/test-resample/Android.mk
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LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
test-resample.cpp \
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils)
LOCAL_STATIC_LIBRARIES := \
libsndfile
LOCAL_SHARED_LIBRARIES := \
libaudioresampler \
libaudioutils \
libdl \
libcutils \
libutils \
liblog
LOCAL_MODULE:= test-resample
LOCAL_MODULE_TAGS := optional
include $(BUILD_EXECUTABLE)
02_day/test-resample/AudioResampler.h
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/*
* Copyright (C) 2007 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_RESAMPLER_H
#define ANDROID_AUDIO_RESAMPLER_H
#include <stdint.h>
#include <sys/types.h>
#include <cutils/compiler.h>
#include <media/AudioBufferProvider.h>
#include <system/audio.h>
namespace android {
// ----------------------------------------------------------------------------
class ANDROID_API AudioResampler {
public:
// Determines quality of SRC.
// LOW_QUALITY: linear interpolator (1st order)
// MED_QUALITY: cubic interpolator (3rd order)
// HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
// NOTE: high quality SRC will only be supported for
// certain fixed rate conversions. Sample rate cannot be
// changed dynamically.
enum src_quality {
DEFAULT_QUALITY=0,
LOW_QUALITY=1,
MED_QUALITY=2,
HIGH_QUALITY=3,
VERY_HIGH_QUALITY=4,
DYN_LOW_QUALITY=5,
DYN_MED_QUALITY=6,
DYN_HIGH_QUALITY=7,
};
static const float UNITY_GAIN_FLOAT = 1.0f;
static AudioResampler* create(audio_format_t format, int inChannelCount,
int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
virtual ~AudioResampler();
virtual void init() = 0;
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(float left, float right);
virtual void setLocalTimeFreq(uint64_t freq);
// set the PTS of the next buffer output by the resampler
virtual void setPTS(int64_t pts);
// 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,
AudioBufferProvider* provider) = 0;
virtual void reset();
virtual size_t getUnreleasedFrames() const { return mInputIndex; }
// called from destructor, so must not be virtual
src_quality getQuality() const { return mQuality; }
protected:
// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
static const int kNumPhaseBits = 30;
// phase mask for fraction
static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
// multiplier to calculate fixed point phase increment
static const double kPhaseMultiplier;
AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
// prevent copying
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
int64_t calculateOutputPTS(int outputFrameIndex);
const int32_t mChannelCount;
const int32_t mSampleRate;
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
int16_t mVolume[2];
uint32_t mVolumeRL;
};
int16_t mTargetVolume[2];
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
uint64_t mLocalTimeFreq;
int64_t mPTS;
// returns the inFrameCount required to generate outFrameCount frames.
//
// Placed here to be a consistent for all resamplers.
//
// Right now, we use the upper bound without regards to the current state of the
// input buffer using integer arithmetic, as follows:
//
// (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
//
// The double precision equivalent (float may not be precise enough):
// ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
//
// this relies on the fact that the mPhaseIncrement is rounded down from
// #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
// http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
//
// (so long as double precision is computed accurately enough to be considered
// greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
// will not necessarily hold for floats).
//
// TODO:
// Greater accuracy and a tight bound is obtained by:
// 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
// 2) using the exact integer formula where (ignoring 64b casting)
// inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
// phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
//
inline size_t getInFrameCountRequired(size_t outFrameCount) {
return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
+ (mSampleRate - 1))/mSampleRate;
}
inline float clampFloatVol(float volume) {
if (volume > UNITY_GAIN_FLOAT) {
return UNITY_GAIN_FLOAT;
} else if (volume >= 0.) {
return volume;
}
return 0.; // NaN or negative volume maps to 0.
}
private:
const src_quality mQuality;
// Return 'true' if the quality level is supported without explicit request
static bool qualityIsSupported(src_quality quality);
// For pthread_once()
static void init_routine();
// Return the estimated CPU load for specific resampler in MHz.
// The absolute number is irrelevant, it's the relative values that matter.
static uint32_t qualityMHz(src_quality quality);
};
// ----------------------------------------------------------------------------
}
; // namespace android
#endif // ANDROID_AUDIO_RESAMPLER_H
02_day/test-resample/test-resample.cpp
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/*
* Copyright (C) 2012 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.
*/
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <string.h>
#include <sys/mman.h>
#include <sys/stat.h>
#include <errno.h>
#include <inttypes.h>
#include <time.h>
#include <math.h>
#include <audio_utils/primitives.h>
#include <audio_utils/sndfile.h>
#include <utils/Vector.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
using namespace android;
static bool gVerbose = false;
static int usage(const char* name) {
fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-v] [-c channels]"
" [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
" [-i input-sample-rate] [-o output-sample-rate]"
" [-O csv] [-P csv] [<input-file>]"
" <output-file>\n", name);
fprintf(stderr," -p enable profiling\n");
fprintf(stderr," -f enable filter profiling\n");
fprintf(stderr," -F enable floating point -q {dlq|dmq|dhq} only");
fprintf(stderr," -v verbose : log buffer provider calls\n");
fprintf(stderr," -c # channels (1-2 for lq|mq|hq; 1-8 for dlq|dmq|dhq)\n");
fprintf(stderr," -q resampler quality\n");
fprintf(stderr," dq : default quality\n");
fprintf(stderr," lq : low quality\n");
fprintf(stderr," mq : medium quality\n");
fprintf(stderr," hq : high quality\n");
fprintf(stderr," vhq : very high quality\n");
fprintf(stderr," dlq : dynamic low quality\n");
fprintf(stderr," dmq : dynamic medium quality\n");
fprintf(stderr," dhq : dynamic high quality\n");
fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n");
fprintf(stderr," -o output file sample rate\n");
fprintf(stderr," -O # frames output per call to resample() in CSV format\n");
fprintf(stderr," -P # frames provided per call to resample() in CSV format\n");
return -1;
}
// Convert a list of integers in CSV format to a Vector of those values.
// Returns the number of elements in the list, or -1 on error.
int parseCSV(const char *string, Vector<int>& values)
{
// pass 1: count the number of values and do syntax check
size_t numValues = 0;
bool hadDigit = false;
for (const char *p = string; ; ) {
switch (*p++) {
case '0': case '1': case '2': case '3': case '4':
case '5': case '6': case '7': case '8': case '9':
hadDigit = true;
break;
case '\0':
if (hadDigit) {
// pass 2: allocate and initialize vector of values
values.resize(++numValues);
values.editItemAt(0) = atoi(p = optarg);
for (size_t i = 1; i < numValues; ) {
if (*p++ == ',') {
values.editItemAt(i++) = atoi(p);
}
}
return numValues;
}
// fall through
case ',':
if (hadDigit) {
hadDigit = false;
numValues++;
break;
}
// fall through
default:
return -1;
}
}
}
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
bool profileResample = false;
bool profileFilter = false;
bool useFloat = false;
int channels = 1;
int input_freq = 0;
int output_freq = 0;
AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
Vector<int> Ovalues;
Vector<int> Pvalues;
int ch;
while ((ch = getopt(argc, argv, "pfFvc:q:i:o:O:P:")) != -1) {
switch (ch) {
case 'p':
profileResample = true;
break;
case 'f':
profileFilter = true;
break;
case 'F':
useFloat = true;
break;
case 'v':
gVerbose = true;
break;
case 'c':
channels = atoi(optarg);
break;
case 'q':
if (!strcmp(optarg, "dq"))
quality = AudioResampler::DEFAULT_QUALITY;
else if (!strcmp(optarg, "lq"))
quality = AudioResampler::LOW_QUALITY;
else if (!strcmp(optarg, "mq"))
quality = AudioResampler::MED_QUALITY;
else if (!strcmp(optarg, "hq"))
quality = AudioResampler::HIGH_QUALITY;
else if (!strcmp(optarg, "vhq"))
quality = AudioResampler::VERY_HIGH_QUALITY;
else if (!strcmp(optarg, "dlq"))
quality = AudioResampler::DYN_LOW_QUALITY;
else if (!strcmp(optarg, "dmq"))
quality = AudioResampler::DYN_MED_QUALITY;
else if (!strcmp(optarg, "dhq"))
quality = AudioResampler::DYN_HIGH_QUALITY;
else {
usage(progname);
return -1;
}
break;
case 'i':
input_freq = atoi(optarg);
break;
case 'o':
output_freq = atoi(optarg);
break;
case 'O':
if (parseCSV(optarg, Ovalues) < 0) {
fprintf(stderr, "incorrect syntax for -O option\n");
return -1;
}
break;
case 'P':
if (parseCSV(optarg, Pvalues) < 0) {
fprintf(stderr, "incorrect syntax for -P option\n");
return -1;
}
break;
case '?':
default:
usage(progname);
return -1;
}
}
if (channels < 1
|| channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) {
fprintf(stderr, "invalid number of audio channels %d\n", channels);
return -1;
}
if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) {
fprintf(stderr, "float processing is only possible for dynamic resamplers\n");
return -1;
}
argc -= optind;
argv += optind;
const char* file_in = NULL;
const char* file_out = NULL;
if (argc == 1) {
file_out = argv[0];
} else if (argc == 2) {
file_in = argv[0];
file_out = argv[1];
} else {
usage(progname);
return -1;
}
// ----------------------------------------------------------
size_t input_size;
void* input_vaddr;
if (argc == 2) {
SF_INFO info;
info.format = 0;
SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
if (sf == NULL) {
perror(file_in);
return EXIT_FAILURE;
}
input_size = info.frames * info.channels * sizeof(short);
input_vaddr = malloc(input_size);
(void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
sf_close(sf);
channels = info.channels;
input_freq = info.samplerate;
} else {
// data for testing is exactly (input sampling rate/1000)/2 seconds
// so 44.1khz input is 22.05 seconds
double k = 1000; // Hz / s
double time = (input_freq / 2) / k;
size_t input_frames = size_t(input_freq * time);
input_size = channels * sizeof(int16_t) * input_frames;
input_vaddr = malloc(input_size);
int16_t* in = (int16_t*)input_vaddr;
for (size_t i=0 ; i<input_frames ; i++) {
double t = double(i) / input_freq;
double y = sin(M_PI * k * t * t);
int16_t yi = floor(y * 32767.0 + 0.5);
for (int j = 0; j < channels; j++) {
in[i*channels + j] = yi / (1 + j);
}
}
}
size_t input_framesize = channels * sizeof(int16_t);
size_t input_frames = input_size / input_framesize;
// For float processing, convert input int16_t to float array
if (useFloat) {
void *new_vaddr;
input_framesize = channels * sizeof(float);
input_size = input_frames * input_framesize;
new_vaddr = malloc(input_size);
memcpy_to_float_from_i16(reinterpret_cast<float*>(new_vaddr),
reinterpret_cast<int16_t*>(input_vaddr), input_frames * channels);
free(input_vaddr);
input_vaddr = new_vaddr;
}
// ----------------------------------------------------------
class Provider: public AudioBufferProvider {
const void* mAddr; // base address
const size_t mNumFrames; // total frames
const size_t mFrameSize; // size of each frame in bytes
size_t mNextFrame; // index of next frame to provide
size_t mUnrel; // number of frames not yet released
const Vector<int> mPvalues; // number of frames provided per call
size_t mNextPidx; // index of next entry in mPvalues to use
public:
Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues)
: mAddr(addr),
mNumFrames(frames),
mFrameSize(frameSize),
mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
}
virtual status_t getNextBuffer(Buffer* buffer,
int64_t pts = kInvalidPTS) {
(void)pts; // suppress warning
size_t requestedFrames = buffer->frameCount;
if (requestedFrames > mNumFrames - mNextFrame) {
buffer->frameCount = mNumFrames - mNextFrame;
}
if (!mPvalues.isEmpty()) {
size_t provided = mPvalues[mNextPidx++];
printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount);
if (provided < buffer->frameCount) {
buffer->frameCount = provided;
}
if (mNextPidx >= mPvalues.size()) {
mNextPidx = 0;
}
}
if (gVerbose) {
printf("getNextBuffer() requested %zu frames out of %zu frames available,"
" and returned %zu frames\n",
requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount);
}
mUnrel = buffer->frameCount;
if (buffer->frameCount > 0) {
buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
return NO_ERROR;
} else {
buffer->raw = NULL;
return NOT_ENOUGH_DATA;
}
}
virtual void releaseBuffer(Buffer* buffer) {
if (buffer->frameCount > mUnrel) {
fprintf(stderr, "ERROR releaseBuffer() released %zu frames but only %zu available "
"to release\n", buffer->frameCount, mUnrel);
mNextFrame += mUnrel;
mUnrel = 0;
} else {
if (gVerbose) {
printf("releaseBuffer() released %zu frames out of %zu frames available "
"to release\n", buffer->frameCount, mUnrel);
}
mNextFrame += buffer->frameCount;
mUnrel -= buffer->frameCount;
}
buffer->frameCount = 0;
buffer->raw = NULL;
}
void reset() {
mNextFrame = 0;
}
} provider(input_vaddr, input_frames, input_framesize, Pvalues);
if (gVerbose) {
printf("%zu input frames\n", input_frames);
}
audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples
size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t));
size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq;
size_t output_size = output_frames * output_framesize;
if (profileFilter) {
// Check how fast sample rate changes are that require filter changes.
// The delta sample rate changes must indicate a downsampling ratio,
// and must be larger than 10% changes.
//
// On fast devices, filters should be generated between 0.1ms - 1ms.
// (single threaded).
AudioResampler* resampler = AudioResampler::create(format, channels,
8000, quality);
int looplimit = 100;
timespec start, end;
clock_gettime(CLOCK_MONOTONIC, &start);
for (int i = 0; i < looplimit; ++i) {
resampler->setSampleRate(9000);
resampler->setSampleRate(12000);
resampler->setSampleRate(20000);
resampler->setSampleRate(30000);
}
clock_gettime(CLOCK_MONOTONIC, &end);
int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
int64_t time = end_ns - start_ns;
printf("%.2f sample rate changes with filter calculation/sec\n",
looplimit * 4 / (time / 1e9));
// Check how fast sample rate changes are without filter changes.
// This should be very fast, probably 0.1us - 1us per sample rate
// change.
resampler->setSampleRate(1000);
looplimit = 1000;
clock_gettime(CLOCK_MONOTONIC, &start);
for (int i = 0; i < looplimit; ++i) {
resampler->setSampleRate(1000+i);
}
clock_gettime(CLOCK_MONOTONIC, &end);
start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
time = end_ns - start_ns;
printf("%.2f sample rate changes without filter calculation/sec\n",
looplimit / (time / 1e9));
resampler->reset();
delete resampler;
}
void* output_vaddr = malloc(output_size);
AudioResampler* resampler = AudioResampler::create(format, channels,
output_freq, quality);
resampler->setSampleRate(input_freq);
resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
if (profileResample) {
/*
* For profiling on mobile devices, upon experimentation
* it is better to run a few trials with a shorter loop limit,
* and take the minimum time.
*
* Long tests can cause CPU temperature to build up and thermal throttling
* to reduce CPU frequency.
*
* For frequency checks (index=0, or 1, etc.):
* "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
*
* For temperature checks (index=0, or 1, etc.):
* "cat /sys/class/thermal/thermal_zone${index}/temp"
*
* Another way to avoid thermal throttling is to fix the CPU frequency
* at a lower level which prevents excessive temperatures.
*/
const int trials = 4;
const int looplimit = 4;
timespec start, end;
int64_t time = 0;
for (int n = 0; n < trials; ++n) {
clock_gettime(CLOCK_MONOTONIC, &start);
for (int i = 0; i < looplimit; ++i) {
resampler->resample((int*) output_vaddr, output_frames, &provider);
provider.reset(); // during benchmarking reset only the provider
}
clock_gettime(CLOCK_MONOTONIC, &end);
int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
int64_t diff_ns = end_ns - start_ns;
if (n == 0 || diff_ns < time) {
time = diff_ns; // save the best out of our trials.
}
}
// Mfrms/s is "Millions of output frames per second".
printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n",
quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
resampler->reset();
}
memset(output_vaddr, 0, output_size);
if (gVerbose) {
printf("resample() %zu output frames\n", output_frames);
}
if (Ovalues.isEmpty()) {
Ovalues.push(output_frames);
}
for (size_t i = 0, j = 0; i < output_frames; ) {
size_t thisFrames = Ovalues[j++];
if (j >= Ovalues.size()) {
j = 0;
}
if (thisFrames == 0 || thisFrames > output_frames - i) {
thisFrames = output_frames - i;
}
resampler->resample((int*) output_vaddr + output_channels*i, thisFrames, &provider);
i += thisFrames;
}
if (gVerbose) {
printf("resample() complete\n");
}
resampler->reset();
if (gVerbose) {
printf("reset() complete\n");
}
delete resampler;
resampler = NULL;
// For float processing, convert output format from float to Q4.27,
// which is then converted to int16_t for final storage.
if (useFloat) {
memcpy_to_q4_27_from_float(reinterpret_cast<int32_t*>(output_vaddr),
reinterpret_cast<float*>(output_vaddr), output_frames * output_channels);
}
// mono takes left channel only (out of stereo output pair)
// stereo and multichannel preserve all channels.
int32_t* out = (int32_t*) output_vaddr;
int16_t* convert = (int16_t*) malloc(output_frames * channels * sizeof(int16_t));
const int volumeShift = 12; // shift requirement for Q4.27 to Q.15
// round to half towards zero and saturate at int16 (non-dithered)
const int roundVal = (1<<(volumeShift-1)) - 1; // volumePrecision > 0
for (size_t i = 0; i < output_frames; i++) {
for (int j = 0; j < channels; j++) {
int32_t s = out[i * output_channels + j] + roundVal; // add offset here
if (s < 0) {
s = (s + 1) >> volumeShift; // round to 0
if (s < -32768) {
s = -32768;
}
} else {
s = s >> volumeShift;
if (s > 32767) {
s = 32767;
}
}
convert[i * channels + j] = int16_t(s);
}
}
// write output to disk
SF_INFO info;
info.frames = 0;
info.samplerate = output_freq;
info.channels = channels;
info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
if (sf == NULL) {
perror(file_out);
return EXIT_FAILURE;
}
(void) sf_writef_short(sf, convert, output_frames);
sf_close(sf);
return EXIT_SUCCESS;
}