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-rw-r--r--media/libmediaplayerservice/nuplayer/GenericSource.cpp8
-rw-r--r--media/libstagefright/AwesomePlayer.cpp7
-rw-r--r--media/libstagefright/MPEG4Extractor.cpp131
-rw-r--r--media/libstagefright/include/MPEG4Extractor.h11
-rw-r--r--media/mtp/MtpDataPacket.cpp2
-rw-r--r--services/audioflinger/tests/Android.mk38
-rwxr-xr-xservices/audioflinger/tests/mixer_to_wav_tests.sh134
-rw-r--r--services/audioflinger/tests/test-mixer.cpp286
8 files changed, 600 insertions, 17 deletions
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index 06aac33..5cf9238 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -67,6 +67,14 @@ void NuPlayer::GenericSource::initFromDataSource(
CHECK(extractor != NULL);
+ sp<MetaData> fileMeta = extractor->getMetaData();
+ if (fileMeta != NULL) {
+ int64_t duration;
+ if (fileMeta->findInt64(kKeyDuration, &duration)) {
+ mDurationUs = duration;
+ }
+ }
+
for (size_t i = 0; i < extractor->countTracks(); ++i) {
sp<MetaData> meta = extractor->getTrackMetaData(i);
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index d679be1..f35a5b1 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -408,6 +408,13 @@ status_t AwesomePlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
totalBitRate += bitrate;
}
+ sp<MetaData> fileMeta = mExtractor->getMetaData();
+ if (fileMeta != NULL) {
+ int64_t duration;
+ if (fileMeta->findInt64(kKeyDuration, &duration)) {
+ mDurationUs = duration;
+ }
+ }
mBitrate = totalBitRate;
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 23b221d..1b9551f 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -51,6 +51,7 @@ public:
int32_t timeScale,
const sp<SampleTable> &sampleTable,
Vector<SidxEntry> &sidx,
+ const Trex *trex,
off64_t firstMoofOffset);
virtual status_t start(MetaData *params = NULL);
@@ -74,6 +75,7 @@ private:
uint32_t mCurrentSampleIndex;
uint32_t mCurrentFragmentIndex;
Vector<SidxEntry> &mSegments;
+ const Trex *mTrex;
off64_t mFirstMoofOffset;
off64_t mCurrentMoofOffset;
off64_t mNextMoofOffset;
@@ -141,6 +143,7 @@ private:
off64_t offset;
size_t size;
uint32_t duration;
+ int32_t compositionOffset;
uint8_t iv[16];
Vector<size_t> clearsizes;
Vector<size_t> encryptedsizes;
@@ -343,8 +346,7 @@ static bool AdjustChannelsAndRate(uint32_t fourcc, uint32_t *channels, uint32_t
}
MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source)
- : mSidxDuration(0),
- mMoofOffset(0),
+ : mMoofOffset(0),
mDataSource(source),
mInitCheck(NO_INIT),
mHasVideo(false),
@@ -1163,6 +1165,8 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
mLastTrack->timescale = ntohl(timescale);
+ // 14496-12 says all ones means indeterminate, but some files seem to use
+ // 0 instead. We treat both the same.
int64_t duration = 0;
if (version == 1) {
if (mDataSource->readAt(
@@ -1170,7 +1174,9 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
< (ssize_t)sizeof(duration)) {
return ERROR_IO;
}
- duration = ntoh64(duration);
+ if (duration != -1) {
+ duration = ntoh64(duration);
+ }
} else {
uint32_t duration32;
if (mDataSource->readAt(
@@ -1178,13 +1184,14 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
< (ssize_t)sizeof(duration32)) {
return ERROR_IO;
}
- // ffmpeg sets duration to -1, which is incorrect.
if (duration32 != 0xffffffff) {
duration = ntohl(duration32);
}
}
- mLastTrack->meta->setInt64(
- kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
+ if (duration != 0) {
+ mLastTrack->meta->setInt64(
+ kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
+ }
uint8_t lang[2];
off64_t lang_offset;
@@ -1724,11 +1731,11 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
{
*offset += chunk_size;
- if (chunk_data_size < 24) {
+ if (chunk_data_size < 32) {
return ERROR_MALFORMED;
}
- uint8_t header[24];
+ uint8_t header[32];
if (mDataSource->readAt(
data_offset, header, sizeof(header))
< (ssize_t)sizeof(header)) {
@@ -1736,14 +1743,27 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
}
uint64_t creationTime;
+ uint64_t duration = 0;
if (header[0] == 1) {
creationTime = U64_AT(&header[4]);
mHeaderTimescale = U32_AT(&header[20]);
+ duration = U64_AT(&header[24]);
+ if (duration == 0xffffffffffffffff) {
+ duration = 0;
+ }
} else if (header[0] != 0) {
return ERROR_MALFORMED;
} else {
creationTime = U32_AT(&header[4]);
mHeaderTimescale = U32_AT(&header[12]);
+ uint32_t d32 = U32_AT(&header[16]);
+ if (d32 == 0xffffffff) {
+ d32 = 0;
+ }
+ duration = d32;
+ }
+ if (duration != 0) {
+ mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
}
String8 s;
@@ -1754,6 +1774,50 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
break;
}
+ case FOURCC('m', 'e', 'h', 'd'):
+ {
+ *offset += chunk_size;
+
+ if (chunk_data_size < 8) {
+ return ERROR_MALFORMED;
+ }
+
+ uint8_t flags[4];
+ if (mDataSource->readAt(
+ data_offset, flags, sizeof(flags))
+ < (ssize_t)sizeof(flags)) {
+ return ERROR_IO;
+ }
+
+ uint64_t duration = 0;
+ if (flags[0] == 1) {
+ // 64 bit
+ if (chunk_data_size < 12) {
+ return ERROR_MALFORMED;
+ }
+ mDataSource->getUInt64(data_offset + 4, &duration);
+ if (duration == 0xffffffffffffffff) {
+ duration = 0;
+ }
+ } else if (flags[0] == 0) {
+ // 32 bit
+ uint32_t d32;
+ mDataSource->getUInt32(data_offset + 4, &d32);
+ if (d32 == 0xffffffff) {
+ d32 = 0;
+ }
+ duration = d32;
+ } else {
+ return ERROR_MALFORMED;
+ }
+
+ if (duration != 0) {
+ mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
+ }
+
+ break;
+ }
+
case FOURCC('m', 'd', 'a', 't'):
{
ALOGV("mdat chunk, drm: %d", mIsDrm);
@@ -1790,6 +1854,26 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
break;
}
+ case FOURCC('t', 'r', 'e', 'x'):
+ {
+ *offset += chunk_size;
+
+ if (chunk_data_size < 24) {
+ return ERROR_IO;
+ }
+ uint32_t duration;
+ Trex trex;
+ if (!mDataSource->getUInt32(data_offset + 4, &trex.track_ID) ||
+ !mDataSource->getUInt32(data_offset + 8, &trex.default_sample_description_index) ||
+ !mDataSource->getUInt32(data_offset + 12, &trex.default_sample_duration) ||
+ !mDataSource->getUInt32(data_offset + 16, &trex.default_sample_size) ||
+ !mDataSource->getUInt32(data_offset + 20, &trex.default_sample_flags)) {
+ return ERROR_IO;
+ }
+ mTrex.add(trex);
+ break;
+ }
+
case FOURCC('t', 'x', '3', 'g'):
{
uint32_t type;
@@ -2014,12 +2098,11 @@ status_t MPEG4Extractor::parseSegmentIndex(off64_t offset, size_t size) {
mSidxEntries.add(se);
}
- mSidxDuration = total_duration * 1000000 / timeScale;
- ALOGV("duration: %lld", mSidxDuration);
+ uint64_t sidxDuration = total_duration * 1000000 / timeScale;
int64_t metaDuration;
if (!mLastTrack->meta->findInt64(kKeyDuration, &metaDuration) || metaDuration == 0) {
- mLastTrack->meta->setInt64(kKeyDuration, mSidxDuration);
+ mLastTrack->meta->setInt64(kKeyDuration, sidxDuration);
}
return OK;
}
@@ -2492,11 +2575,24 @@ sp<MediaSource> MPEG4Extractor::getTrack(size_t index) {
return NULL;
}
+
+ Trex *trex = NULL;
+ int32_t trackId;
+ if (track->meta->findInt32(kKeyTrackID, &trackId)) {
+ for (size_t i = 0; i < mTrex.size(); i++) {
+ Trex *t = &mTrex.editItemAt(index);
+ if (t->track_ID == (uint32_t) trackId) {
+ trex = t;
+ break;
+ }
+ }
+ }
+
ALOGV("getTrack called, pssh: %d", mPssh.size());
return new MPEG4Source(
track->meta, mDataSource, track->timescale, track->sampleTable,
- mSidxEntries, mMoofOffset);
+ mSidxEntries, trex, mMoofOffset);
}
// static
@@ -2826,6 +2922,7 @@ MPEG4Source::MPEG4Source(
int32_t timeScale,
const sp<SampleTable> &sampleTable,
Vector<SidxEntry> &sidx,
+ const Trex *trex,
off64_t firstMoofOffset)
: mFormat(format),
mDataSource(dataSource),
@@ -2834,6 +2931,7 @@ MPEG4Source::MPEG4Source(
mCurrentSampleIndex(0),
mCurrentFragmentIndex(0),
mSegments(sidx),
+ mTrex(trex),
mFirstMoofOffset(firstMoofOffset),
mCurrentMoofOffset(firstMoofOffset),
mCurrentTime(0),
@@ -2850,6 +2948,8 @@ MPEG4Source::MPEG4Source(
mWantsNALFragments(false),
mSrcBuffer(NULL) {
+ memset(&mTrackFragmentHeaderInfo, 0, sizeof(mTrackFragmentHeaderInfo));
+
mFormat->findInt32(kKeyCryptoMode, &mCryptoMode);
mDefaultIVSize = 0;
mFormat->findInt32(kKeyCryptoDefaultIVSize, &mDefaultIVSize);
@@ -3421,8 +3521,8 @@ status_t MPEG4Source::parseTrackFragmentRun(off64_t offset, off64_t size) {
} else if (mTrackFragmentHeaderInfo.mFlags
& TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
- } else {
- sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
+ } else if (mTrex) {
+ sampleDuration = mTrex->default_sample_duration;
}
if (flags & kSampleSizePresent) {
@@ -3491,6 +3591,7 @@ status_t MPEG4Source::parseTrackFragmentRun(off64_t offset, off64_t size) {
tmp.offset = dataOffset;
tmp.size = sampleSize;
tmp.duration = sampleDuration;
+ tmp.compositionOffset = sampleCtsOffset;
mCurrentSamples.add(tmp);
dataOffset += sampleSize;
@@ -3893,7 +3994,7 @@ status_t MPEG4Source::fragmentedRead(
const Sample *smpl = &mCurrentSamples[mCurrentSampleIndex];
offset = smpl->offset;
size = smpl->size;
- cts = mCurrentTime;
+ cts = mCurrentTime + smpl->compositionOffset;
mCurrentTime += smpl->duration;
isSyncSample = (mCurrentSampleIndex == 0); // XXX
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 7b4bc6d..1fe6fcf 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -39,6 +39,14 @@ struct SidxEntry {
uint32_t mDurationUs;
};
+struct Trex {
+ uint32_t track_ID;
+ uint32_t default_sample_description_index;
+ uint32_t default_sample_duration;
+ uint32_t default_sample_size;
+ uint32_t default_sample_flags;
+};
+
class MPEG4Extractor : public MediaExtractor {
public:
// Extractor assumes ownership of "source".
@@ -74,11 +82,12 @@ private:
};
Vector<SidxEntry> mSidxEntries;
- uint64_t mSidxDuration;
off64_t mMoofOffset;
Vector<PsshInfo> mPssh;
+ Vector<Trex> mTrex;
+
sp<DataSource> mDataSource;
status_t mInitCheck;
bool mHasVideo;
diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp
index c4f87a0..e6e19e3 100644
--- a/media/mtp/MtpDataPacket.cpp
+++ b/media/mtp/MtpDataPacket.cpp
@@ -363,7 +363,7 @@ int MtpDataPacket::write(int fd) {
}
int MtpDataPacket::writeData(int fd, void* data, uint32_t length) {
- allocate(length);
+ allocate(length + MTP_CONTAINER_HEADER_SIZE);
memcpy(mBuffer + MTP_CONTAINER_HEADER_SIZE, data, length);
length += MTP_CONTAINER_HEADER_SIZE;
MtpPacket::putUInt32(MTP_CONTAINER_LENGTH_OFFSET, length);
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index f365637..7bba05b 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -33,3 +33,41 @@ LOCAL_MODULE := resampler_tests
LOCAL_MODULE_TAGS := tests
include $(BUILD_EXECUTABLE)
+
+#
+# audio mixer test tool
+#
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ test-mixer.cpp \
+ ../AudioMixer.cpp.arm \
+
+LOCAL_C_INCLUDES := \
+ bionic \
+ bionic/libstdc++/include \
+ external/stlport/stlport \
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/services/audioflinger
+
+LOCAL_STATIC_LIBRARIES := \
+ libsndfile
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libeffects \
+ libnbaio \
+ libcommon_time_client \
+ libaudioresampler \
+ libaudioutils \
+ libdl \
+ libcutils \
+ libutils \
+ liblog
+
+LOCAL_MODULE:= test-mixer
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_EXECUTABLE)
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
new file mode 100755
index 0000000..93bff47
--- /dev/null
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -0,0 +1,134 @@
+#!/bin/bash
+#
+# This script uses test-mixer to generate WAV files
+# for evaluation of the AudioMixer component.
+#
+# Sine and chirp signals are used for input because they
+# show up as clear lines, either horizontal or diagonal,
+# on a spectrogram. This means easy verification of multiple
+# track mixing.
+#
+# After execution, look for created subdirectories like
+# mixer_i_i
+# mixer_i_f
+# mixer_f_f
+#
+# Recommend using a program such as audacity to evaluate
+# the output WAV files, e.g.
+#
+# cd testdir
+# audacity *.wav
+#
+# Using Audacity:
+#
+# Under "Waveform" view mode you can zoom into the
+# start of the WAV file to verify proper ramping.
+#
+# Select "Spectrogram" to see verify the lines
+# (sine = horizontal, chirp = diagonal) which should
+# be clear (except for around the start as the volume
+# ramping causes spectral distortion).
+
+if [ -z "$ANDROID_BUILD_TOP" ]; then
+ echo "Android build environment not set"
+ exit -1
+fi
+
+# ensure we have mm
+. $ANDROID_BUILD_TOP/build/envsetup.sh
+
+pushd $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/
+
+# build
+pwd
+mm
+
+# send to device
+echo "waiting for device"
+adb root && adb wait-for-device remount
+adb push $OUT/system/lib/libaudioresampler.so /system/lib
+adb push $OUT/system/bin/test-mixer /system/bin
+
+# createwav creates a series of WAV files testing various
+# mixer settings
+# $1 = flags
+# $2 = directory
+function createwav() {
+# create directory if it doesn't exist
+ if [ ! -d $2 ]; then
+ mkdir $2
+ fi
+
+# Test:
+# process__genericResampling
+# track__Resample / track__genericResample
+ adb shell test-mixer $1 -s 48000 \
+ -o /sdcard/tm48000gr.wav \
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+ adb pull /sdcard/tm48000gr.wav $2
+
+# Test:
+# process__genericResample
+# track__Resample / track__genericResample
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+# Aux buffer
+ adb shell test-mixer $1 -s 9307 \
+ -a /sdcard/aux9307gra.wav -o /sdcard/tm9307gra.wav \
+ sine:2,1000,3000 sine:1,2000,9307 chirp:2,9307
+ adb pull /sdcard/tm9307gra.wav $2
+ adb pull /sdcard/aux9307gra.wav $2
+
+# Test:
+# process__genericNoResampling
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+ adb shell test-mixer $1 -s 32000 \
+ -o /sdcard/tm32000gnr.wav \
+ sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000
+ adb pull /sdcard/tm32000gnr.wav $2
+
+# Test:
+# process__genericNoResampling
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+# Aux buffer
+ adb shell test-mixer $1 -s 32000 \
+ -a /sdcard/aux32000gnra.wav -o /sdcard/tm32000gnra.wav \
+ sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000
+ adb pull /sdcard/tm32000gnra.wav $2
+ adb pull /sdcard/aux32000gnra.wav $2
+
+# Test:
+# process__NoResampleOneTrack / process__OneTrack16BitsStereoNoResampling
+# Downmixer
+ adb shell test-mixer $1 -s 32000 \
+ -o /sdcard/tm32000nrot.wav \
+ sine:6,1000,32000
+ adb pull /sdcard/tm32000nrot.wav $2
+
+# Test:
+# process__NoResampleOneTrack / OneTrack16BitsStereoNoResampling
+# Aux buffer
+ adb shell test-mixer $1 -s 44100 \
+ -a /sdcard/aux44100nrota.wav -o /sdcard/tm44100nrota.wav \
+ sine:2,2000,44100
+ adb pull /sdcard/tm44100nrota.wav $2
+ adb pull /sdcard/aux44100nrota.wav $2
+}
+
+#
+# Call createwav to generate WAV files in various combinations
+#
+# i_i = integer input track, integer mixer output
+# f_f = float input track, float mixer output
+# i_f = integer input track, float_mixer output
+#
+# If the mixer output is float, then the output WAV file is pcm float.
+#
+# TODO: create a "snr" like "diff" to automatically
+# compare files in these directories together.
+#
+
+createwav "" "tests/mixer_i_i"
+createwav "-f -m" "tests/mixer_f_f"
+createwav "-m" "tests/mixer_i_f"
+
+popd
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
new file mode 100644
index 0000000..3940702
--- /dev/null
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -0,0 +1,286 @@
+/*
+ * Copyright (C) 2014 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 <stdio.h>
+#include <inttypes.h>
+#include <math.h>
+#include <vector>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioMixer.h"
+#include "test_utils.h"
+
+/* Testing is typically through creation of an output WAV file from several
+ * source inputs, to be later analyzed by an audio program such as Audacity.
+ *
+ * Sine or chirp functions are typically more useful as input to the mixer
+ * as they show up as straight lines on a spectrogram if successfully mixed.
+ *
+ * A sample shell script is provided: mixer_to_wave_tests.sh
+ */
+
+using namespace android;
+
+static void usage(const char* name) {
+ fprintf(stderr, "Usage: %s [-f] [-m]"
+ " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
+ " (<input-file> | <command>)+\n", name);
+ fprintf(stderr, " -f enable floating point input track\n");
+ fprintf(stderr, " -m enable floating point mixer output\n");
+ fprintf(stderr, " -s mixer sample-rate\n");
+ fprintf(stderr, " -o <output-file> WAV file, pcm16 (or float if -m specified)\n");
+ fprintf(stderr, " -a <aux-buffer-file>\n");
+ fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n");
+ fprintf(stderr, " <input-file> is a WAV file\n");
+ fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
+ fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n");
+}
+
+static int writeFile(const char *filename, const void *buffer,
+ uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) {
+ if (filename == NULL) {
+ return 0; // ok to pass in NULL filename
+ }
+ // write output to file.
+ SF_INFO info;
+ info.frames = 0;
+ info.samplerate = sampleRate;
+ info.channels = channels;
+ info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16);
+ printf("saving file:%s channels:%d samplerate:%d frames:%d\n",
+ filename, info.channels, info.samplerate, frames);
+ SNDFILE *sf = sf_open(filename, SFM_WRITE, &info);
+ if (sf == NULL) {
+ perror(filename);
+ return EXIT_FAILURE;
+ }
+ if (isBufferFloat) {
+ (void) sf_writef_float(sf, (float*)buffer, frames);
+ } else {
+ (void) sf_writef_short(sf, (short*)buffer, frames);
+ }
+ sf_close(sf);
+ return EXIT_SUCCESS;
+}
+
+int main(int argc, char* argv[]) {
+ const char* const progname = argv[0];
+ bool useInputFloat = false;
+ bool useMixerFloat = false;
+ bool useRamp = true;
+ uint32_t outputSampleRate = 48000;
+ uint32_t outputChannels = 2; // stereo for now
+ std::vector<int> Pvalues;
+ const char* outputFilename = NULL;
+ const char* auxFilename = NULL;
+ std::vector<int32_t> Names;
+ std::vector<SignalProvider> Providers;
+
+ for (int ch; (ch = getopt(argc, argv, "fms:o:a:P:")) != -1;) {
+ switch (ch) {
+ case 'f':
+ useInputFloat = true;
+ break;
+ case 'm':
+ useMixerFloat = true;
+ break;
+ case 's':
+ outputSampleRate = atoi(optarg);
+ break;
+ case 'o':
+ outputFilename = optarg;
+ break;
+ case 'a':
+ auxFilename = optarg;
+ break;
+ case 'P':
+ if (parseCSV(optarg, Pvalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -P option\n");
+ return EXIT_FAILURE;
+ }
+ break;
+ case '?':
+ default:
+ usage(progname);
+ return EXIT_FAILURE;
+ }
+ }
+ argc -= optind;
+ argv += optind;
+
+ if (argc == 0) {
+ usage(progname);
+ return EXIT_FAILURE;
+ }
+ if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) {
+ fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS);
+ return EXIT_FAILURE;
+ }
+
+ size_t outputFrames = 0;
+
+ // create providers for each track
+ Providers.resize(argc);
+ for (int i = 0; i < argc; ++i) {
+ static const char chirp[] = "chirp:";
+ static const char sine[] = "sine:";
+ static const double kSeconds = 1;
+
+ if (!strncmp(argv[i], chirp, strlen(chirp))) {
+ std::vector<int> v;
+
+ parseCSV(argv[i] + strlen(chirp), v);
+ if (v.size() == 2) {
+ printf("creating chirp(%d %d)\n", v[0], v[1]);
+ if (useInputFloat) {
+ Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ } else {
+ Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ }
+ Providers[i].setIncr(Pvalues);
+ } else {
+ fprintf(stderr, "malformed input '%s'\n", argv[i]);
+ }
+ } else if (!strncmp(argv[i], sine, strlen(sine))) {
+ std::vector<int> v;
+
+ parseCSV(argv[i] + strlen(sine), v);
+ if (v.size() == 3) {
+ printf("creating sine(%d %d)\n", v[0], v[1]);
+ if (useInputFloat) {
+ Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ } else {
+ Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ }
+ Providers[i].setIncr(Pvalues);
+ } else {
+ fprintf(stderr, "malformed input '%s'\n", argv[i]);
+ }
+ } else {
+ printf("creating filename(%s)\n", argv[i]);
+ if (useInputFloat) {
+ Providers[i].setFile<float>(argv[i]);
+ } else {
+ Providers[i].setFile<short>(argv[i]);
+ }
+ Providers[i].setIncr(Pvalues);
+ }
+ // calculate the number of output frames
+ size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
+ / Providers[i].getSampleRate();
+ if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
+ outputFrames = nframes;
+ }
+ }
+
+ // create the output buffer.
+ const size_t outputFrameSize = outputChannels
+ * (useMixerFloat ? sizeof(float) : sizeof(int16_t));
+ const size_t outputSize = outputFrames * outputFrameSize;
+ void *outputAddr = NULL;
+ (void) posix_memalign(&outputAddr, 32, outputSize);
+ memset(outputAddr, 0, outputSize);
+
+ // create the aux buffer, if needed.
+ const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always
+ const size_t auxSize = outputFrames * auxFrameSize;
+ void *auxAddr = NULL;
+ if (auxFilename) {
+ (void) posix_memalign(&auxAddr, 32, auxSize);
+ memset(auxAddr, 0, auxSize);
+ }
+
+ // create the mixer.
+ const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
+ AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
+ audio_format_t inputFormat = useInputFloat
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ audio_format_t mixerFormat = useMixerFloat
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+ static float f0; // zero
+
+ // set up the tracks.
+ for (size_t i = 0; i < Providers.size(); ++i) {
+ //printf("track %d out of %d\n", i, Providers.size());
+ uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+ int32_t name = mixer->getTrackName(channelMask,
+ inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+ ALOG_ASSERT(name >= 0);
+ Names.push_back(name);
+ mixer->setBufferProvider(name, &Providers[i]);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ (void *) outputAddr);
+ mixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mixerFormat);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ (void *)(uintptr_t)inputFormat);
+ mixer->setParameter(
+ name,
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ (void *)(uintptr_t)Providers[i].getSampleRate());
+ if (useRamp) {
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f);
+ } else {
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
+ }
+ if (auxFilename) {
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ (void *) auxAddr);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f);
+ }
+ mixer->enable(name);
+ }
+
+ // pump the mixer to process data.
+ size_t i;
+ for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
+ for (size_t j = 0; j < Names.size(); ++j) {
+ mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ (char *) outputAddr + i * outputFrameSize);
+ if (auxFilename) {
+ mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ (char *) auxAddr + i * auxFrameSize);
+ }
+ }
+ mixer->process(AudioBufferProvider::kInvalidPTS);
+ }
+ outputFrames = i; // reset output frames to the data actually produced.
+
+ // write to files
+ writeFile(outputFilename, outputAddr,
+ outputSampleRate, outputChannels, outputFrames, useMixerFloat);
+ if (auxFilename) {
+ // Aux buffer is always in q4_27 format for now.
+ // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count)
+ ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1);
+ writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
+ }
+
+ delete mixer;
+ free(outputAddr);
+ free(auxAddr);
+ return EXIT_SUCCESS;
+}
generated by cgit v1.1 at 2025-01-08 00:29:03 +0000