Add test-mixer AudioMixer test program

The shell script mixer_to_wav_tests.sh shows how to use
test-mixer.

Change-Id: Ia7f1a368972c9c33fadc96df4cb1fc8b22446c8c

3 files changed, 458 insertions, 0 deletions

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;
+}