From 7f47549516ae5938759b5c834c8423378a60b3d8 Mon Sep 17 00:00:00 2001 From: Andy Hung Date: Mon, 25 Aug 2014 16:36:37 -0700 Subject: Fix AudioMixer floating interaction with downmixer Previously _if_ the full floating point mixer is enabled, a downmixer would force the mixer input for a session submix to integer, breaking other mixer inputs to the same submix that were in float. Use another ReformatBufferProvider after the downmixer to solve this issue. Update the test-mixer app and the mixer_to_wave_tests shell script to detect this issue. Bug: 17363939 Change-Id: I74a56333f9ee75ddde39a75392c021c5eebddbef --- services/audioflinger/tests/mixer_to_wav_tests.sh | 12 ++- services/audioflinger/tests/test-mixer.cpp | 92 ++++++++++++++--------- 2 files changed, 68 insertions(+), 36 deletions(-) (limited to 'services/audioflinger/tests') diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh index 9b39e77..e60e6d5 100755 --- a/services/audioflinger/tests/mixer_to_wav_tests.sh +++ b/services/audioflinger/tests/mixer_to_wav_tests.sh @@ -63,8 +63,18 @@ function createwav() { # process__genericResampling # track__Resample / track__genericResample adb shell test-mixer $1 -s 48000 \ + -o /sdcard/tm48000grif.wav \ + sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \ + sine:f,6,6000,19000 chirp:i,4,30000 + adb pull /sdcard/tm48000grif.wav $2 + +# 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 + sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \ + sine:6,6000,19000 adb pull /sdcard/tm48000gr.wav $2 # Test: diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp index 9a4fad6..8da6245 100644 --- a/services/audioflinger/tests/test-mixer.cpp +++ b/services/audioflinger/tests/test-mixer.cpp @@ -39,7 +39,7 @@ static void usage(const char* name) { fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]" " [-s sample-rate] [-o ] [-a ] [-P csv]" " ( | )+\n", name); - fprintf(stderr, " -f enable floating point input track\n"); + fprintf(stderr, " -f enable floating point input track by default\n"); fprintf(stderr, " -m enable floating point mixer output\n"); fprintf(stderr, " -c number of mixer output channels\n"); fprintf(stderr, " -s mixer sample-rate\n"); @@ -47,8 +47,8 @@ static void usage(const char* name) { fprintf(stderr, " -a \n"); fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n"); fprintf(stderr, " is a WAV file\n"); - fprintf(stderr, " can be 'sine:,,'\n"); - fprintf(stderr, " 'chirp:,'\n"); + fprintf(stderr, " can be 'sine:[(i|f),],,'\n"); + fprintf(stderr, " 'chirp:[(i|f),],'\n"); } static int writeFile(const char *filename, const void *buffer, @@ -78,6 +78,18 @@ static int writeFile(const char *filename, const void *buffer, return EXIT_SUCCESS; } +const char *parseFormat(const char *s, bool *useFloat) { + if (!strncmp(s, "f,", 2)) { + *useFloat = true; + return s + 2; + } + if (!strncmp(s, "i,", 2)) { + *useFloat = false; + return s + 2; + } + return s; +} + int main(int argc, char* argv[]) { const char* const progname = argv[0]; bool useInputFloat = false; @@ -88,8 +100,9 @@ int main(int argc, char* argv[]) { std::vector Pvalues; const char* outputFilename = NULL; const char* auxFilename = NULL; - std::vector Names; - std::vector Providers; + std::vector names; + std::vector providers; + std::vector formats; for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) { switch (ch) { @@ -138,54 +151,65 @@ int main(int argc, char* argv[]) { size_t outputFrames = 0; // create providers for each track - Providers.resize(argc); + names.resize(argc); + providers.resize(argc); + formats.resize(argc); for (int i = 0; i < argc; ++i) { static const char chirp[] = "chirp:"; static const char sine[] = "sine:"; static const double kSeconds = 1; + bool useFloat = useInputFloat; if (!strncmp(argv[i], chirp, strlen(chirp))) { std::vector v; + const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat); - parseCSV(argv[i] + strlen(chirp), v); + parseCSV(s, v); if (v.size() == 2) { printf("creating chirp(%d %d)\n", v[0], v[1]); - if (useInputFloat) { - Providers[i].setChirp(v[0], 0, v[1]/2, v[1], kSeconds); + if (useFloat) { + providers[i].setChirp(v[0], 0, v[1]/2, v[1], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setChirp(v[0], 0, v[1]/2, v[1], kSeconds); + providers[i].setChirp(v[0], 0, v[1]/2, v[1], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + providers[i].setIncr(Pvalues); } else { fprintf(stderr, "malformed input '%s'\n", argv[i]); } } else if (!strncmp(argv[i], sine, strlen(sine))) { std::vector v; + const char *s = parseFormat(argv[i] + strlen(sine), &useFloat); - parseCSV(argv[i] + strlen(sine), v); + parseCSV(s, v); if (v.size() == 3) { printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]); - if (useInputFloat) { - Providers[i].setSine(v[0], v[1], v[2], kSeconds); + if (useFloat) { + providers[i].setSine(v[0], v[1], v[2], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setSine(v[0], v[1], v[2], kSeconds); + providers[i].setSine(v[0], v[1], v[2], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + 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(argv[i]); + providers[i].setFile(argv[i]); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setFile(argv[i]); + providers[i].setFile(argv[i]); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + providers[i].setIncr(Pvalues); } // calculate the number of output frames - size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate - / Providers[i].getSampleRate(); + size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate + / providers[i].getSampleRate(); if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames outputFrames = nframes; } @@ -213,22 +237,20 @@ int main(int argc, char* argv[]) { // 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 + 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()); + 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); + formats[i], AUDIO_SESSION_OUTPUT_MIX); ALOG_ASSERT(name >= 0); - Names.push_back(name); - mixer->setBufferProvider(name, &Providers[i]); + names[i] = name; + mixer->setBufferProvider(name, &providers[i]); mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, (void *)outputAddr); mixer->setParameter( @@ -240,7 +262,7 @@ int main(int argc, char* argv[]) { name, AudioMixer::TRACK, AudioMixer::FORMAT, - (void *)(uintptr_t)inputFormat); + (void *)(uintptr_t)formats[i]); mixer->setParameter( name, AudioMixer::TRACK, @@ -255,7 +277,7 @@ int main(int argc, char* argv[]) { name, AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, - (void *)(uintptr_t)Providers[i].getSampleRate()); + (void *)(uintptr_t)providers[i].getSampleRate()); if (useRamp) { mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0); mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0); @@ -277,11 +299,11 @@ int main(int argc, char* argv[]) { // 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, + 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, + mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER, (char *) auxAddr + i * auxFrameSize); } } -- cgit v1.1