diff options
Diffstat (limited to 'services/audioflinger')
-rw-r--r-- | services/audioflinger/AudioMixer.cpp | 68 | ||||
-rw-r--r-- | services/audioflinger/AudioMixer.h | 20 | ||||
-rwxr-xr-x | services/audioflinger/tests/mixer_to_wav_tests.sh | 12 | ||||
-rw-r--r-- | services/audioflinger/tests/test-mixer.cpp | 92 |
4 files changed, 136 insertions, 56 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index beab7c2..0d4b358 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -430,6 +430,10 @@ void AudioMixer::setLog(NBLog::Writer *log) mState.mLog = log; } +static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) { + return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; +} + int AudioMixer::getTrackName(audio_channel_mask_t channelMask, audio_format_t format, int sessionId) { @@ -492,10 +496,11 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, t->mInputBufferProvider = NULL; t->mReformatBufferProvider = NULL; t->downmixerBufferProvider = NULL; + t->mPostDownmixReformatBufferProvider = NULL; t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT; t->mFormat = format; - t->mMixerInFormat = kUseFloat && kUseNewMixer - ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; + t->mMixerInFormat = selectMixerInFormat(format); + t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits( AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO); t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask); @@ -505,9 +510,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask); return -1; } - // prepareForDownmix() may change the input format requirement. - // If you desire floating point input to the mixer, it may change - // to integer because the downmixer requires integer to process. + // prepareForDownmix() may change mDownmixRequiresFormat ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat); t->prepareForReformat(); mTrackNames |= 1 << n; @@ -526,7 +529,7 @@ void AudioMixer::invalidateState(uint32_t mask) } // Called when channel masks have changed for a track name -// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format, +// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format, // which will simplify this logic. bool AudioMixer::setChannelMasks(int name, audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) { @@ -551,21 +554,18 @@ bool AudioMixer::setChannelMasks(int name, // channel masks have changed, does this track need a downmixer? // update to try using our desired format (if we aren't already using it) - const audio_format_t prevMixerInFormat = track.mMixerInFormat; - track.mMixerInFormat = kUseFloat && kUseNewMixer - ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; + const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat; const status_t status = mState.tracks[name].prepareForDownmix(); ALOGE_IF(status != OK, "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x", status, track.channelMask, track.mMixerChannelMask); - const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat; - if (mixerInFormatChanged) { + if (prevDownmixerFormat != track.mDownmixRequiresFormat) { track.prepareForReformat(); // because of downmixer, track format may change! } - if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) { - // resampler input format or channels may have changed. + if (track.resampler && mixerChannelCountChanged) { + // resampler channels may have changed. const uint32_t resetToSampleRate = track.sampleRate; delete track.resampler; track.resampler = NULL; @@ -579,6 +579,7 @@ bool AudioMixer::setChannelMasks(int name, void AudioMixer::track_t::unprepareForDownmix() { ALOGV("AudioMixer::unprepareForDownmix(%p)", this); + mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; if (downmixerBufferProvider != NULL) { // this track had previously been configured with a downmixer, delete it ALOGV(" deleting old downmixer"); @@ -611,7 +612,7 @@ status_t AudioMixer::track_t::prepareForDownmix() sampleRate, sessionId, kCopyBufferFrameCount); if (pDbp->isValid()) { // if constructor completed properly - mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix + mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix downmixerBufferProvider = pDbp; reconfigureBufferProviders(); return NO_ERROR; @@ -630,9 +631,18 @@ status_t AudioMixer::track_t::prepareForDownmix() void AudioMixer::track_t::unprepareForReformat() { ALOGV("AudioMixer::unprepareForReformat(%p)", this); + bool requiresReconfigure = false; if (mReformatBufferProvider != NULL) { delete mReformatBufferProvider; mReformatBufferProvider = NULL; + requiresReconfigure = true; + } + if (mPostDownmixReformatBufferProvider != NULL) { + delete mPostDownmixReformatBufferProvider; + mPostDownmixReformatBufferProvider = NULL; + requiresReconfigure = true; + } + if (requiresReconfigure) { reconfigureBufferProviders(); } } @@ -640,14 +650,29 @@ void AudioMixer::track_t::unprepareForReformat() { status_t AudioMixer::track_t::prepareForReformat() { ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat); - // discard the previous reformatter if there was one + // discard previous reformatters unprepareForReformat(); - // only configure reformatter if needed - if (mFormat != mMixerInFormat) { + // only configure reformatters as needed + const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID + ? mDownmixRequiresFormat : mMixerInFormat; + bool requiresReconfigure = false; + if (mFormat != targetFormat) { mReformatBufferProvider = new ReformatBufferProvider( audio_channel_count_from_out_mask(channelMask), - mFormat, mMixerInFormat, + mFormat, + targetFormat, kCopyBufferFrameCount); + requiresReconfigure = true; + } + if (targetFormat != mMixerInFormat) { + mPostDownmixReformatBufferProvider = new ReformatBufferProvider( + audio_channel_count_from_out_mask(mMixerChannelMask), + targetFormat, + mMixerInFormat, + kCopyBufferFrameCount); + requiresReconfigure = true; + } + if (requiresReconfigure) { reconfigureBufferProviders(); } return NO_ERROR; @@ -664,6 +689,10 @@ void AudioMixer::track_t::reconfigureBufferProviders() downmixerBufferProvider->setBufferProvider(bufferProvider); bufferProvider = downmixerBufferProvider; } + if (mPostDownmixReformatBufferProvider) { + mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider); + bufferProvider = mPostDownmixReformatBufferProvider; + } } void AudioMixer::deleteTrackName(int name) @@ -1026,6 +1055,9 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider if (mState.tracks[name].mReformatBufferProvider != NULL) { mState.tracks[name].mReformatBufferProvider->reset(); } else if (mState.tracks[name].downmixerBufferProvider != NULL) { + mState.tracks[name].downmixerBufferProvider->reset(); + } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) { + mState.tracks[name].mPostDownmixReformatBufferProvider->reset(); } mState.tracks[name].mInputBufferProvider = bufferProvider; diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index 6ed6c1b..88e94c5 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -205,17 +205,34 @@ private: int32_t* auxBuffer; // 16-byte boundary + + /* Buffer providers are constructed to translate the track input data as needed. + * + * 1) mInputBufferProvider: The AudioTrack buffer provider. + * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to + * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer + * requires reformat. For example, it may convert floating point input to + * PCM_16_bit if that's required by the downmixer. + * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match + * the number of channels required by the mixer sink. + * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from + * the downmixer requirements to the mixer engine input requirements. + */ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. + CopyBufferProvider* mPostDownmixReformatBufferProvider; + // 16-byte boundary int32_t sessionId; - // 16-byte boundary audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) audio_format_t mFormat; // input track format audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) // each track must be converted to this format. + audio_format_t mDownmixRequiresFormat; // required downmixer format + // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary + // AUDIO_FORMAT_INVALID if no required format float mVolume[MAX_NUM_VOLUMES]; // floating point set volume float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume @@ -225,7 +242,6 @@ private: float mPrevAuxLevel; // floating point prev aux level float mAuxInc; // floating point aux increment - // 16-byte boundary audio_channel_mask_t mMixerChannelMask; uint32_t mMixerChannelCount; 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 <output-file>] [-a <aux-buffer-file>] [-P csv]" " (<input-file> | <command>)+\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 <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"); + fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n"); + fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\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<int> Pvalues; const char* outputFilename = NULL; const char* auxFilename = NULL; - std::vector<int32_t> Names; - std::vector<SignalProvider> Providers; + std::vector<int32_t> names; + std::vector<SignalProvider> providers; + std::vector<audio_format_t> 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<int> 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<float>(v[0], 0, v[1]/2, v[1], kSeconds); + if (useFloat) { + providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds); + providers[i].setChirp<int16_t>(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<int> 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<float>(v[0], v[1], v[2], kSeconds); + if (useFloat) { + providers[i].setSine<float>(v[0], v[1], v[2], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds); + providers[i].setSine<int16_t>(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<float>(argv[i]); + providers[i].setFile<float>(argv[i]); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setFile<short>(argv[i]); + providers[i].setFile<short>(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); } } |