AudioFlinger: Split off audio processing library
Test: native AudioResampler test, general playback test
Bug: 31015569
Change-Id: Ifb248f4402a583438d756c014dcd7a4577aef713
diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
new file mode 100644
index 0000000..87ada76
--- /dev/null
+++ b/include/media/AudioMixer.h
@@ -0,0 +1,389 @@
+/*
+**
+** Copyright 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_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/BufferProviders.h>
+#include <media/nbaio/NBLog.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+#include <utils/threads.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+ AudioMixer(size_t frameCount, uint32_t sampleRate,
+ uint32_t maxNumTracks = MAX_NUM_TRACKS);
+
+ /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
+
+
+ // This mixer has a hard-coded upper limit of 32 active track inputs.
+ // Adding support for > 32 tracks would require more than simply changing this value.
+ static const uint32_t MAX_NUM_TRACKS = 32;
+ // maximum number of channels supported by the mixer
+
+ // This mixer has a hard-coded upper limit of 8 channels for output.
+ static const uint32_t MAX_NUM_CHANNELS = 8;
+ static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
+ // maximum number of channels supported for the content
+ static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+ static const uint16_t UNITY_GAIN_INT = 0x1000;
+ static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+ enum { // names
+
+ // track names (MAX_NUM_TRACKS units)
+ TRACK0 = 0x1000,
+
+ // 0x2000 is unused
+
+ // setParameter targets
+ TRACK = 0x3000,
+ RESAMPLE = 0x3001,
+ RAMP_VOLUME = 0x3002, // ramp to new volume
+ VOLUME = 0x3003, // don't ramp
+ TIMESTRETCH = 0x3004,
+
+ // set Parameter names
+ // for target TRACK
+ CHANNEL_MASK = 0x4000,
+ FORMAT = 0x4001,
+ MAIN_BUFFER = 0x4002,
+ AUX_BUFFER = 0x4003,
+ DOWNMIX_TYPE = 0X4004,
+ MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ // for target RESAMPLE
+ SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
+ // parameter 'value' is the new sample rate in Hz.
+ // Only creates a sample rate converter the first time that
+ // the track sample rate is different from the mix sample rate.
+ // If the new sample rate is the same as the mix sample rate,
+ // and a sample rate converter already exists,
+ // then the sample rate converter remains present but is a no-op.
+ RESET = 0x4101, // Reset sample rate converter without changing sample rate.
+ // This clears out the resampler's input buffer.
+ REMOVE = 0x4102, // Remove the sample rate converter on this track name;
+ // the track is restored to the mix sample rate.
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
+ VOLUME0 = 0x4200,
+ VOLUME1 = 0x4201,
+ AUXLEVEL = 0x4210,
+ // for target TIMESTRETCH
+ PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
+ // parameter 'value' is a pointer to the new playback rate.
+ };
+
+
+ // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
+
+ // Allocate a track name. Returns new track name if successful, -1 on failure.
+ // The failure could be because of an invalid channelMask or format, or that
+ // the track capacity of the mixer is exceeded.
+ int getTrackName(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId);
+
+ // Free an allocated track by name
+ void deleteTrackName(int name);
+
+ // Enable or disable an allocated track by name
+ void enable(int name);
+ void disable(int name);
+
+ void setParameter(int name, int target, int param, void *value);
+
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+ void process();
+
+ uint32_t trackNames() const { return mTrackNames; }
+
+ size_t getUnreleasedFrames(int name) const;
+
+ static inline bool isValidPcmTrackFormat(audio_format_t format) {
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
+ }
+
+private:
+
+ enum {
+ // FIXME this representation permits up to 8 channels
+ NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
+ };
+
+ enum {
+ NEEDS_CHANNEL_1 = 0x00000000, // mono
+ NEEDS_CHANNEL_2 = 0x00000001, // stereo
+
+ // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+ NEEDS_MUTE = 0x00000100,
+ NEEDS_RESAMPLE = 0x00001000,
+ NEEDS_AUX = 0x00010000,
+ };
+
+ struct state_t;
+ struct track_t;
+
+ typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
+ int32_t* aux);
+ static const int BLOCKSIZE = 16; // 4 cache lines
+
+ struct track_t {
+ uint32_t needs;
+
+ // TODO: Eventually remove legacy integer volume settings
+ union {
+ int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+ int32_t volumeRL;
+ };
+
+ int32_t prevVolume[MAX_NUM_VOLUMES];
+
+ // 16-byte boundary
+
+ int32_t volumeInc[MAX_NUM_VOLUMES];
+ int32_t auxInc;
+ int32_t prevAuxLevel;
+
+ // 16-byte boundary
+
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+ uint16_t frameCount;
+
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t unused_padding; // formerly format, was always 16
+ uint16_t enabled; // actually bool
+ audio_channel_mask_t channelMask;
+
+ // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
+ // for how the Track buffer provider is wrapped by another one when dowmixing is required
+ AudioBufferProvider* bufferProvider;
+
+ // 16-byte boundary
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+ hook_t hook;
+ const void* in; // current location in buffer
+
+ // 16-byte boundary
+
+ AudioResampler* resampler;
+ uint32_t sampleRate;
+ int32_t* mainBuffer;
+ int32_t* auxBuffer;
+
+ // 16-byte boundary
+
+ /* Buffer providers are constructed to translate the track input data as needed.
+ *
+ * TODO: perhaps make a single PlaybackConverterProvider class to move
+ * all pre-mixer track buffer conversions outside the AudioMixer class.
+ *
+ * 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.
+ * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
+ */
+ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
+ PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
+ PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
+ PassthruBufferProvider* mPostDownmixReformatBufferProvider;
+ PassthruBufferProvider* mTimestretchBufferProvider;
+
+ int32_t sessionId;
+
+ 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
+ float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
+
+ float mAuxLevel; // floating point set aux level
+ float mPrevAuxLevel; // floating point prev aux level
+ float mAuxInc; // floating point aux increment
+
+ audio_channel_mask_t mMixerChannelMask;
+ uint32_t mMixerChannelCount;
+
+ AudioPlaybackRate mPlaybackRate;
+
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+ bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+ bool doesResample() const { return resampler != NULL; }
+ void resetResampler() { if (resampler != NULL) resampler->reset(); }
+ void adjustVolumeRamp(bool aux, bool useFloat = false);
+ size_t getUnreleasedFrames() const { return resampler != NULL ?
+ resampler->getUnreleasedFrames() : 0; };
+
+ status_t prepareForDownmix();
+ void unprepareForDownmix();
+ status_t prepareForReformat();
+ void unprepareForReformat();
+ bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
+ void reconfigureBufferProviders();
+ };
+
+ typedef void (*process_hook_t)(state_t* state);
+
+ // pad to 32-bytes to fill cache line
+ struct state_t {
+ uint32_t enabledTracks;
+ uint32_t needsChanged;
+ size_t frameCount;
+ process_hook_t hook; // one of process__*, never NULL
+ int32_t *outputTemp;
+ int32_t *resampleTemp;
+ NBLog::Writer* mLog;
+ int32_t reserved[1];
+ // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
+ track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
+ };
+
+ // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
+ uint32_t mTrackNames;
+
+ // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
+ // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
+ const uint32_t mConfiguredNames;
+
+ const uint32_t mSampleRate;
+
+ NBLog::Writer mDummyLog;
+public:
+ void setLog(NBLog::Writer* log);
+private:
+ state_t mState __attribute__((aligned(32)));
+
+ // Call after changing either the enabled status of a track, or parameters of an enabled track.
+ // OK to call more often than that, but unnecessary.
+ void invalidateState(uint32_t mask);
+
+ bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+ static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+ int32_t* aux);
+ static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+ int32_t* aux);
+ static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+ int32_t* aux);
+
+ static void process__validate(state_t* state);
+ static void process__nop(state_t* state);
+ static void process__genericNoResampling(state_t* state);
+ static void process__genericResampling(state_t* state);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state);
+
+ static pthread_once_t sOnceControl;
+ static void sInitRoutine();
+
+ /* multi-format volume mixing function (calls template functions
+ * in AudioMixerOps.h). The template parameters are as follows:
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+ template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+ static void volumeMix(TO *out, size_t outFrames,
+ const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
+
+ // multi-format process hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void process_NoResampleOneTrack(state_t* state);
+
+ // multi-format track hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void track__Resample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK,
+ };
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // functions for determining the proper process and track hooks.
+ static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+ static hook_t getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H