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Commit d6fadf04 authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Simplify track 'needs' bits 

Use more standard coding convention for bit masks,
and add a FIXME about max channel count.

Change-Id: I856784016703417ee480b92ae73757c472f9cf95
parent 8f32537d

services/audioflinger/AudioMixer.cpp

+13 −11
Original line number Diff line number Diff line
@@ -644,27 +644,29 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) 
        countActiveTracks++;

        track_t& t = state->tracks[i];

        uint32_t n = 0;

        // FIXME can overflow (mask is only 3 bits)

        n |= NEEDS_CHANNEL_1 + t.channelCount - 1;

        n |= NEEDS_FORMAT_16;

        n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;

        if (t.doesResample()) {

            n |= NEEDS_RESAMPLE;

        }

        if (t.auxLevel != 0 && t.auxBuffer != NULL) {

            n |= NEEDS_AUX_ENABLED;

            n |= NEEDS_AUX;

        }



        if (t.volumeInc[0]|t.volumeInc[1]) {

            volumeRamp = true;

        } else if (!t.doesResample() && t.volumeRL == 0) {

            n |= NEEDS_MUTE_ENABLED;

            n |= NEEDS_MUTE;

        }

        t.needs = n;



        if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {

        if (n & NEEDS_MUTE) {

            t.hook = track__nop;

        } else {

            if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {

            if (n & NEEDS_AUX) {

                all16BitsStereoNoResample = false;

            }

            if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {

            if (n & NEEDS_RESAMPLE) {

                all16BitsStereoNoResample = false;

                resampling = true;

                t.hook = track__genericResample;
@@ -730,7 +732,7 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) 
            en &= ~(1<<i);

            track_t& t = state->tracks[i];

            if (!t.doesResample() && t.volumeRL == 0) {

                t.needs |= NEEDS_MUTE_ENABLED;

                t.needs |= NEEDS_MUTE;

                t.hook = track__nop;

            } else {

                allMuted = false;
@@ -1134,7 +1136,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 
                track_t& t = state->tracks[i];

                size_t outFrames = BLOCKSIZE;

                int32_t *aux = NULL;

                if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {

                if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {

                    aux = t.auxBuffer + numFrames;

                }

                while (outFrames) {
@@ -1215,14 +1217,14 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 
            e1 &= ~(1<<i);

            track_t& t = state->tracks[i];

            int32_t *aux = NULL;

            if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {

            if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {

                aux = t.auxBuffer;

            }



            // this is a little goofy, on the resampling case we don't

            // acquire/release the buffers because it's done by

            // the resampler.

            if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {

            if (t.needs & NEEDS_RESAMPLE) {

                t.resampler->setPTS(pts);

                t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);

            } else {

services/audioflinger/AudioMixer.h

+7 −15
Original line number Diff line number Diff line
@@ -120,27 +120,19 @@ public: 
private:



    enum {

        // FIXME this representation permits up to 8 channels

        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,

        NEEDS_FORMAT__MASK          = 0x000000F0,

        NEEDS_MUTE__MASK            = 0x00000100,

        NEEDS_RESAMPLE__MASK        = 0x00001000,

        NEEDS_AUX__MASK             = 0x00010000,

    };



    enum {

        NEEDS_CHANNEL_1             = 0x00000000,

        NEEDS_CHANNEL_2             = 0x00000001,

        NEEDS_CHANNEL_1             = 0x00000000,   // mono

        NEEDS_CHANNEL_2             = 0x00000001,   // stereo



        NEEDS_FORMAT_16             = 0x00000010,

        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT



        NEEDS_MUTE_DISABLED         = 0x00000000,

        NEEDS_MUTE_ENABLED          = 0x00000100,



        NEEDS_RESAMPLE_DISABLED     = 0x00000000,

        NEEDS_RESAMPLE_ENABLED      = 0x00001000,



        NEEDS_AUX_DISABLED     = 0x00000000,

        NEEDS_AUX_ENABLED      = 0x00010000,

        NEEDS_MUTE                  = 0x00000100,

        NEEDS_RESAMPLE              = 0x00001000,

        NEEDS_AUX                   = 0x00010000,

    };



    struct state_t;