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Commit 04e805b0 authored by Phil Burk's avatar Phil Burk
Browse files

aaudio: add support for 24 and 32 bit PCM 

Add new formats: AAUDIO_FORMAT_PCM_I24_PACKED
and AAUDIO_FORMAT_PCM_I32.

Pass formats to Legacy AudioTrack and AuddioRecord.

Request 24_PACKED in AAudio Service for MMAP streams.
Also add a flowgraph module for converting the I32 data
to/from FLOAT.

Bug:  65067568
Bug: 157671580

Test: To determine which format the MMAP path is using for the DSP,
Test: adb shell dumpsys media.aaudio
Test: then look for the "Exclusive MMAP Endpoints" and "Format:".
Test: The formats listed are the internal AudioFlinger formats:
Test:    1=16_BIT, 3=32_BIT, 4=8_24_BIT, 5=FLOAT, 6=24_BIT_PACKED

Test: atest AAudioTestCases
Test: adb shell write_sine_callback -? # to list new formats
Test: adb shell write_sine_callback -pl -f3 # MMAP LOWLAT I24_PACKED
Test: adb shell write_sine_callback -pl -f4 # MMAP LOWLAT I32
Test: # Legacy LOWLAT I24_PACKED
Test: adb shell write_sine_callback -pl -f3 -m1
Test: adb shell write_sine_callback -pl -f4 -m1 # Legacy LOWLAT I32
Test: adb shell write_sine_callback -pn -f3 # Legacy I24_PACKED
Test: adb shell write_sine_callback -pn -f4 # Legacy I32
Change-Id: Ibe13bfd54425d110f50f89eb10c63872a2f99839
parent bb29fdf9

media/libaaudio/include/aaudio/AAudio.h

+26 −2
Original line number Diff line number Diff line
@@ -65,7 +65,7 @@ enum { 


    /**

     * This format uses the int16_t data type.

     * The maximum range of the data is -32768 to 32767.

     * The maximum range of the data is -32768 (0x8000) to 32767 (0x7FFF).

     */

    AAUDIO_FORMAT_PCM_I16,
@@ -77,7 +77,31 @@ enum { 
     * See also 'floatData' at

     * https://developer.android.com/reference/android/media/AudioTrack#write(float[],%20int,%20int,%20int)

     */

    AAUDIO_FORMAT_PCM_FLOAT

    AAUDIO_FORMAT_PCM_FLOAT,



    /**

     * This format uses 24-bit samples packed into 3 bytes.

     * The bytes are in the native endian order.

     * The maximum range of the data is -8388608 (0x800000)

     * to 8388607 (0x7FFFFF).

     *

     * Note that the lower precision bits may be ignored by the device.

     *

     * Available since API level 31.

     */

    AAUDIO_FORMAT_PCM_I24_PACKED,



    /**

     * This format uses 32-bit samples stored in an int32_t data type.

     * The maximum range of the data is -2147483648 (0x80000000)

     * to 2147483647 (0x7FFFFFFF).

     *

     * Note that the lower precision bits may be ignored by the device.

     *

     * Available since API level 31.

     */

    AAUDIO_FORMAT_PCM_I32



};

typedef int32_t aaudio_format_t;

media/libaaudio/src/Android.bp

+2 −0
Original line number Diff line number Diff line
@@ -130,9 +130,11 @@ cc_library { 
        "flowgraph/SinkFloat.cpp",

        "flowgraph/SinkI16.cpp",

        "flowgraph/SinkI24.cpp",

        "flowgraph/SinkI32.cpp",

        "flowgraph/SourceFloat.cpp",

        "flowgraph/SourceI16.cpp",

        "flowgraph/SourceI24.cpp",

        "flowgraph/SourceI32.cpp",

    ],

    sanitize: {

        integer_overflow: true,

media/libaaudio/src/client/AAudioFlowGraph.cpp

+12 −3
Original line number Diff line number Diff line
@@ -26,9 +26,11 @@ 
#include <flowgraph/SinkFloat.h>

#include <flowgraph/SinkI16.h>

#include <flowgraph/SinkI24.h>

#include <flowgraph/SinkI32.h>

#include <flowgraph/SourceFloat.h>

#include <flowgraph/SourceI16.h>

#include <flowgraph/SourceI24.h>

#include <flowgraph/SourceI32.h>



using namespace flowgraph;
@@ -38,7 +40,8 @@ aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat, 
                          int32_t sinkChannelCount) {

    AudioFloatOutputPort *lastOutput = nullptr;



    ALOGV("%s() source format = 0x%08x, channels = %d, sink format = 0x%08x, channels = %d",

    // TODO change back to ALOGD

    ALOGI("%s() source format = 0x%08x, channels = %d, sink format = 0x%08x, channels = %d",

          __func__, sourceFormat, sourceChannelCount, sinkFormat, sinkChannelCount);



    switch (sourceFormat) {
@@ -51,7 +54,10 @@ aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat, 
        case AUDIO_FORMAT_PCM_24_BIT_PACKED:

            mSource = std::make_unique<SourceI24>(sourceChannelCount);

            break;

        default: // TODO add I32

        case AUDIO_FORMAT_PCM_32_BIT:

            mSource = std::make_unique<SourceI32>(sourceChannelCount);

            break;

        default:

            ALOGE("%s() Unsupported source format = %d", __func__, sourceFormat);

            return AAUDIO_ERROR_UNIMPLEMENTED;

    }
@@ -90,7 +96,10 @@ aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat, 
        case AUDIO_FORMAT_PCM_24_BIT_PACKED:

            mSink = std::make_unique<SinkI24>(sinkChannelCount);

            break;

        default: // TODO add I32

        case AUDIO_FORMAT_PCM_32_BIT:

            mSink = std::make_unique<SinkI32>(sinkChannelCount);

            break;

        default:

            ALOGE("%s() Unsupported sink format = %d", __func__, sinkFormat);

            return AAUDIO_ERROR_UNIMPLEMENTED;

    }

media/libaaudio/src/client/AudioStreamInternal.cpp

+1 −1
Original line number Diff line number Diff line
@@ -104,7 +104,7 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { 
    if (getFormat() == AUDIO_FORMAT_DEFAULT) {

        setFormat(AUDIO_FORMAT_PCM_FLOAT);

    }

    // Request FLOAT for the shared mixer.

    // Request FLOAT for the shared mixer or the device.

    request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);



    // Build the request to send to the server.

media/libaaudio/src/core/AAudioStreamParameters.cpp

+2 −0
Original line number Diff line number Diff line
@@ -54,7 +54,9 @@ static aaudio_result_t isFormatValid(audio_format_t format) { 
    switch (format) {

        case AUDIO_FORMAT_DEFAULT:

        case AUDIO_FORMAT_PCM_16_BIT:

        case AUDIO_FORMAT_PCM_32_BIT:

        case AUDIO_FORMAT_PCM_FLOAT:

        case AUDIO_FORMAT_PCM_24_BIT_PACKED:

            break; // valid

        default:

            ALOGD("audioFormat not valid, audio_format_t = 0x%08x", format);