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Commit 306c92da authored by Andy Hung's avatar Andy Hung Committed by Android (Google) Code Review
Browse files

Merge "Extend downsampling ratios greater than 2:1" into lmp-dev

parents de975ca8 cd04484f

include/media/AudioResamplerPublic.h

0 → 100644
+29 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2014 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_RESAMPLER_PUBLIC_H

#define ANDROID_AUDIO_RESAMPLER_PUBLIC_H



// AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original

// audio sample rate and the target rate when downsampling,

// as permitted in the audio framework, e.g. AudioTrack and AudioFlinger.

// In practice, it is not recommended to downsample more than 6:1

// for best audio quality, even though the audio framework permits a larger

// downsampling ratio.

// TODO: replace with an API

#define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256



#endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H

media/libmedia/AudioTrack.cpp

+4 −4
Original line number Diff line number Diff line
@@ -28,6 +28,7 @@ 
#include <utils/Log.h>

#include <private/media/AudioTrackShared.h>

#include <media/IAudioFlinger.h>

#include <media/AudioResamplerPublic.h>



#define WAIT_PERIOD_MS                  10

#define WAIT_STREAM_END_TIMEOUT_SEC     120
@@ -82,7 +83,7 @@ status_t AudioTrack::getMinFrameCount( 
    }



    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :

            afFrameCount * minBufCount * sampleRate / afSampleRate;

            afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate;

    // The formula above should always produce a non-zero value, but return an error

    // in the unlikely event that it does not, as that's part of the API contract.

    if (*frameCount == 0) {
@@ -646,8 +647,7 @@ status_t AudioTrack::setSampleRate(uint32_t rate) 
    if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {

        return NO_INIT;

    }

    // Resampler implementation limits input sampling rate to 2 x output sampling rate.

    if (rate == 0 || rate > afSamplingRate*2 ) {

    if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {

        return BAD_VALUE;

    }
@@ -1002,7 +1002,7 @@ status_t AudioTrack::createTrack_l(size_t epoch) 
            minBufCount = nBuffering;

        }



        size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;

        size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate;

        ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"

                ", afLatency=%d",

                minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);

services/audioflinger/AudioResamplerDyn.cpp

+1 −1
Original line number Diff line number Diff line
@@ -393,7 +393,7 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
    mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)

            * phaseWrapLimit / oldPhaseWrapLimit;

    mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.

    mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit)

    mPhaseIncrement = static_cast<uint32_t>(static_cast<uint64_t>(phaseWrapLimit)

            * inSampleRate / mSampleRate);



    // determine which resampler to use

services/audioflinger/Threads.cpp

+3 −3
Original line number Diff line number Diff line
@@ -26,6 +26,7 @@ 
#include <sys/stat.h>

#include <cutils/properties.h>

#include <media/AudioParameter.h>

#include <media/AudioResamplerPublic.h>

#include <utils/Log.h>

#include <utils/Trace.h>
@@ -1479,8 +1480,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac 
                lStatus = BAD_VALUE;

                goto Exit;

        }

        // Resampler implementation limits input sampling rate to 2 x output sampling rate.

        if (sampleRate > mSampleRate*2) {

        if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {

            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);

            lStatus = BAD_VALUE;

            goto Exit;
@@ -3500,7 +3500,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 
                AudioMixer::TRACK,

                AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);

            // limit track sample rate to 2 x output sample rate, which changes at re-configuration

            uint32_t maxSampleRate = mSampleRate * 2;

            uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;

            uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();

            if (reqSampleRate == 0) {

                reqSampleRate = mSampleRate;