Update sampling rate to 192kHz for devices (db4c031f) · Commits · e / os / android_frameworks_av

include/media/AudioResamplerPublic.h

+10 −0

Original line number	Diff line number	Diff line
		@@ -143,6 +143,16 @@ static inline size_t sourceFramesNeededWithTimestretch(
		return required * (double)speed + 1 + 1; // accounting for rounding dependencies
		}

		// Identifies sample rates that we associate with music
		// and thus eligible for better resampling and fast capture.
		// This is somewhat less than 44100 to allow for pitch correction
		// involving resampling as well as asynchronous resampling.
		#define AUDIO_PROCESSING_MUSIC_RATE 40000

		static inline bool isMusicRate(uint32_t sampleRate) {
		return sampleRate >= AUDIO_PROCESSING_MUSIC_RATE;
		}

		} // namespace android

		// ---------------------------------------------------------------------------

+3 −4

Original line number	Diff line number	Diff line
		@@ -708,11 +708,10 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam
		// FIXME this is flawed for dynamic sample rates, as we choose the resampler
		// quality level based on the initial ratio, but that could change later.
		// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
		if (!((trackSampleRate == 44100 && devSampleRate == 48000) \|\|
		(trackSampleRate == 48000 && devSampleRate == 44100))) {
		quality = AudioResampler::DYN_LOW_QUALITY;
		} else {
		if (isMusicRate(trackSampleRate)) {
		quality = AudioResampler::DEFAULT_QUALITY;
		} else {
		quality = AudioResampler::DYN_LOW_QUALITY;
		}

		// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer

+5 −2

Original line number	Diff line number	Diff line
		@@ -5328,11 +5328,11 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
		}
		initFastCapture =
		// either capture sample rate is same as (a reasonable) primary output sample rate
		(((primaryOutputSampleRate == 44100 \|\| primaryOutputSampleRate == 48000) &&
		((isMusicRate(primaryOutputSampleRate) &&
		(mSampleRate == primaryOutputSampleRate)) \|\|
		// or primary output sample rate is unknown, and capture sample rate is reasonable
		((primaryOutputSampleRate == 0) &&
		((mSampleRate == 44100 \|\| mSampleRate == 48000)))) &&
		isMusicRate(mSampleRate))) &&
		// and the buffer size is < 12 ms
		(mFrameCount * 1000) / mSampleRate < 12;
		break;
		@@ -6435,6 +6435,9 @@ status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
		return NO_ERROR;
		}

		ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
		" srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u",
		srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
		const bool valid =
		audio_is_input_channel(srcChannelMask)
		&& audio_is_input_channel(dstChannelMask)

+1 −1

Original line number	Diff line number	Diff line
		@@ -20,7 +20,7 @@

		// For mixed output and inputs, the policy will use max mixer sampling rates.
		// Do not limit sampling rate otherwise
		#define MAX_MIXER_SAMPLING_RATE 48000
		#define MAX_MIXER_SAMPLING_RATE 192000

		// For mixed output and inputs, the policy will use max mixer channel count.
		// Do not limit channel count otherwise

+4 −0

Original line number	Diff line number	Diff line
		@@ -611,9 +611,13 @@ uint32_t AudioPort::pickSamplingRate() const

		// For mixed output and inputs, use max mixer sampling rates. Do not
		// limit sampling rate otherwise
		// For inputs, also see checkCompatibleSamplingRate().
		if (mType != AUDIO_PORT_TYPE_MIX) {
		maxRate = UINT_MAX;
		}
		// TODO: should mSamplingRates[] be ordered in terms of our preference
		// and we return the first (and hence most preferred) match? This is of concern if
		// we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
		for (size_t i = 0; i < mSamplingRates.size(); i ++) {
		if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
		samplingRate = mSamplingRates[i];