Snap for 6601700 from e28883fe1be2f291516adb167ff0dc4367f8adf5 to rvc-release (acc9757b) · Commits · e / devices / smdk4412 / android_frameworks_av

media/codec2/components/aac/C2SoftAacDec.cpp

+64 −4

Original line number	Diff line number	Diff line
		@@ -89,10 +89,17 @@ public:
		addParameter(
		DefineParam(mChannelCount, C2_PARAMKEY_CHANNEL_COUNT)
		.withDefault(new C2StreamChannelCountInfo::output(0u, 1))
		.withFields({C2F(mChannelCount, value).inRange(1, 8)})
		.withFields({C2F(mChannelCount, value).inRange(1, MAX_CHANNEL_COUNT)})
		.withSetter(Setter<decltype(*mChannelCount)>::StrictValueWithNoDeps)
		.build());

		addParameter(
		DefineParam(mMaxChannelCount, C2_PARAMKEY_MAX_CHANNEL_COUNT)
		.withDefault(new C2StreamMaxChannelCountInfo::input(0u, MAX_CHANNEL_COUNT))
		.withFields({C2F(mMaxChannelCount, value).inRange(1, MAX_CHANNEL_COUNT)})
		.withSetter(Setter<decltype(*mMaxChannelCount)>::StrictValueWithNoDeps)
		.build());

		addParameter(
		DefineParam(mBitrate, C2_PARAMKEY_BITRATE)
		.withDefault(new C2StreamBitrateInfo::input(0u, 64000))
		@@ -225,6 +232,7 @@ public:
		int32_t getDrcAttenuationFactor() const { return mDrcAttenuationFactor->value * 127. + 0.5; }
		int32_t getDrcEffectType() const { return mDrcEffectType->value; }
		int32_t getDrcAlbumMode() const { return mDrcAlbumMode->value; }
		u_int32_t getMaxChannelCount() const { return mMaxChannelCount->value; }
		int32_t getDrcOutputLoudness() const { return (mDrcOutputLoudness->value <= 0 ? -mDrcOutputLoudness->value * 4. + 0.5 : -1); }

		private:
		@@ -241,6 +249,7 @@ private:
		std::shared_ptr<C2StreamDrcAttenuationFactorTuning::input> mDrcAttenuationFactor;
		std::shared_ptr<C2StreamDrcEffectTypeTuning::input> mDrcEffectType;
		std::shared_ptr<C2StreamDrcAlbumModeTuning::input> mDrcAlbumMode;
		std::shared_ptr<C2StreamMaxChannelCountInfo::input> mMaxChannelCount;
		std::shared_ptr<C2StreamDrcOutputLoudnessTuning::output> mDrcOutputLoudness;
		// TODO Add : C2StreamAacSbrModeTuning
		};
		@@ -366,9 +375,10 @@ status_t C2SoftAacDec::initDecoder() {
		ALOGV("AAC decoder using MPEG-D DRC album mode %d", albumMode);
		aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_ALBUM_MODE, albumMode);

		// By default, the decoder creates a 5.1 channel downmix signal.
		// For seven and eight channel input streams, enable 6.1 and 7.1 channel output
		aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1);
		// AAC_PCM_MAX_OUTPUT_CHANNELS
		u_int32_t maxChannelCount = mIntf->getMaxChannelCount();
		ALOGV("AAC decoder using maximum output channel count %d", maxChannelCount);
		aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, maxChannelCount);

		return status;
		}
		@@ -707,6 +717,11 @@ void C2SoftAacDec::process(
		ALOGV("AAC decoder using MPEG-D DRC album mode %d", albumMode);
		aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_ALBUM_MODE, albumMode);

		// AAC_PCM_MAX_OUTPUT_CHANNELS
		int32_t maxChannelCount = mIntf->getMaxChannelCount();
		ALOGV("AAC decoder using maximum output channel count %d", maxChannelCount);
		aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, maxChannelCount);

		mDrcWrap.update();

		UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
		@@ -847,6 +862,51 @@ void C2SoftAacDec::process(
		ALOGE("Getting output loudness failed");
		}
		}

		// update config with values used for decoding:
		// Album mode, target reference level, DRC effect type, DRC attenuation and boost
		// factor, DRC compression mode, encoder target level and max channel count
		// with input values as they were not modified by decoder

		C2StreamDrcAttenuationFactorTuning::input currentAttenuationFactor(0u,
		(C2FloatValue) (attenuationFactor/127.));
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentAttenuationFactor));

		C2StreamDrcBoostFactorTuning::input currentBoostFactor(0u,
		(C2FloatValue) (boostFactor/127.));
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentBoostFactor));

		C2StreamDrcCompressionModeTuning::input currentCompressMode(0u,
		(C2Config::drc_compression_mode_t) compressMode);
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentCompressMode));

		C2StreamDrcEncodedTargetLevelTuning::input currentEncodedTargetLevel(0u,
		(C2FloatValue) (encTargetLevel*-0.25));
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentEncodedTargetLevel));

		C2StreamDrcAlbumModeTuning::input currentAlbumMode(0u,
		(C2Config::drc_album_mode_t) albumMode);
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentAlbumMode));

		C2StreamDrcTargetReferenceLevelTuning::input currentTargetRefLevel(0u,
		(float) (targetRefLevel*-0.25));
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentTargetRefLevel));

		C2StreamDrcEffectTypeTuning::input currentEffectype(0u,
		(C2Config::drc_effect_type_t) effectType);
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentEffectype));

		C2StreamMaxChannelCountInfo::input currentMaxChannelCnt(0u, maxChannelCount);
		work->worklets.front()->output.configUpdate.push_back(
		C2Param::Copy(currentMaxChannelCnt));

		} while (decoderErr == AAC_DEC_OK);
		}

media/codec2/sfplugin/CCodecBufferChannel.cpp

+4 −1

Original line number	Diff line number	Diff line
		@@ -732,6 +732,9 @@ status_t CCodecBufferChannel::renderOutputBuffer(
		std::shared_ptr<const C2StreamHdr10PlusInfo::output> hdr10PlusInfo =
		std::static_pointer_cast<const C2StreamHdr10PlusInfo::output>(
		c2Buffer->getInfo(C2StreamHdr10PlusInfo::output::PARAM_TYPE));
		if (hdr10PlusInfo && hdr10PlusInfo->flexCount() == 0) {
		hdr10PlusInfo.reset();
		}

		{
		Mutexed<OutputSurface>::Locked output(mOutputSurface);
		@@ -783,7 +786,7 @@ status_t CCodecBufferChannel::renderOutputBuffer(
		.maxLuminance = hdrStaticInfo->mastering.maxLuminance,
		.minLuminance = hdrStaticInfo->mastering.minLuminance,
		};
		hdr.validTypes = HdrMetadata::SMPTE2086;
		hdr.validTypes \|= HdrMetadata::SMPTE2086;
		hdr.smpte2086 = smpte2086_meta;
		}
		// If the content light level fields are 0, do not use them, it

services/audioflinger/AudioStreamOut.cpp

+4 −0

Original line number	Diff line number	Diff line
		@@ -164,6 +164,10 @@ status_t AudioStreamOut::open(
		stream = outStream;
		mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
		status = stream->getFrameSize(&mHalFrameSize);
		LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
		LOG_ALWAYS_FATAL_IF(mHalFrameSize <= 0, "Error frame size was %zu but must be greater than"
		" zero", mHalFrameSize);

		}

		return status;

services/audioflinger/Threads.cpp

+7 −4

Original line number	Diff line number	Diff line
		@@ -8436,13 +8436,14 @@ void AudioFlinger::RecordThread::readInputParameters_l()
		}
		result = mInput->stream->getFrameSize(&mFrameSize);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
		LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
		mFrameSize);
		result = mInput->stream->getBufferSize(&mBufferSize);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
		mFrameCount = mBufferSize / mFrameSize;
		ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%lld, "
		"mBufferSize=%lld, mFrameCount=%lld",
		this, mChannelCount, mFormat, (long long)mFrameSize, (long long)mBufferSize,
		(long long)mFrameCount);
		ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
		"mBufferSize=%zu, mFrameCount=%zu",
		this, mChannelCount, mFormat, mFrameSize, mBufferSize, mFrameCount);
		// This is the formula for calculating the temporary buffer size.
		// With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
		// 1 full output buffer, regardless of the alignment of the available input.
		@@ -9018,6 +9019,8 @@ void AudioFlinger::MmapThread::readHalParameters_l()
		LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat);
		result = mHalStream->getFrameSize(&mFrameSize);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
		LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
		mFrameSize);
		result = mHalStream->getBufferSize(&mBufferSize);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
		mFrameCount = mBufferSize / mFrameSize;

services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp

+1 −1

Original line number	Diff line number	Diff line
		@@ -593,7 +593,7 @@ status_t AudioPolicyService::startInput(audio_port_handle_t portId)
		}

		// including successes gets very verbose
		// but once we cut over to westworld, log them all.
		// but once we cut over to statsd, log them all.
		if (status != NO_ERROR) {

		static constexpr char kAudioPolicy[] = "audiopolicy";