diff options
107 files changed, 4571 insertions, 1051 deletions
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp index 02df1d2..36a7e73 100644 --- a/cmds/screenrecord/screenrecord.cpp +++ b/cmds/screenrecord/screenrecord.cpp @@ -23,7 +23,10 @@ #include <stdio.h> #include <stdlib.h> #include <string.h> +#include <sys/stat.h> +#include <sys/types.h> #include <sys/wait.h> + #include <termios.h> #include <unistd.h> @@ -637,7 +640,13 @@ static status_t recordScreen(const char* fileName) { case FORMAT_MP4: { // Configure muxer. We have to wait for the CSD blob from the encoder // before we can start it. - muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4); + int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + if (fd < 0) { + fprintf(stderr, "ERROR: couldn't open file\n"); + abort(); + } + muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4); + close(fd); if (gRotate) { muxer->setOrientationHint(90); // TODO: does this do anything? } diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk index 561ce02..0e3bc68 100644 --- a/cmds/stagefright/Android.mk +++ b/cmds/stagefright/Android.mk @@ -169,6 +169,48 @@ include $(BUILD_EXECUTABLE) include $(CLEAR_VARS) +LOCAL_SRC_FILES:= \ + filters/argbtorgba.rs \ + filters/nightvision.rs \ + filters/saturation.rs \ + mediafilter.cpp \ + +LOCAL_SHARED_LIBRARIES := \ + libstagefright \ + liblog \ + libutils \ + libbinder \ + libstagefright_foundation \ + libmedia \ + libgui \ + libcutils \ + libui \ + libRScpp \ + +LOCAL_C_INCLUDES:= \ + $(TOP)/frameworks/av/media/libstagefright \ + $(TOP)/frameworks/native/include/media/openmax \ + $(TOP)/frameworks/rs/cpp \ + $(TOP)/frameworks/rs \ + +intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,) +LOCAL_C_INCLUDES += $(intermediates) + +LOCAL_STATIC_LIBRARIES:= \ + libstagefright_mediafilter + +LOCAL_CFLAGS += -Wno-multichar + +LOCAL_MODULE_TAGS := optional + +LOCAL_MODULE:= mediafilter + +include $(BUILD_EXECUTABLE) + +################################################################################ + +include $(CLEAR_VARS) + LOCAL_SRC_FILES:= \ muxer.cpp \ diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp index 96073f1..7b0de24 100644 --- a/cmds/stagefright/audioloop.cpp +++ b/cmds/stagefright/audioloop.cpp @@ -14,6 +14,10 @@ * limitations under the License. */ +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> + #include <binder/ProcessState.h> #include <media/mediarecorder.h> #include <media/stagefright/foundation/ADebug.h> @@ -109,7 +113,12 @@ int main(int argc, char* argv[]) if (fileOut != NULL) { // target file specified, write encoded AMR output - sp<AMRWriter> writer = new AMRWriter(fileOut); + int fd = open(fileOut, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + if (fd < 0) { + return 1; + } + sp<AMRWriter> writer = new AMRWriter(fd); + close(fd); writer->addSource(encoder); writer->start(); sleep(duration); diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp index fd02bcc..d987250 100644 --- a/cmds/stagefright/codec.cpp +++ b/cmds/stagefright/codec.cpp @@ -45,9 +45,10 @@ static void usage(const char *me) { fprintf(stderr, "usage: %s [-a] use audio\n" "\t\t[-v] use video\n" "\t\t[-p] playback\n" - "\t\t[-S] allocate buffers from a surface\n", + "\t\t[-S] allocate buffers from a surface\n" + "\t\t[-R] render output to surface (enables -S)\n" + "\t\t[-T] use render timestamps (enables -R)\n", me); - exit(1); } @@ -71,7 +72,9 @@ static int decode( const char *path, bool useAudio, bool useVideo, - const android::sp<android::Surface> &surface) { + const android::sp<android::Surface> &surface, + bool renderSurface, + bool useTimestamp) { using namespace android; static int64_t kTimeout = 500ll; @@ -136,6 +139,7 @@ static int decode( CHECK(!stateByTrack.isEmpty()); int64_t startTimeUs = ALooper::GetNowUs(); + int64_t startTimeRender = -1; for (size_t i = 0; i < stateByTrack.size(); ++i) { CodecState *state = &stateByTrack.editValueAt(i); @@ -260,7 +264,23 @@ static int decode( ++state->mNumBuffersDecoded; state->mNumBytesDecoded += size; - err = state->mCodec->releaseOutputBuffer(index); + if (surface == NULL || !renderSurface) { + err = state->mCodec->releaseOutputBuffer(index); + } else if (useTimestamp) { + if (startTimeRender == -1) { + // begin rendering 2 vsyncs (~33ms) after first decode + startTimeRender = + systemTime(SYSTEM_TIME_MONOTONIC) + 33000000 + - (presentationTimeUs * 1000); + } + presentationTimeUs = + (presentationTimeUs * 1000) + startTimeRender; + err = state->mCodec->renderOutputBufferAndRelease( + index, presentationTimeUs); + } else { + err = state->mCodec->renderOutputBufferAndRelease(index); + } + CHECK_EQ(err, (status_t)OK); if (flags & MediaCodec::BUFFER_FLAG_EOS) { @@ -320,34 +340,42 @@ int main(int argc, char **argv) { bool useVideo = false; bool playback = false; bool useSurface = false; + bool renderSurface = false; + bool useTimestamp = false; int res; - while ((res = getopt(argc, argv, "havpSD")) >= 0) { + while ((res = getopt(argc, argv, "havpSDRT")) >= 0) { switch (res) { case 'a': { useAudio = true; break; } - case 'v': { useVideo = true; break; } - case 'p': { playback = true; break; } - + case 'T': + { + useTimestamp = true; + } + // fall through + case 'R': + { + renderSurface = true; + } + // fall through case 'S': { useSurface = true; break; } - case '?': case 'h': default: @@ -422,7 +450,8 @@ int main(int argc, char **argv) { player->stop(); player->reset(); } else { - decode(looper, argv[0], useAudio, useVideo, surface); + decode(looper, argv[0], useAudio, useVideo, surface, renderSurface, + useTimestamp); } if (playback || (useSurface && useVideo)) { diff --git a/include/media/nbaio/roundup.h b/cmds/stagefright/filters/argbtorgba.rs index 4c3cc25..229ff8c 100644 --- a/include/media/nbaio/roundup.h +++ b/cmds/stagefright/filters/argbtorgba.rs @@ -1,5 +1,5 @@ /* - * Copyright (C) 2012 The Android Open Source Project + * 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. @@ -14,18 +14,13 @@ * limitations under the License. */ -#ifndef ROUNDUP_H -#define ROUNDUP_H +#pragma version(1) +#pragma rs java_package_name(com.android.rs.cppbasic) +#pragma rs_fp_relaxed -#ifdef __cplusplus -extern "C" { -#endif - -// Round up to the next highest power of 2 -unsigned roundup(unsigned v); - -#ifdef __cplusplus -} -#endif - -#endif // ROUNDUP_H +void root(const uchar4 *v_in, uchar4 *v_out) { + v_out->x = v_in->y; + v_out->y = v_in->z; + v_out->z = v_in->w; + v_out->w = v_in->x; +}
\ No newline at end of file diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rs new file mode 100644 index 0000000..f61413c --- /dev/null +++ b/cmds/stagefright/filters/nightvision.rs @@ -0,0 +1,38 @@ +/* + * 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. + */ + +#pragma version(1) +#pragma rs java_package_name(com.android.rs.cppbasic) +#pragma rs_fp_relaxed + +const static float3 gMonoMult = {0.299f, 0.587f, 0.114f}; +const static float3 gNightVisionMult = {0.5f, 1.f, 0.5f}; + +// calculates luminance of pixel, then biases color balance toward green +void root(const uchar4 *v_in, uchar4 *v_out) { + v_out->x = v_in->x; // don't modify A + + // get RGB, scale 0-255 uchar to 0-1.0 float + float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f, + v_in->w * 0.003921569f}; + + // apply filter + float3 result = dot(rgb, gMonoMult) * gNightVisionMult; + + v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f); + v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f); + v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f); +} diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rs new file mode 100644 index 0000000..1de9dd8 --- /dev/null +++ b/cmds/stagefright/filters/saturation.rs @@ -0,0 +1,40 @@ +/* + * 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. + */ + +#pragma version(1) +#pragma rs java_package_name(com.android.rs.cppbasic) +#pragma rs_fp_relaxed + +const static float3 gMonoMult = {0.299f, 0.587f, 0.114f}; + +// global variables (parameters accessible to application code) +float gSaturation = 1.0f; + +void root(const uchar4 *v_in, uchar4 *v_out) { + v_out->x = v_in->x; // don't modify A + + // get RGB, scale 0-255 uchar to 0-1.0 float + float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f, + v_in->w * 0.003921569f}; + + // apply saturation filter + float3 result = dot(rgb, gMonoMult); + result = mix(result, rgb, gSaturation); + + v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f); + v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f); + v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f); +} diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp new file mode 100644 index 0000000..f77b38b --- /dev/null +++ b/cmds/stagefright/mediafilter.cpp @@ -0,0 +1,785 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "mediafilterTest" + +#include <inttypes.h> + +#include <binder/ProcessState.h> +#include <filters/ColorConvert.h> +#include <gui/ISurfaceComposer.h> +#include <gui/SurfaceComposerClient.h> +#include <gui/Surface.h> +#include <media/ICrypto.h> +#include <media/IMediaHTTPService.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/DataSource.h> +#include <media/stagefright/MediaCodec.h> +#include <media/stagefright/NuMediaExtractor.h> +#include <media/stagefright/RenderScriptWrapper.h> +#include <OMX_IVCommon.h> +#include <ui/DisplayInfo.h> + +#include "RenderScript.h" +#include "ScriptC_argbtorgba.h" +#include "ScriptC_nightvision.h" +#include "ScriptC_saturation.h" + +// test parameters +static const bool kTestFlush = true; // Note: true will drop 1 out of +static const int kFlushAfterFrames = 25; // kFlushAfterFrames output frames +static const int64_t kTimeout = 500ll; + +// built-in filter parameters +static const int32_t kInvert = false; // ZeroFilter param +static const float kBlurRadius = 15.0f; // IntrinsicBlurFilter param +static const float kSaturation = 0.0f; // SaturationFilter param + +static void usage(const char *me) { + fprintf(stderr, "usage: [flags] %s\n" + "\t[-b] use IntrinsicBlurFilter\n" + "\t[-c] use argb to rgba conversion RSFilter\n" + "\t[-n] use night vision RSFilter\n" + "\t[-r] use saturation RSFilter\n" + "\t[-s] use SaturationFilter\n" + "\t[-z] use ZeroFilter (copy filter)\n" + "\t[-R] render output to surface (enables -S)\n" + "\t[-S] allocate buffers from a surface\n" + "\t[-T] use render timestamps (enables -R)\n", + me); + exit(1); +} + +namespace android { + +struct SaturationRSFilter : RenderScriptWrapper::RSFilterCallback { + void init(RSC::sp<RSC::RS> context) { + mScript = new ScriptC_saturation(context); + mScript->set_gSaturation(3.f); + } + + virtual status_t processBuffers( + RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) { + mScript->forEach_root(inBuffer, outBuffer); + + return OK; + } + + status_t handleSetParameters(const sp<AMessage> &msg) { + return OK; + } + +private: + RSC::sp<ScriptC_saturation> mScript; +}; + +struct NightVisionRSFilter : RenderScriptWrapper::RSFilterCallback { + void init(RSC::sp<RSC::RS> context) { + mScript = new ScriptC_nightvision(context); + } + + virtual status_t processBuffers( + RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) { + mScript->forEach_root(inBuffer, outBuffer); + + return OK; + } + + status_t handleSetParameters(const sp<AMessage> &msg) { + return OK; + } + +private: + RSC::sp<ScriptC_nightvision> mScript; +}; + +struct ARGBToRGBARSFilter : RenderScriptWrapper::RSFilterCallback { + void init(RSC::sp<RSC::RS> context) { + mScript = new ScriptC_argbtorgba(context); + } + + virtual status_t processBuffers( + RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) { + mScript->forEach_root(inBuffer, outBuffer); + + return OK; + } + + status_t handleSetParameters(const sp<AMessage> &msg) { + return OK; + } + +private: + RSC::sp<ScriptC_argbtorgba> mScript; +}; + +struct CodecState { + sp<MediaCodec> mCodec; + Vector<sp<ABuffer> > mInBuffers; + Vector<sp<ABuffer> > mOutBuffers; + bool mSignalledInputEOS; + bool mSawOutputEOS; + int64_t mNumBuffersDecoded; +}; + +struct DecodedFrame { + size_t index; + size_t offset; + size_t size; + int64_t presentationTimeUs; + uint32_t flags; +}; + +enum FilterType { + FILTERTYPE_ZERO, + FILTERTYPE_INTRINSIC_BLUR, + FILTERTYPE_SATURATION, + FILTERTYPE_RS_SATURATION, + FILTERTYPE_RS_NIGHT_VISION, + FILTERTYPE_RS_ARGB_TO_RGBA, +}; + +size_t inputFramesSinceFlush = 0; +void tryCopyDecodedBuffer( + List<DecodedFrame> *decodedFrameIndices, + CodecState *filterState, + CodecState *vidState) { + if (decodedFrameIndices->empty()) { + return; + } + + size_t filterIndex; + status_t err = filterState->mCodec->dequeueInputBuffer( + &filterIndex, kTimeout); + if (err != OK) { + return; + } + + ++inputFramesSinceFlush; + + DecodedFrame frame = *decodedFrameIndices->begin(); + + // only consume a buffer if we are not going to flush, since we expect + // the dequeue -> flush -> queue operation to cause an error and + // not produce an output frame + if (!kTestFlush || inputFramesSinceFlush < kFlushAfterFrames) { + decodedFrameIndices->erase(decodedFrameIndices->begin()); + } + size_t outIndex = frame.index; + + const sp<ABuffer> &srcBuffer = + vidState->mOutBuffers.itemAt(outIndex); + const sp<ABuffer> &destBuffer = + filterState->mInBuffers.itemAt(filterIndex); + + sp<AMessage> srcFormat, destFormat; + vidState->mCodec->getOutputFormat(&srcFormat); + filterState->mCodec->getInputFormat(&destFormat); + + int32_t srcWidth, srcHeight, srcStride, srcSliceHeight; + int32_t srcColorFormat, destColorFormat; + int32_t destWidth, destHeight, destStride, destSliceHeight; + CHECK(srcFormat->findInt32("stride", &srcStride) + && srcFormat->findInt32("slice-height", &srcSliceHeight) + && srcFormat->findInt32("width", &srcWidth) + && srcFormat->findInt32("height", & srcHeight) + && srcFormat->findInt32("color-format", &srcColorFormat)); + CHECK(destFormat->findInt32("stride", &destStride) + && destFormat->findInt32("slice-height", &destSliceHeight) + && destFormat->findInt32("width", &destWidth) + && destFormat->findInt32("height", & destHeight) + && destFormat->findInt32("color-format", &destColorFormat)); + + CHECK(srcWidth <= destStride && srcHeight <= destSliceHeight); + + convertYUV420spToARGB( + srcBuffer->data(), + srcBuffer->data() + srcStride * srcSliceHeight, + srcWidth, + srcHeight, + destBuffer->data()); + + // copy timestamp + int64_t timeUs; + CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs)); + destBuffer->meta()->setInt64("timeUs", timeUs); + + if (kTestFlush && inputFramesSinceFlush >= kFlushAfterFrames) { + inputFramesSinceFlush = 0; + + // check that queueing a buffer that was dequeued before flush + // fails with expected error EACCES + filterState->mCodec->flush(); + + err = filterState->mCodec->queueInputBuffer( + filterIndex, 0 /* offset */, destBuffer->size(), + timeUs, frame.flags); + + if (err == OK) { + ALOGE("FAIL: queue after flush returned OK"); + } else if (err != -EACCES) { + ALOGE("queueInputBuffer after flush returned %d, " + "expected -EACCES (-13)", err); + } + } else { + err = filterState->mCodec->queueInputBuffer( + filterIndex, 0 /* offset */, destBuffer->size(), + timeUs, frame.flags); + CHECK(err == OK); + + err = vidState->mCodec->releaseOutputBuffer(outIndex); + CHECK(err == OK); + } +} + +size_t outputFramesSinceFlush = 0; +void tryDrainOutputBuffer( + CodecState *filterState, + const sp<Surface> &surface, bool renderSurface, + bool useTimestamp, int64_t *startTimeRender) { + size_t index; + size_t offset; + size_t size; + int64_t presentationTimeUs; + uint32_t flags; + status_t err = filterState->mCodec->dequeueOutputBuffer( + &index, &offset, &size, &presentationTimeUs, &flags, + kTimeout); + + if (err != OK) { + return; + } + + ++outputFramesSinceFlush; + + if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) { + filterState->mCodec->flush(); + } + + if (surface == NULL || !renderSurface) { + err = filterState->mCodec->releaseOutputBuffer(index); + } else if (useTimestamp) { + if (*startTimeRender == -1) { + // begin rendering 2 vsyncs after first decode + *startTimeRender = systemTime(SYSTEM_TIME_MONOTONIC) + + 33000000 - (presentationTimeUs * 1000); + } + presentationTimeUs = + (presentationTimeUs * 1000) + *startTimeRender; + err = filterState->mCodec->renderOutputBufferAndRelease( + index, presentationTimeUs); + } else { + err = filterState->mCodec->renderOutputBufferAndRelease(index); + } + + if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) { + outputFramesSinceFlush = 0; + + // releasing the buffer dequeued before flush should cause an error + // if so, the frame will also be skipped in output stream + if (err == OK) { + ALOGE("FAIL: release after flush returned OK"); + } else if (err != -EACCES) { + ALOGE("releaseOutputBuffer after flush returned %d, " + "expected -EACCES (-13)", err); + } + } else { + CHECK(err == OK); + } + + if (flags & MediaCodec::BUFFER_FLAG_EOS) { + ALOGV("reached EOS on output."); + filterState->mSawOutputEOS = true; + } +} + +static int decode( + const sp<ALooper> &looper, + const char *path, + const sp<Surface> &surface, + bool renderSurface, + bool useTimestamp, + FilterType filterType) { + + static int64_t kTimeout = 500ll; + + sp<NuMediaExtractor> extractor = new NuMediaExtractor; + if (extractor->setDataSource(NULL /* httpService */, path) != OK) { + fprintf(stderr, "unable to instantiate extractor.\n"); + return 1; + } + + KeyedVector<size_t, CodecState> stateByTrack; + + CodecState *vidState = NULL; + for (size_t i = 0; i < extractor->countTracks(); ++i) { + sp<AMessage> format; + status_t err = extractor->getTrackFormat(i, &format); + CHECK(err == OK); + + AString mime; + CHECK(format->findString("mime", &mime)); + bool isVideo = !strncasecmp(mime.c_str(), "video/", 6); + if (!isVideo) { + continue; + } + + ALOGV("selecting track %zu", i); + + err = extractor->selectTrack(i); + CHECK(err == OK); + + CodecState *state = + &stateByTrack.editValueAt(stateByTrack.add(i, CodecState())); + + vidState = state; + + state->mNumBuffersDecoded = 0; + + state->mCodec = MediaCodec::CreateByType( + looper, mime.c_str(), false /* encoder */); + + CHECK(state->mCodec != NULL); + + err = state->mCodec->configure( + format, NULL /* surface */, NULL /* crypto */, 0 /* flags */); + + CHECK(err == OK); + + state->mSignalledInputEOS = false; + state->mSawOutputEOS = false; + + break; + } + CHECK(!stateByTrack.isEmpty()); + CHECK(vidState != NULL); + sp<AMessage> vidFormat; + vidState->mCodec->getOutputFormat(&vidFormat); + + // set filter to use ARGB8888 + vidFormat->setInt32("color-format", OMX_COLOR_Format32bitARGB8888); + // set app cache directory path + vidFormat->setString("cacheDir", "/system/bin"); + + // create RenderScript context for RSFilters + RSC::sp<RSC::RS> context = new RSC::RS(); + context->init("/system/bin"); + + sp<RenderScriptWrapper::RSFilterCallback> rsFilter; + + // create renderscript wrapper for RSFilters + sp<RenderScriptWrapper> rsWrapper = new RenderScriptWrapper; + rsWrapper->mContext = context.get(); + + CodecState *filterState = new CodecState(); + filterState->mNumBuffersDecoded = 0; + + sp<AMessage> params = new AMessage(); + + switch (filterType) { + case FILTERTYPE_ZERO: + { + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.zerofilter"); + params->setInt32("invert", kInvert); + break; + } + case FILTERTYPE_INTRINSIC_BLUR: + { + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.intrinsicblur"); + params->setFloat("blur-radius", kBlurRadius); + break; + } + case FILTERTYPE_SATURATION: + { + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.saturation"); + params->setFloat("saturation", kSaturation); + break; + } + case FILTERTYPE_RS_SATURATION: + { + SaturationRSFilter *satFilter = new SaturationRSFilter; + satFilter->init(context); + rsFilter = satFilter; + rsWrapper->mCallback = rsFilter; + vidFormat->setObject("rs-wrapper", rsWrapper); + + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.RenderScript"); + break; + } + case FILTERTYPE_RS_NIGHT_VISION: + { + NightVisionRSFilter *nightVisionFilter = new NightVisionRSFilter; + nightVisionFilter->init(context); + rsFilter = nightVisionFilter; + rsWrapper->mCallback = rsFilter; + vidFormat->setObject("rs-wrapper", rsWrapper); + + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.RenderScript"); + break; + } + case FILTERTYPE_RS_ARGB_TO_RGBA: + { + ARGBToRGBARSFilter *argbToRgbaFilter = new ARGBToRGBARSFilter; + argbToRgbaFilter->init(context); + rsFilter = argbToRgbaFilter; + rsWrapper->mCallback = rsFilter; + vidFormat->setObject("rs-wrapper", rsWrapper); + + filterState->mCodec = MediaCodec::CreateByComponentName( + looper, "android.filter.RenderScript"); + break; + } + default: + { + LOG_ALWAYS_FATAL("mediacodec.cpp error: unrecognized FilterType"); + break; + } + } + CHECK(filterState->mCodec != NULL); + + status_t err = filterState->mCodec->configure( + vidFormat /* format */, surface, NULL /* crypto */, 0 /* flags */); + CHECK(err == OK); + + filterState->mSignalledInputEOS = false; + filterState->mSawOutputEOS = false; + + int64_t startTimeUs = ALooper::GetNowUs(); + int64_t startTimeRender = -1; + + for (size_t i = 0; i < stateByTrack.size(); ++i) { + CodecState *state = &stateByTrack.editValueAt(i); + + sp<MediaCodec> codec = state->mCodec; + + CHECK_EQ((status_t)OK, codec->start()); + + CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers)); + CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers)); + + ALOGV("got %zu input and %zu output buffers", + state->mInBuffers.size(), state->mOutBuffers.size()); + } + + CHECK_EQ((status_t)OK, filterState->mCodec->setParameters(params)); + + if (kTestFlush) { + status_t flushErr = filterState->mCodec->flush(); + if (flushErr == OK) { + ALOGE("FAIL: Flush before start returned OK"); + } else { + ALOGV("Flush before start returned status %d, usually ENOSYS (-38)", + flushErr); + } + } + + CHECK_EQ((status_t)OK, filterState->mCodec->start()); + CHECK_EQ((status_t)OK, filterState->mCodec->getInputBuffers( + &filterState->mInBuffers)); + CHECK_EQ((status_t)OK, filterState->mCodec->getOutputBuffers( + &filterState->mOutBuffers)); + + if (kTestFlush) { + status_t flushErr = filterState->mCodec->flush(); + if (flushErr != OK) { + ALOGE("FAIL: Flush after start returned %d, expect OK (0)", + flushErr); + } else { + ALOGV("Flush immediately after start OK"); + } + } + + List<DecodedFrame> decodedFrameIndices; + + // loop until decoder reaches EOS + bool sawInputEOS = false; + bool sawOutputEOSOnAllTracks = false; + while (!sawOutputEOSOnAllTracks) { + if (!sawInputEOS) { + size_t trackIndex; + status_t err = extractor->getSampleTrackIndex(&trackIndex); + + if (err != OK) { + ALOGV("saw input eos"); + sawInputEOS = true; + } else { + CodecState *state = &stateByTrack.editValueFor(trackIndex); + + size_t index; + err = state->mCodec->dequeueInputBuffer(&index, kTimeout); + + if (err == OK) { + ALOGV("filling input buffer %zu", index); + + const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index); + + err = extractor->readSampleData(buffer); + CHECK(err == OK); + + int64_t timeUs; + err = extractor->getSampleTime(&timeUs); + CHECK(err == OK); + + uint32_t bufferFlags = 0; + + err = state->mCodec->queueInputBuffer( + index, 0 /* offset */, buffer->size(), + timeUs, bufferFlags); + + CHECK(err == OK); + + extractor->advance(); + } else { + CHECK_EQ(err, -EAGAIN); + } + } + } else { + for (size_t i = 0; i < stateByTrack.size(); ++i) { + CodecState *state = &stateByTrack.editValueAt(i); + + if (!state->mSignalledInputEOS) { + size_t index; + status_t err = + state->mCodec->dequeueInputBuffer(&index, kTimeout); + + if (err == OK) { + ALOGV("signalling input EOS on track %zu", i); + + err = state->mCodec->queueInputBuffer( + index, 0 /* offset */, 0 /* size */, + 0ll /* timeUs */, MediaCodec::BUFFER_FLAG_EOS); + + CHECK(err == OK); + + state->mSignalledInputEOS = true; + } else { + CHECK_EQ(err, -EAGAIN); + } + } + } + } + + sawOutputEOSOnAllTracks = true; + for (size_t i = 0; i < stateByTrack.size(); ++i) { + CodecState *state = &stateByTrack.editValueAt(i); + + if (state->mSawOutputEOS) { + continue; + } else { + sawOutputEOSOnAllTracks = false; + } + + DecodedFrame frame; + status_t err = state->mCodec->dequeueOutputBuffer( + &frame.index, &frame.offset, &frame.size, + &frame.presentationTimeUs, &frame.flags, kTimeout); + + if (err == OK) { + ALOGV("draining decoded buffer %zu, time = %lld us", + frame.index, frame.presentationTimeUs); + + ++(state->mNumBuffersDecoded); + + decodedFrameIndices.push_back(frame); + + if (frame.flags & MediaCodec::BUFFER_FLAG_EOS) { + ALOGV("reached EOS on decoder output."); + state->mSawOutputEOS = true; + } + + } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) { + ALOGV("INFO_OUTPUT_BUFFERS_CHANGED"); + CHECK_EQ((status_t)OK, state->mCodec->getOutputBuffers( + &state->mOutBuffers)); + + ALOGV("got %zu output buffers", state->mOutBuffers.size()); + } else if (err == INFO_FORMAT_CHANGED) { + sp<AMessage> format; + CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format)); + + ALOGV("INFO_FORMAT_CHANGED: %s", + format->debugString().c_str()); + } else { + CHECK_EQ(err, -EAGAIN); + } + + tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState); + + tryDrainOutputBuffer( + filterState, surface, renderSurface, + useTimestamp, &startTimeRender); + } + } + + // after EOS on decoder, let filter reach EOS + while (!filterState->mSawOutputEOS) { + tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState); + + tryDrainOutputBuffer( + filterState, surface, renderSurface, + useTimestamp, &startTimeRender); + } + + int64_t elapsedTimeUs = ALooper::GetNowUs() - startTimeUs; + + for (size_t i = 0; i < stateByTrack.size(); ++i) { + CodecState *state = &stateByTrack.editValueAt(i); + + CHECK_EQ((status_t)OK, state->mCodec->release()); + + printf("track %zu: %" PRId64 " frames decoded and filtered, " + "%.2f fps.\n", i, state->mNumBuffersDecoded, + state->mNumBuffersDecoded * 1E6 / elapsedTimeUs); + } + + return 0; +} + +} // namespace android + +int main(int argc, char **argv) { + using namespace android; + + const char *me = argv[0]; + + bool useSurface = false; + bool renderSurface = false; + bool useTimestamp = false; + FilterType filterType = FILTERTYPE_ZERO; + + int res; + while ((res = getopt(argc, argv, "bcnrszTRSh")) >= 0) { + switch (res) { + case 'b': + { + filterType = FILTERTYPE_INTRINSIC_BLUR; + break; + } + case 'c': + { + filterType = FILTERTYPE_RS_ARGB_TO_RGBA; + break; + } + case 'n': + { + filterType = FILTERTYPE_RS_NIGHT_VISION; + break; + } + case 'r': + { + filterType = FILTERTYPE_RS_SATURATION; + break; + } + case 's': + { + filterType = FILTERTYPE_SATURATION; + break; + } + case 'z': + { + filterType = FILTERTYPE_ZERO; + break; + } + case 'T': + { + useTimestamp = true; + } + // fall through + case 'R': + { + renderSurface = true; + } + // fall through + case 'S': + { + useSurface = true; + break; + } + case '?': + case 'h': + default: + { + usage(me); + break; + } + } + } + + argc -= optind; + argv += optind; + + if (argc != 1) { + usage(me); + } + + ProcessState::self()->startThreadPool(); + + DataSource::RegisterDefaultSniffers(); + + android::sp<ALooper> looper = new ALooper; + looper->start(); + + android::sp<SurfaceComposerClient> composerClient; + android::sp<SurfaceControl> control; + android::sp<Surface> surface; + + if (useSurface) { + composerClient = new SurfaceComposerClient; + CHECK_EQ((status_t)OK, composerClient->initCheck()); + + android::sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay( + ISurfaceComposer::eDisplayIdMain)); + DisplayInfo info; + SurfaceComposerClient::getDisplayInfo(display, &info); + ssize_t displayWidth = info.w; + ssize_t displayHeight = info.h; + + ALOGV("display is %zd x %zd", displayWidth, displayHeight); + + control = composerClient->createSurface( + String8("A Surface"), displayWidth, displayHeight, + PIXEL_FORMAT_RGBA_8888, 0); + + CHECK(control != NULL); + CHECK(control->isValid()); + + SurfaceComposerClient::openGlobalTransaction(); + CHECK_EQ((status_t)OK, control->setLayer(INT_MAX)); + CHECK_EQ((status_t)OK, control->show()); + SurfaceComposerClient::closeGlobalTransaction(); + + surface = control->getSurface(); + CHECK(surface != NULL); + } + + decode(looper, argv[0], surface, renderSurface, useTimestamp, filterType); + + if (useSurface) { + composerClient->dispose(); + } + + looper->stop(); + + return 0; +} diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp index f4a33e8..461b56c 100644 --- a/cmds/stagefright/muxer.cpp +++ b/cmds/stagefright/muxer.cpp @@ -17,6 +17,9 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "muxer" #include <inttypes.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> #include <utils/Log.h> #include <binder/ProcessState.h> @@ -72,8 +75,15 @@ static int muxing( ALOGV("input file %s, output file %s", path, outputFileName); ALOGV("useAudio %d, useVideo %d", useAudio, useVideo); - sp<MediaMuxer> muxer = new MediaMuxer(outputFileName, + int fd = open(outputFileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + + if (fd < 0) { + ALOGE("couldn't open file"); + return fd; + } + sp<MediaMuxer> muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4); + close(fd); size_t trackCount = extractor->countTracks(); // Map the extractor's track index to the muxer's track index. diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp index 9f547c7..2ad40bd 100644 --- a/cmds/stagefright/recordvideo.cpp +++ b/cmds/stagefright/recordvideo.cpp @@ -17,6 +17,10 @@ #include "SineSource.h" #include <inttypes.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> + #include <binder/ProcessState.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/AudioPlayer.h> @@ -300,7 +304,13 @@ int main(int argc, char **argv) { client.interface(), enc_meta, true /* createEncoder */, source, 0, preferSoftwareCodec ? OMXCodec::kPreferSoftwareCodecs : 0); - sp<MPEG4Writer> writer = new MPEG4Writer(fileName); + int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + if (fd < 0) { + fprintf(stderr, "couldn't open file"); + return 1; + } + sp<MPEG4Writer> writer = new MPEG4Writer(fd); + close(fd); writer->addSource(encoder); int64_t start = systemTime(); CHECK_EQ((status_t)OK, writer->start()); diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp index 81edcb4..318b56d 100644 --- a/cmds/stagefright/stagefright.cpp +++ b/cmds/stagefright/stagefright.cpp @@ -19,6 +19,8 @@ #include <stdlib.h> #include <string.h> #include <sys/time.h> +#include <sys/types.h> +#include <sys/stat.h> //#define LOG_NDEBUG 0 #define LOG_TAG "stagefright" @@ -506,8 +508,13 @@ static void writeSourcesToMP4( sp<MPEG4Writer> writer = new MPEG4Writer(gWriteMP4Filename.string()); #else + int fd = open(gWriteMP4Filename.string(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); + if (fd < 0) { + fprintf(stderr, "couldn't open file"); + return; + } sp<MPEG2TSWriter> writer = - new MPEG2TSWriter(gWriteMP4Filename.string()); + new MPEG2TSWriter(fd); #endif // at most one minute. diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index fd51b8f..2e1ed6c 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -79,9 +79,7 @@ public: size_t size; // input/output in bytes == frameCount * frameSize // on input it is unused // on output is the number of bytes actually filled - // FIXME this is redundant with respect to frameCount, - // and TRANSFER_OBTAIN mode is broken for 8-bit data - // since we don't define the frame format + // FIXME this is redundant with respect to frameCount. union { void* raw; @@ -154,9 +152,9 @@ public: * streamType: Select the type of audio stream this track is attached to * (e.g. AUDIO_STREAM_MUSIC). * sampleRate: Data source sampling rate in Hz. - * format: Audio format. For mixed tracks, any PCM format supported by server is OK - * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct - * and offloaded tracks, the possible format(s) depends on the output sink. + * format: Audio format. For mixed tracks, any PCM format supported by server is OK. + * For direct and offloaded tracks, the possible format(s) depends on the + * output sink. * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. * frameCount: Minimum size of track PCM buffer in frames. This defines the * application's contribution to the @@ -193,7 +191,6 @@ public: /* Creates an audio track and registers it with AudioFlinger. * With this constructor, the track is configured for static buffer mode. - * The format must not be 8-bit linear PCM. * Data to be rendered is passed in a shared memory buffer * identified by the argument sharedBuffer, which must be non-0. * The memory should be initialized to the desired data before calling start(). @@ -614,6 +611,7 @@ protected: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void wake(); // wake to handle changed notification conditions. private: void pauseInternal(nsecs_t ns = 0LL); @@ -628,7 +626,9 @@ protected: bool mPaused; // whether thread is requested to pause at next loop entry bool mPausedInt; // whether thread internally requests pause nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored - bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request + bool mIgnoreNextPausedInt; // skip any internal pause and go immediately + // to processAudioBuffer() as state may have changed + // since pause time calculated. }; // body of AudioTrackThread::threadLoop() @@ -680,7 +680,7 @@ protected: float mVolume[2]; float mSendLevel; - mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. + mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it size_t mFrameCount; // corresponds to current IAudioTrack, value is // reported back by AudioFlinger to the client size_t mReqFrameCount; // frame count to request the first or next time @@ -698,10 +698,7 @@ protected: const audio_offload_info_t* mOffloadInfo; audio_attributes_t mAttributes; - // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's - // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. - size_t mFrameSize; // app-level frame size - size_t mFrameSizeAF; // AudioFlinger frame size + size_t mFrameSize; // frame size in bytes status_t mStatus; @@ -732,13 +729,20 @@ protected: bool mRefreshRemaining; // processAudioBuffer() should refresh // mRemainingFrames and mRetryOnPartialBuffer + // used for static track cbf and restoration + int32_t mLoopCount; // last setLoop loopCount; zero means disabled + uint32_t mLoopStart; // last setLoop loopStart + uint32_t mLoopEnd; // last setLoop loopEnd + int32_t mLoopCountNotified; // the last loopCount notified by callback. + // mLoopCountNotified counts down, matching + // the remaining loop count for static track + // playback. + // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence - uint32_t mLoopPeriod; // in frames, zero means looping is disabled - uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; uint32_t mNewPosition; // in frames diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h index b1ed7b0..64a3212 100644 --- a/include/media/EffectsFactoryApi.h +++ b/include/media/EffectsFactoryApi.h @@ -171,6 +171,8 @@ int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *p //////////////////////////////////////////////////////////////////////////////// int EffectIsNullUuid(const effect_uuid_t *pEffectUuid); +int EffectDumpEffects(int fd); + #if __cplusplus } // extern "C" #endif diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h index 67b599a..49a3d61 100644 --- a/include/media/IMediaPlayerService.h +++ b/include/media/IMediaPlayerService.h @@ -49,7 +49,8 @@ public: virtual sp<IMediaRecorder> createMediaRecorder() = 0; virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0; - virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0; + virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) + = 0; virtual sp<IOMX> getOMX() = 0; virtual sp<ICrypto> makeCrypto() = 0; diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h index 3e67550..509c06b 100644 --- a/include/media/IMediaRecorder.h +++ b/include/media/IMediaRecorder.h @@ -41,7 +41,6 @@ public: virtual status_t setOutputFormat(int of) = 0; virtual status_t setVideoEncoder(int ve) = 0; virtual status_t setAudioEncoder(int ae) = 0; - virtual status_t setOutputFile(const char* path) = 0; virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0; virtual status_t setVideoSize(int width, int height) = 0; virtual status_t setVideoFrameRate(int frames_per_second) = 0; diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h index d7ac302..f55063e 100644 --- a/include/media/MediaRecorderBase.h +++ b/include/media/MediaRecorderBase.h @@ -43,7 +43,6 @@ struct MediaRecorderBase { virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy) = 0; virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface) = 0; - virtual status_t setOutputFile(const char *path) = 0; virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0; virtual status_t setOutputFileAuxiliary(int fd) {return INVALID_OPERATION;} virtual status_t setParameters(const String8& params) = 0; diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h index 04c5fd0..d423962 100644 --- a/include/media/SingleStateQueue.h +++ b/include/media/SingleStateQueue.h @@ -21,6 +21,7 @@ // Non-blocking single-reader / single-writer multi-word atomic load / store #include <stdint.h> +#include <cutils/atomic.h> namespace android { @@ -31,6 +32,12 @@ public: class Mutator; class Observer; + enum SSQ_STATUS { + SSQ_PENDING, /* = 0 */ + SSQ_READ, + SSQ_DONE, + }; + struct Shared { // needs to be part of a union so don't define constructor or destructor @@ -41,28 +48,56 @@ private: void init() { mAck = 0; mSequence = 0; } volatile int32_t mAck; -#if 0 - int mPad[7]; - // cache line boundary -#endif volatile int32_t mSequence; T mValue; }; class Mutator { public: - Mutator(Shared *shared); - /*virtual*/ ~Mutator() { } + Mutator(Shared *shared) + : mSequence(0), mShared(shared) + { + // exactly one of Mutator and Observer must initialize, currently it is Observer + // shared->init(); + } // push new value onto state queue, overwriting previous value; // returns a sequence number which can be used with ack() - int32_t push(const T& value); - - // return true if most recent push has been observed - bool ack(); + int32_t push(const T& value) + { + Shared *shared = mShared; + int32_t sequence = mSequence; + sequence++; + android_atomic_acquire_store(sequence, &shared->mSequence); + shared->mValue = value; + sequence++; + android_atomic_release_store(sequence, &shared->mSequence); + mSequence = sequence; + // consider signalling a futex here, if we know that observer is waiting + return sequence; + } + + // returns the status of the last state push. This may be a stale value. + // + // SSQ_PENDING, or 0, means it has not been observed + // SSQ_READ means it has been read + // SSQ_DONE means it has been acted upon, after Observer::done() is called + enum SSQ_STATUS ack() const + { + // in the case of SSQ_DONE, prevent any subtle data-races of subsequent reads + // being performed (out-of-order) before the ack read, should the caller be + // depending on sequentiality of reads. + const int32_t ack = android_atomic_acquire_load(&mShared->mAck); + return ack - mSequence & ~1 ? SSQ_PENDING /* seq differ */ : + ack & 1 ? SSQ_DONE : SSQ_READ; + } // return true if a push with specified sequence number or later has been observed - bool ack(int32_t sequence); + bool ack(int32_t sequence) const + { + // this relies on 2's complement rollover to detect an ancient sequence number + return mShared->mAck - sequence >= 0; + } private: int32_t mSequence; @@ -71,11 +106,54 @@ private: class Observer { public: - Observer(Shared *shared); - /*virtual*/ ~Observer() { } + Observer(Shared *shared) + : mSequence(0), mSeed(1), mShared(shared) + { + // exactly one of Mutator and Observer must initialize, currently it is Observer + shared->init(); + } // return true if value has changed - bool poll(T& value); + bool poll(T& value) + { + Shared *shared = mShared; + int32_t before = shared->mSequence; + if (before == mSequence) { + return false; + } + for (int tries = 0; ; ) { + const int MAX_TRIES = 5; + if (before & 1) { + if (++tries >= MAX_TRIES) { + return false; + } + before = shared->mSequence; + } else { + android_memory_barrier(); + T temp = shared->mValue; + int32_t after = android_atomic_release_load(&shared->mSequence); + if (after == before) { + value = temp; + shared->mAck = before; + mSequence = before; // mSequence is even after poll success + return true; + } + if (++tries >= MAX_TRIES) { + return false; + } + before = after; + } + } + } + + // (optional) used to indicate to the Mutator that the state that has been polled + // has also been acted upon. + void done() + { + const int32_t ack = mShared->mAck + 1; + // ensure all previous writes have been performed. + android_atomic_release_store(ack, &mShared->mAck); // mSequence is odd after "done" + } private: int32_t mSequence; diff --git a/include/media/StringArray.h b/include/media/StringArray.h index ae47085..48d98bf 100644 --- a/include/media/StringArray.h +++ b/include/media/StringArray.h @@ -16,7 +16,7 @@ // // Sortable array of strings. STL-ish, but STL-free. -// +// #ifndef _LIBS_MEDIA_STRING_ARRAY_H #define _LIBS_MEDIA_STRING_ARRAY_H diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h index b0a62a7..74a6469 100644 --- a/include/media/mediarecorder.h +++ b/include/media/mediarecorder.h @@ -221,7 +221,6 @@ public: status_t setOutputFormat(int of); status_t setVideoEncoder(int ve); status_t setAudioEncoder(int ae); - status_t setOutputFile(const char* path); status_t setOutputFile(int fd, int64_t offset, int64_t length); status_t setVideoSize(int width, int height); status_t setVideoFrameRate(int frames_per_second); diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h index d422576..d9bbc8d 100644 --- a/include/media/nbaio/NBAIO.h +++ b/include/media/nbaio/NBAIO.h @@ -231,7 +231,8 @@ public: virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; } protected: - NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { } + NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) + { } virtual ~NBAIO_Sink() { } // Implementations are free to ignore these if they don't need them @@ -322,7 +323,8 @@ public: virtual void onTimestamp(const AudioTimestamp& timestamp) { } protected: - NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { } + NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) + { } virtual ~NBAIO_Source() { } // Implementations are free to ignore these if they don't need them diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h index bcbbc04..1297b51 100644 --- a/include/media/nbaio/NBLog.h +++ b/include/media/nbaio/NBLog.h @@ -21,7 +21,7 @@ #include <binder/IMemory.h> #include <utils/Mutex.h> -#include <media/nbaio/roundup.h> +#include <audio_utils/roundup.h> namespace android { diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h index d22707a..86417a5 100644 --- a/include/media/stagefright/AACWriter.h +++ b/include/media/stagefright/AACWriter.h @@ -27,7 +27,6 @@ struct MediaSource; struct MetaData; struct AACWriter : public MediaWriter { - AACWriter(const char *filename); AACWriter(int fd); status_t initCheck() const; diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h index 392f968..bac878b 100644 --- a/include/media/stagefright/AMRWriter.h +++ b/include/media/stagefright/AMRWriter.h @@ -29,7 +29,6 @@ struct MediaSource; struct MetaData; struct AMRWriter : public MediaWriter { - AMRWriter(const char *filename); AMRWriter(int fd); status_t initCheck() const; diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h index 2e2922e..3d7960b 100644 --- a/include/media/stagefright/MPEG2TSWriter.h +++ b/include/media/stagefright/MPEG2TSWriter.h @@ -29,7 +29,6 @@ struct ABuffer; struct MPEG2TSWriter : public MediaWriter { MPEG2TSWriter(int fd); - MPEG2TSWriter(const char *filename); MPEG2TSWriter( void *cookie, diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h index 26ce5f9..899b324 100644 --- a/include/media/stagefright/MPEG4Writer.h +++ b/include/media/stagefright/MPEG4Writer.h @@ -32,7 +32,6 @@ class MetaData; class MPEG4Writer : public MediaWriter { public: - MPEG4Writer(const char *filename); MPEG4Writer(int fd); // Limitations diff --git a/include/media/stagefright/MediaFilter.h b/include/media/stagefright/MediaFilter.h new file mode 100644 index 0000000..7b3f700 --- /dev/null +++ b/include/media/stagefright/MediaFilter.h @@ -0,0 +1,167 @@ +/* + * 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 MEDIA_FILTER_H_ +#define MEDIA_FILTER_H_ + +#include <media/stagefright/CodecBase.h> + +namespace android { + +struct ABuffer; +struct GraphicBufferListener; +struct MemoryDealer; +struct SimpleFilter; + +struct MediaFilter : public CodecBase { + MediaFilter(); + + virtual void setNotificationMessage(const sp<AMessage> &msg); + + virtual void initiateAllocateComponent(const sp<AMessage> &msg); + virtual void initiateConfigureComponent(const sp<AMessage> &msg); + virtual void initiateCreateInputSurface(); + virtual void initiateStart(); + virtual void initiateShutdown(bool keepComponentAllocated = false); + + virtual void signalFlush(); + virtual void signalResume(); + + virtual void signalRequestIDRFrame(); + virtual void signalSetParameters(const sp<AMessage> &msg); + virtual void signalEndOfInputStream(); + + virtual void onMessageReceived(const sp<AMessage> &msg); + + struct PortDescription : public CodecBase::PortDescription { + virtual size_t countBuffers(); + virtual IOMX::buffer_id bufferIDAt(size_t index) const; + virtual sp<ABuffer> bufferAt(size_t index) const; + + protected: + PortDescription(); + + private: + friend struct MediaFilter; + + Vector<IOMX::buffer_id> mBufferIDs; + Vector<sp<ABuffer> > mBuffers; + + void addBuffer(IOMX::buffer_id id, const sp<ABuffer> &buffer); + + DISALLOW_EVIL_CONSTRUCTORS(PortDescription); + }; + +protected: + virtual ~MediaFilter(); + +private: + struct BufferInfo { + enum Status { + OWNED_BY_US, + OWNED_BY_UPSTREAM, + }; + + IOMX::buffer_id mBufferID; + int32_t mGeneration; + int32_t mOutputFlags; + Status mStatus; + + sp<ABuffer> mData; + }; + + enum State { + UNINITIALIZED, + INITIALIZED, + CONFIGURED, + STARTED, + }; + + enum { + kWhatInputBufferFilled = 'inpF', + kWhatOutputBufferDrained = 'outD', + kWhatShutdown = 'shut', + kWhatFlush = 'flus', + kWhatResume = 'resm', + kWhatAllocateComponent = 'allo', + kWhatConfigureComponent = 'conf', + kWhatCreateInputSurface = 'cisf', + kWhatSignalEndOfInputStream = 'eois', + kWhatStart = 'star', + kWhatSetParameters = 'setP', + kWhatProcessBuffers = 'proc', + }; + + enum { + kPortIndexInput = 0, + kPortIndexOutput = 1 + }; + + // member variables + AString mComponentName; + State mState; + status_t mInputEOSResult; + int32_t mWidth, mHeight; + int32_t mStride, mSliceHeight; + int32_t mColorFormatIn, mColorFormatOut; + size_t mMaxInputSize, mMaxOutputSize; + int32_t mGeneration; + sp<AMessage> mNotify; + sp<AMessage> mInputFormat; + sp<AMessage> mOutputFormat; + + sp<MemoryDealer> mDealer[2]; + Vector<BufferInfo> mBuffers[2]; + Vector<BufferInfo*> mAvailableInputBuffers; + Vector<BufferInfo*> mAvailableOutputBuffers; + bool mPortEOS[2]; + + sp<SimpleFilter> mFilter; + sp<GraphicBufferListener> mGraphicBufferListener; + + // helper functions + void signalProcessBuffers(); + void signalError(status_t error); + + status_t allocateBuffersOnPort(OMX_U32 portIndex); + BufferInfo *findBufferByID( + uint32_t portIndex, IOMX::buffer_id bufferID, + ssize_t *index = NULL); + void postFillThisBuffer(BufferInfo *info); + void postDrainThisBuffer(BufferInfo *info); + void postEOS(); + void sendFormatChange(); + void requestFillEmptyInput(); + void processBuffers(); + + void onAllocateComponent(const sp<AMessage> &msg); + void onConfigureComponent(const sp<AMessage> &msg); + void onStart(); + void onInputBufferFilled(const sp<AMessage> &msg); + void onOutputBufferDrained(const sp<AMessage> &msg); + void onShutdown(const sp<AMessage> &msg); + void onFlush(); + void onSetParameters(const sp<AMessage> &msg); + void onCreateInputSurface(); + void onInputFrameAvailable(); + void onSignalEndOfInputStream(); + + DISALLOW_EVIL_CONSTRUCTORS(MediaFilter); +}; + +} // namespace android + +#endif // MEDIA_FILTER_H_ diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h index 9da98d9..e6538d1 100644 --- a/include/media/stagefright/MediaMuxer.h +++ b/include/media/stagefright/MediaMuxer.h @@ -50,9 +50,6 @@ public: OUTPUT_FORMAT_LIST_END // must be last - used to validate format type }; - // Construct the muxer with the output file path. - MediaMuxer(const char *path, OutputFormat format); - // Construct the muxer with the file descriptor. Note that the MediaMuxer // will close this file at stop(). MediaMuxer(int fd, OutputFormat format); diff --git a/include/media/stagefright/RenderScriptWrapper.h b/include/media/stagefright/RenderScriptWrapper.h new file mode 100644 index 0000000..b42649e --- /dev/null +++ b/include/media/stagefright/RenderScriptWrapper.h @@ -0,0 +1,42 @@ +/* + * 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 RENDERSCRIPT_WRAPPER_H_ +#define RENDERSCRIPT_WRAPPER_H_ + +#include <RenderScript.h> + +namespace android { + +struct RenderScriptWrapper : public RefBase { +public: + struct RSFilterCallback : public RefBase { + public: + // called by RSFilter to process each input buffer + virtual status_t processBuffers( + RSC::Allocation* inBuffer, + RSC::Allocation* outBuffer) = 0; + + virtual status_t handleSetParameters(const sp<AMessage> &msg) = 0; + }; + + sp<RSFilterCallback> mCallback; + RSC::sp<RSC::RS> mContext; +}; + +} // namespace android + +#endif // RENDERSCRIPT_WRAPPER_H_ diff --git a/include/ndk/NdkMediaCodec.h b/include/ndk/NdkMediaCodec.h index c07f4c9..4f6a1ef 100644 --- a/include/ndk/NdkMediaCodec.h +++ b/include/ndk/NdkMediaCodec.h @@ -142,7 +142,8 @@ media_status_t AMediaCodec_queueSecureInputBuffer(AMediaCodec*, /** * Get the index of the next available buffer of processed data. */ -ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info, int64_t timeoutUs); +ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info, + int64_t timeoutUs); AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*); /** diff --git a/include/ndk/NdkMediaExtractor.h b/include/ndk/NdkMediaExtractor.h index 7a4e702..7324d31 100644 --- a/include/ndk/NdkMediaExtractor.h +++ b/include/ndk/NdkMediaExtractor.h @@ -55,12 +55,14 @@ media_status_t AMediaExtractor_delete(AMediaExtractor*); /** * Set the file descriptor from which the extractor will read. */ -media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset, off64_t length); +media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset, + off64_t length); /** * Set the URI from which the extractor will read. */ -media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location); // TODO support headers +media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location); + // TODO support headers /** * Return the number of tracks in the previously specified media file diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 31dff36..7143f1a 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -24,9 +24,8 @@ #include <utils/threads.h> #include <utils/Log.h> #include <utils/RefBase.h> -#include <media/nbaio/roundup.h> +#include <audio_utils/roundup.h> #include <media/SingleStateQueue.h> -#include <private/media/StaticAudioTrackState.h> namespace android { @@ -61,15 +60,57 @@ struct AudioTrackSharedStreaming { volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame }; +// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer +// supplied by the client). This state needs to be communicated from the client to server. As this +// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the +// state is wrapped by a SingleStateQueue. +struct StaticAudioTrackState { + // Do not define constructors, destructors, or virtual methods as this is part of a + // union in shared memory and they will not get called properly. + + // These fields should both be size_t, but since they are located in shared memory we + // force to 32-bit. The client and server may have different typedefs for size_t. + + // The state has a sequence counter to indicate whether changes are made to loop or position. + // The sequence counter also currently indicates whether loop or position is first depending + // on which is greater; it jumps by max(mLoopSequence, mPositionSequence) + 1. + + uint32_t mLoopStart; + uint32_t mLoopEnd; + int32_t mLoopCount; + uint32_t mLoopSequence; // a sequence counter to indicate changes to loop + uint32_t mPosition; + uint32_t mPositionSequence; // a sequence counter to indicate changes to position +}; + typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue; +struct StaticAudioTrackPosLoop { + // Do not define constructors, destructors, or virtual methods as this is part of a + // union in shared memory and will not get called properly. + + // These fields should both be size_t, but since they are located in shared memory we + // force to 32-bit. The client and server may have different typedefs for size_t. + + // This struct information is stored in a single state queue to communicate the + // static AudioTrack server state to the client while data is consumed. + // It is smaller than StaticAudioTrackState to prevent unnecessary information from + // being sent. + + uint32_t mBufferPosition; + int32_t mLoopCount; +}; + +typedef SingleStateQueue<StaticAudioTrackPosLoop> StaticAudioTrackPosLoopQueue; + struct AudioTrackSharedStatic { + // client requests to the server for loop or position changes. StaticAudioTrackSingleStateQueue::Shared mSingleStateQueue; - // This field should be a size_t, but since it is located in shared memory we - // force to 32-bit. The client and server may have different typedefs for size_t. - uint32_t mBufferPosition; // updated asynchronously by server, - // "for entertainment purposes only" + // position info updated asynchronously by server and read by client, + // "for entertainment purposes only" + StaticAudioTrackPosLoopQueue::Shared + mPosLoopQueue; }; // ---------------------------------------------------------------------------- @@ -96,7 +137,8 @@ struct audio_track_cblk_t uint32_t mServer; // Number of filled frames consumed by server (mIsOut), // or filled frames provided by server (!mIsOut). // It is updated asynchronously by server without a barrier. - // The value should be used "for entertainment purposes only", + // The value should be used + // "for entertainment purposes only", // which means don't make important decisions based on it. uint32_t mPad1; // unused @@ -313,8 +355,28 @@ public: virtual void flush(); #define MIN_LOOP 16 // minimum length of each loop iteration in frames + + // setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the + // static buffer position and looping parameters. These commands are not + // synchronous (they do not wait or block); instead they take effect at the + // next buffer data read from the server side. However, the client side + // getters will read a cached version of the position and loop variables + // until the setting takes effect. + // + // setBufferPositionAndLoop() is equivalent to calling, in order, setLoop() and + // setBufferPosition(). + // + // The functions should not be relied upon to do parameter or state checking. + // That is done at the AudioTrack level. + void setLoop(size_t loopStart, size_t loopEnd, int loopCount); + void setBufferPosition(size_t position); + void setBufferPositionAndLoop(size_t position, size_t loopStart, size_t loopEnd, + int loopCount); size_t getBufferPosition(); + // getBufferPositionAndLoopCount() provides the proper snapshot of + // position and loopCount together. + void getBufferPositionAndLoopCount(size_t *position, int *loopCount); virtual size_t getMisalignment() { return 0; @@ -326,7 +388,9 @@ public: private: StaticAudioTrackSingleStateQueue::Mutator mMutator; - size_t mBufferPosition; // so that getBufferPosition() appears to be synchronous + StaticAudioTrackPosLoopQueue::Observer mPosLoopObserver; + StaticAudioTrackState mState; // last communicated state to server + StaticAudioTrackPosLoop mPosLoop; // snapshot of position and loop. }; // ---------------------------------------------------------------------------- @@ -447,10 +511,13 @@ public: virtual uint32_t getUnderrunFrames() const { return 0; } private: + status_t updateStateWithLoop(StaticAudioTrackState *localState, + const StaticAudioTrackState &update) const; + status_t updateStateWithPosition(StaticAudioTrackState *localState, + const StaticAudioTrackState &update) const; ssize_t pollPosition(); // poll for state queue update, and return current position StaticAudioTrackSingleStateQueue::Observer mObserver; - size_t mPosition; // server's current play position in frames, relative to 0 - + StaticAudioTrackPosLoopQueue::Mutator mPosLoopMutator; size_t mFramesReadySafe; // Assuming size_t read/writes are atomic on 32 / 64 bit // processors, this is a thread-safe version of // mFramesReady. @@ -459,7 +526,8 @@ private: // can cause a track to appear to have a large number // of frames. INT64_MAX means an infinite loop. bool mFramesReadyIsCalledByMultipleThreads; - StaticAudioTrackState mState; + StaticAudioTrackState mState; // Server side state. Any updates from client must be + // passed by the mObserver SingleStateQueue. }; // Proxy used by AudioFlinger for servicing AudioRecord diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h deleted file mode 100644 index d483061..0000000 --- a/include/private/media/StaticAudioTrackState.h +++ /dev/null @@ -1,39 +0,0 @@ -/* - * Copyright (C) 2012 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 STATIC_AUDIO_TRACK_STATE_H -#define STATIC_AUDIO_TRACK_STATE_H - -namespace android { - -// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer -// supplied by the client). This state needs to be communicated from the client to server. As this -// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the -// state is wrapped by a SingleStateQueue. -struct StaticAudioTrackState { - // do not define constructors, destructors, or virtual methods - - // These fields should both be size_t, but since they are located in shared memory we - // force to 32-bit. The client and server may have different typedefs for size_t. - uint32_t mLoopStart; - uint32_t mLoopEnd; - - int mLoopCount; -}; - -} // namespace android - -#endif // STATIC_AUDIO_TRACK_STATE_H diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c index 6d30d64..c310fe2 100644 --- a/media/libeffects/factory/EffectsFactory.c +++ b/media/libeffects/factory/EffectsFactory.c @@ -28,6 +28,7 @@ static list_elem_t *gEffectList; // list of effect_entry_t: all currently created effects static list_elem_t *gLibraryList; // list of lib_entry_t: all currently loaded libraries +static list_elem_t *gSkippedEffects; // list of effects skipped because of duplicate uuid // list of effect_descriptor and list of sub effects : all currently loaded // It does not contain effects without sub effects. static list_sub_elem_t *gSubEffectList; @@ -63,10 +64,10 @@ static int findEffect(const effect_uuid_t *type, lib_entry_t **lib, effect_descriptor_t **desc); // To search a subeffect in the gSubEffectList -int findSubEffect(const effect_uuid_t *uuid, +static int findSubEffect(const effect_uuid_t *uuid, lib_entry_t **lib, effect_descriptor_t **desc); -static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len); +static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent); static int stringToUuid(const char *str, effect_uuid_t *uuid); static int uuidToString(const effect_uuid_t *uuid, char *str, size_t maxLen); @@ -237,8 +238,8 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) } #if (LOG_NDEBUG == 0) - char str[256]; - dumpEffectDescriptor(pDescriptor, str, 256); + char str[512]; + dumpEffectDescriptor(pDescriptor, str, sizeof(str), 0 /* indent */); ALOGV("EffectQueryEffect() desc:%s", str); #endif pthread_mutex_unlock(&gLibLock); @@ -503,15 +504,31 @@ int loadLibrary(cnode *root, const char *name) audio_effect_library_t *desc; list_elem_t *e; lib_entry_t *l; + char path[PATH_MAX]; + char *str; + size_t len; node = config_find(root, PATH_TAG); if (node == NULL) { return -EINVAL; } + // audio_effects.conf always specifies 32 bit lib path: convert to 64 bit path if needed + strlcpy(path, node->value, PATH_MAX); +#ifdef __LP64__ + str = strstr(path, "/lib/"); + if (str == NULL) + return -EINVAL; + len = str - path; + path[len] = '\0'; + strlcat(path, "/lib64/", PATH_MAX); + strlcat(path, node->value + len + strlen("/lib/"), PATH_MAX); +#endif + if (strlen(path) >= PATH_MAX - 1) + return -EINVAL; - hdl = dlopen(node->value, RTLD_NOW); + hdl = dlopen(path, RTLD_NOW); if (hdl == NULL) { - ALOGW("loadLibrary() failed to open %s", node->value); + ALOGW("loadLibrary() failed to open %s", path); goto error; } @@ -535,7 +552,7 @@ int loadLibrary(cnode *root, const char *name) // add entry for library in gLibraryList l = malloc(sizeof(lib_entry_t)); l->name = strndup(name, PATH_MAX); - l->path = strndup(node->value, PATH_MAX); + l->path = strndup(path, PATH_MAX); l->handle = hdl; l->desc = desc; l->effects = NULL; @@ -547,7 +564,7 @@ int loadLibrary(cnode *root, const char *name) e->next = gLibraryList; gLibraryList = e; pthread_mutex_unlock(&gLibLock); - ALOGV("getLibrary() linked library %p for path %s", l, node->value); + ALOGV("getLibrary() linked library %p for path %s", l, path); return 0; @@ -595,8 +612,8 @@ int addSubEffect(cnode *root) return -EINVAL; } #if (LOG_NDEBUG==0) - char s[256]; - dumpEffectDescriptor(d, s, 256); + char s[512]; + dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */); ALOGV("addSubEffect() read descriptor %p:%s",d, s); #endif if (EFFECT_API_VERSION_MAJOR(d->apiVersion) != @@ -660,6 +677,13 @@ int loadEffect(cnode *root) ALOGW("loadEffect() invalid uuid %s", node->value); return -EINVAL; } + lib_entry_t *tmp; + bool skip = false; + if (findEffect(NULL, &uuid, &tmp, NULL) == 0) { + ALOGW("skipping duplicate uuid %s %s", node->value, + node->next ? "and its sub-effects" : ""); + skip = true; + } d = malloc(sizeof(effect_descriptor_t)); if (l->desc->get_descriptor(&uuid, d) != 0) { @@ -670,8 +694,8 @@ int loadEffect(cnode *root) return -EINVAL; } #if (LOG_NDEBUG==0) - char s[256]; - dumpEffectDescriptor(d, s, 256); + char s[512]; + dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */); ALOGV("loadEffect() read descriptor %p:%s",d, s); #endif if (EFFECT_API_VERSION_MAJOR(d->apiVersion) != @@ -682,8 +706,14 @@ int loadEffect(cnode *root) } e = malloc(sizeof(list_elem_t)); e->object = d; - e->next = l->effects; - l->effects = e; + if (skip) { + e->next = gSkippedEffects; + gSkippedEffects = e; + return -EINVAL; + } else { + e->next = l->effects; + l->effects = e; + } // After the UUID node in the config_tree, if node->next is valid, // that would be sub effect node. @@ -876,22 +906,30 @@ int findEffect(const effect_uuid_t *type, return ret; } -void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len) { +void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent) { char s[256]; + char ss[256]; + char idt[indent + 1]; - snprintf(str, len, "\nEffect Descriptor %p:\n", desc); - strncat(str, "- TYPE: ", len); - uuidToString(&desc->uuid, s, 256); - snprintf(str, len, "- UUID: %s\n", s); - uuidToString(&desc->type, s, 256); - snprintf(str, len, "- TYPE: %s\n", s); - sprintf(s, "- apiVersion: %08X\n- flags: %08X\n", - desc->apiVersion, desc->flags); - strncat(str, s, len); - sprintf(s, "- name: %s\n", desc->name); - strncat(str, s, len); - sprintf(s, "- implementor: %s\n", desc->implementor); - strncat(str, s, len); + memset(idt, ' ', indent); + idt[indent] = 0; + + str[0] = 0; + + snprintf(s, sizeof(s), "%s%s / %s\n", idt, desc->name, desc->implementor); + strlcat(str, s, len); + + uuidToString(&desc->uuid, s, sizeof(s)); + snprintf(ss, sizeof(ss), "%s UUID: %s\n", idt, s); + strlcat(str, ss, len); + + uuidToString(&desc->type, s, sizeof(s)); + snprintf(ss, sizeof(ss), "%s TYPE: %s\n", idt, s); + strlcat(str, ss, len); + + sprintf(s, "%s apiVersion: %08X\n%s flags: %08X\n", idt, + desc->apiVersion, idt, desc->flags); + strlcat(str, s, len); } int stringToUuid(const char *str, effect_uuid_t *uuid) @@ -934,3 +972,40 @@ int uuidToString(const effect_uuid_t *uuid, char *str, size_t maxLen) return 0; } +int EffectDumpEffects(int fd) { + char s[512]; + list_elem_t *e = gLibraryList; + lib_entry_t *l = NULL; + effect_descriptor_t *d = NULL; + int found = 0; + int ret = 0; + + while (e) { + l = (lib_entry_t *)e->object; + list_elem_t *efx = l->effects; + dprintf(fd, "Library %s\n", l->name); + if (!efx) { + dprintf(fd, " (no effects)\n"); + } + while (efx) { + d = (effect_descriptor_t *)efx->object; + dumpEffectDescriptor(d, s, sizeof(s), 2); + dprintf(fd, "%s", s); + efx = efx->next; + } + e = e->next; + } + + e = gSkippedEffects; + if (e) { + dprintf(fd, "Skipped effects\n"); + while(e) { + d = (effect_descriptor_t *)e->object; + dumpEffectDescriptor(d, s, sizeof(s), 2 /* indent */); + dprintf(fd, "%s", s); + e = e->next; + } + } + return ret; +} + diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk index 6c585fb..5378bf2 100644 --- a/media/libmedia/Android.mk +++ b/media/libmedia/Android.mk @@ -61,15 +61,11 @@ LOCAL_SRC_FILES:= \ StringArray.cpp \ AudioPolicy.cpp -LOCAL_SRC_FILES += ../libnbaio/roundup.c - LOCAL_SHARED_LIBRARIES := \ libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \ libcamera_client libstagefright_foundation \ libgui libdl libaudioutils libnbaio -LOCAL_STATIC_LIBRARIES += libinstantssq - LOCAL_WHOLE_STATIC_LIBRARIES := libmedia_helper LOCAL_MODULE:= libmedia @@ -85,12 +81,3 @@ LOCAL_C_INCLUDES := \ include $(BUILD_SHARED_LIBRARY) -include $(CLEAR_VARS) - -LOCAL_SRC_FILES += SingleStateQueue.cpp -LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"' - -LOCAL_MODULE := libinstantssq -LOCAL_MODULE_TAGS := optional - -include $(BUILD_STATIC_LIBRARY) diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 9cae21c..7a18cba 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -499,8 +499,8 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle OutputDescriptor *outputDesc = new OutputDescriptor(*desc); gOutputs.add(ioHandle, outputDesc); - ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu " - "latency %d", + ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x " + "frameCount %zu latency %d", outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask, outputDesc->frameCount, outputDesc->latency); } break; @@ -523,8 +523,8 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle if (param2 == NULL) break; desc = (const OutputDescriptor *)param2; - ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x " - "frameCount %zu latency %d", + ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x " + "channel mask %#x frameCount %zu latency %d", ioHandle, desc->samplingRate, desc->format, desc->channelMask, desc->frameCount, desc->latency); OutputDescriptor *outputDesc = gOutputs.valueAt(index); diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 735db5c..d4bacc0 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -33,11 +33,16 @@ #define WAIT_PERIOD_MS 10 #define WAIT_STREAM_END_TIMEOUT_SEC 120 - +static const int kMaxLoopCountNotifications = 32; namespace android { // --------------------------------------------------------------------------- +template <typename T> +const T &min(const T &x, const T &y) { + return x < y ? x : y; +} + static int64_t convertTimespecToUs(const struct timespec &tv) { return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; @@ -317,12 +322,6 @@ status_t AudioTrack::set( uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); mChannelCount = channelCount; - // AudioFlinger does not currently support 8-bit data in shared memory - if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { - ALOGE("8-bit data in shared memory is not supported"); - return BAD_VALUE; - } - // force direct flag if format is not linear PCM // or offload was requested if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) @@ -346,12 +345,9 @@ status_t AudioTrack::set( } else { mFrameSize = sizeof(uint8_t); } - mFrameSizeAF = mFrameSize; } else { ALOG_ASSERT(audio_is_linear_pcm(format)); mFrameSize = channelCount * audio_bytes_per_sample(format); - mFrameSizeAF = channelCount * audio_bytes_per_sample( - format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); // createTrack will return an error if PCM format is not supported by server, // so no need to check for specific PCM formats here } @@ -420,7 +416,10 @@ status_t AudioTrack::set( mStatus = NO_ERROR; mState = STATE_STOPPED; mUserData = user; - mLoopPeriod = 0; + mLoopCount = 0; + mLoopStart = 0; + mLoopEnd = 0; + mLoopCountNotified = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; @@ -531,14 +530,12 @@ void AudioTrack::stop() // the playback head position will reset to 0, so if a marker is set, we need // to activate it again mMarkerReached = false; -#if 0 - // Force flush if a shared buffer is used otherwise audioflinger - // will not stop before end of buffer is reached. - // It may be needed to make sure that we stop playback, likely in case looping is on. + if (mSharedBuffer != 0) { - flush_l(); + // clear buffer position and loop count. + mStaticProxy->setBufferPositionAndLoop(0 /* position */, + 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); } -#endif sp<AudioTrackThread> t = mAudioTrackThread; if (t != 0) { @@ -740,10 +737,15 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) { - // Setting the loop will reset next notification update period (like setPosition). - mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; - mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; + // We do not update the periodic notification point. + // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; + mLoopCount = loopCount; + mLoopEnd = loopEnd; + mLoopStart = loopStart; + mLoopCountNotified = loopCount; mStaticProxy->setLoop(loopStart, loopEnd, loopCount); + + // Waking the AudioTrackThread is not needed as this cannot be called when active. } status_t AudioTrack::setMarkerPosition(uint32_t marker) @@ -757,6 +759,10 @@ status_t AudioTrack::setMarkerPosition(uint32_t marker) mMarkerPosition = marker; mMarkerReached = false; + sp<AudioTrackThread> t = mAudioTrackThread; + if (t != 0) { + t->wake(); + } return NO_ERROR; } @@ -786,6 +792,10 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) mNewPosition = updateAndGetPosition_l() + updatePeriod; mUpdatePeriod = updatePeriod; + sp<AudioTrackThread> t = mAudioTrackThread; + if (t != 0) { + t->wake(); + } return NO_ERROR; } @@ -823,12 +833,11 @@ status_t AudioTrack::setPosition(uint32_t position) if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } + // After setting the position, use full update period before notification. mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; - mLoopPeriod = 0; - // FIXME Check whether loops and setting position are incompatible in old code. - // If we use setLoop for both purposes we lose the capability to set the position while looping. - mStaticProxy->setLoop(position, mFrameCount, 0); + mStaticProxy->setBufferPosition(position); + // Waking the AudioTrackThread is not needed as this cannot be called when active. return NO_ERROR; } @@ -893,10 +902,18 @@ status_t AudioTrack::reload() return INVALID_OPERATION; } mNewPosition = mUpdatePeriod; - mLoopPeriod = 0; - // FIXME The new code cannot reload while keeping a loop specified. - // Need to check how the old code handled this, and whether it's a significant change. - mStaticProxy->setLoop(0, mFrameCount, 0); + (void) updateAndGetPosition_l(); + mPosition = 0; +#if 0 + // The documentation is not clear on the behavior of reload() and the restoration + // of loop count. Historically we have not restored loop count, start, end, + // but it makes sense if one desires to repeat playing a particular sound. + if (mLoopCount != 0) { + mLoopCountNotified = mLoopCount; + mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); + } +#endif + mStaticProxy->setBufferPosition(0); return NO_ERROR; } @@ -1019,12 +1036,12 @@ status_t AudioTrack::createTrack_l() mNotificationFramesAct = frameCount; } } else if (mSharedBuffer != 0) { - - // Ensure that buffer alignment matches channel count - // 8-bit data in shared memory is not currently supported by AudioFlinger - size_t alignment = audio_bytes_per_sample( - mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); + // FIXME: Ensure client side memory buffers need + // not have additional alignment beyond sample + // (e.g. 16 bit stereo accessed as 32 bit frame). + size_t alignment = audio_bytes_per_sample(mFormat); if (alignment & 1) { + // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). alignment = 1; } if (mChannelCount > 1) { @@ -1042,7 +1059,7 @@ status_t AudioTrack::createTrack_l() // there's no frameCount parameter. // But when initializing a shared buffer AudioTrack via set(), // there _is_ a frameCount parameter. We silently ignore it. - frameCount = mSharedBuffer->size() / mFrameSizeAF; + frameCount = mSharedBuffer->size() / mFrameSize; } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { @@ -1103,10 +1120,7 @@ status_t AudioTrack::createTrack_l() // but we will still need the original value also sp<IAudioTrack> track = audioFlinger->createTrack(streamType, mSampleRate, - // AudioFlinger only sees 16-bit PCM - mFormat == AUDIO_FORMAT_PCM_8_BIT && - !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? - AUDIO_FORMAT_PCM_16_BIT : mFormat, + mFormat, mChannelMask, &temp, &trackFlags, @@ -1230,9 +1244,9 @@ status_t AudioTrack::createTrack_l() // update proxy if (mSharedBuffer == 0) { mStaticProxy.clear(); - mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); } else { - mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); mProxy = mStaticProxy; } @@ -1352,7 +1366,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *re } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); audioBuffer->frameCount = buffer.mFrameCount; - audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; + audioBuffer->size = buffer.mFrameCount * mFrameSize; audioBuffer->raw = buffer.mRaw; if (nonContig != NULL) { *nonContig = buffer.mNonContig; @@ -1366,7 +1380,7 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer) return; } - size_t stepCount = audioBuffer->size / mFrameSizeAF; + size_t stepCount = audioBuffer->size / mFrameSize; if (stepCount == 0) { return; } @@ -1432,14 +1446,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) } size_t toWrite; - if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { - // Divide capacity by 2 to take expansion into account - toWrite = audioBuffer.size >> 1; - memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); - } else { - toWrite = audioBuffer.size; - memcpy(audioBuffer.i8, buffer, toWrite); - } + toWrite = audioBuffer.size; + memcpy(audioBuffer.i8, buffer, toWrite); buffer = ((const char *) buffer) + toWrite; userSize -= toWrite; written += toWrite; @@ -1559,9 +1567,8 @@ nsecs_t AudioTrack::processAudioBuffer() // that the upper layers can recreate the track if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { status_t status = restoreTrack_l("processAudioBuffer"); - mLock.unlock(); - // Run again immediately, but with a new IAudioTrack - return 0; + // after restoration, continue below to make sure that the loop and buffer events + // are notified because they have been cleared from mCblk->mFlags above. } } @@ -1610,7 +1617,6 @@ nsecs_t AudioTrack::processAudioBuffer() } // Cache other fields that will be needed soon - uint32_t loopPeriod = mLoopPeriod; uint32_t sampleRate = mSampleRate; uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { @@ -1622,8 +1628,30 @@ nsecs_t AudioTrack::processAudioBuffer() uint32_t sequence = mSequence; sp<AudioTrackClientProxy> proxy = mProxy; + // Determine the number of new loop callback(s) that will be needed, while locked. + int loopCountNotifications = 0; + uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END + + if (mLoopCount > 0) { + int loopCount; + size_t bufferPosition; + mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); + loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; + loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); + mLoopCountNotified = loopCount; // discard any excess notifications + } else if (mLoopCount < 0) { + // FIXME: We're not accurate with notification count and position with infinite looping + // since loopCount from server side will always return -1 (we could decrement it). + size_t bufferPosition = mStaticProxy->getBufferPosition(); + loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); + loopPeriod = mLoopEnd - bufferPosition; + } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { + size_t bufferPosition = mStaticProxy->getBufferPosition(); + loopPeriod = mFrameCount - bufferPosition; + } + // These fields don't need to be cached, because they are assigned only by set(): - // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags + // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags // mFlags is also assigned by createTrack_l(), but not the bit we care about. mLock.unlock(); @@ -1662,10 +1690,9 @@ nsecs_t AudioTrack::processAudioBuffer() if (newUnderrun) { mCbf(EVENT_UNDERRUN, mUserData, NULL); } - // FIXME we will miss loops if loop cycle was signaled several times since last call - // to processAudioBuffer() - if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { + while (loopCountNotifications > 0) { mCbf(EVENT_LOOP_END, mUserData, NULL); + --loopCountNotifications; } if (flags & CBLK_BUFFER_END) { mCbf(EVENT_BUFFER_END, mUserData, NULL); @@ -1701,10 +1728,11 @@ nsecs_t AudioTrack::processAudioBuffer() minFrames = markerPosition - position; } if (loopPeriod > 0 && loopPeriod < minFrames) { + // loopPeriod is already adjusted for actual position. minFrames = loopPeriod; } - if (updatePeriod > 0 && updatePeriod < minFrames) { - minFrames = updatePeriod; + if (updatePeriod > 0) { + minFrames = min(minFrames, uint32_t(newPosition - position)); } // If > 0, poll periodically to recover from a stuck server. A good value is 2. @@ -1767,13 +1795,6 @@ nsecs_t AudioTrack::processAudioBuffer() } } - // Divide buffer size by 2 to take into account the expansion - // due to 8 to 16 bit conversion: the callback must fill only half - // of the destination buffer - if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { - audioBuffer.size >>= 1; - } - size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); size_t writtenSize = audioBuffer.size; @@ -1793,13 +1814,7 @@ nsecs_t AudioTrack::processAudioBuffer() return WAIT_PERIOD_MS * 1000000LL; } - if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { - // 8 to 16 bit conversion, note that source and destination are the same address - memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); - audioBuffer.size <<= 1; - } - - size_t releasedFrames = audioBuffer.size / mFrameSizeAF; + size_t releasedFrames = audioBuffer.size / mFrameSize; audioBuffer.frameCount = releasedFrames; mRemainingFrames -= releasedFrames; if (misalignment >= releasedFrames) { @@ -1856,7 +1871,11 @@ status_t AudioTrack::restoreTrack_l(const char *from) } // save the old static buffer position - size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; + size_t bufferPosition = 0; + int loopCount = 0; + if (mStaticProxy != 0) { + mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); + } // If a new IAudioTrack is successfully created, createTrack_l() will modify the // following member variables: mAudioTrack, mCblkMemory and mCblk. @@ -1865,30 +1884,26 @@ status_t AudioTrack::restoreTrack_l(const char *from) result = createTrack_l(); // take the frames that will be lost by track recreation into account in saved position + // For streaming tracks, this is the amount we obtained from the user/client + // (not the number actually consumed at the server - those are already lost). (void) updateAndGetPosition_l(); - mPosition = mReleased; + if (mStaticProxy != 0) { + mPosition = mReleased; + } if (result == NO_ERROR) { - // continue playback from last known position, but - // don't attempt to restore loop after invalidation; it's difficult and not worthwhile - if (mStaticProxy != NULL) { - mLoopPeriod = 0; - mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); - } - // FIXME How do we simulate the fact that all frames present in the buffer at the time of - // track destruction have been played? This is critical for SoundPool implementation - // This must be broken, and needs to be tested/debugged. -#if 0 - // restore write index and set other indexes to reflect empty buffer status - if (!strcmp(from, "start")) { - // Make sure that a client relying on callback events indicating underrun or - // the actual amount of audio frames played (e.g SoundPool) receives them. - if (mSharedBuffer == 0) { - // restart playback even if buffer is not completely filled. - android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); + // Continue playback from last known position and restore loop. + if (mStaticProxy != 0) { + if (loopCount != 0) { + mStaticProxy->setBufferPositionAndLoop(bufferPosition, + mLoopStart, mLoopEnd, loopCount); + } else { + mStaticProxy->setBufferPosition(bufferPosition); + if (bufferPosition == mFrameCount) { + ALOGD("restoring track at end of static buffer"); + } } } -#endif if (mState == STATE_ACTIVE) { result = mAudioTrack->start(); } @@ -2148,8 +2163,8 @@ bool AudioTrack::AudioTrackThread::threadLoop() case NS_NEVER: return false; case NS_WHENEVER: - // FIXME increase poll interval, or make event-driven - ns = 1000000000LL; + // Event driven: call wake() when callback notifications conditions change. + ns = INT64_MAX; // fall through default: LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); @@ -2182,6 +2197,17 @@ void AudioTrack::AudioTrackThread::resume() } } +void AudioTrack::AudioTrackThread::wake() +{ + AutoMutex _l(mMyLock); + if (!mPaused && mPausedInt && mPausedNs > 0) { + // audio track is active and internally paused with timeout. + mIgnoreNextPausedInt = true; + mPausedInt = false; + mMyCond.signal(); + } +} + void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) { AutoMutex _l(mMyLock); diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index ff24475..08241e2 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -28,7 +28,21 @@ namespace android { // used to clamp a value to size_t. TODO: move to another file. template <typename T> size_t clampToSize(T x) { - return x > SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x; + return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x; +} + +// incrementSequence is used to determine the next sequence value +// for the loop and position sequence counters. It should return +// a value between "other" + 1 and "other" + INT32_MAX, the choice of +// which needs to be the "least recently used" sequence value for "self". +// In general, this means (new_self) returned is max(self, other) + 1. + +static uint32_t incrementSequence(uint32_t self, uint32_t other) { + int32_t diff = self - other; + if (diff >= 0 && diff < INT32_MAX) { + return self + 1; // we're already ahead of other. + } + return other + 1; // we're behind, so move just ahead of other. } audio_track_cblk_t::audio_track_cblk_t() @@ -485,8 +499,11 @@ end: StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize), - mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0) + mMutator(&cblk->u.mStatic.mSingleStateQueue), + mPosLoopObserver(&cblk->u.mStatic.mPosLoopQueue) { + memset(&mState, 0, sizeof(mState)); + memset(&mPosLoop, 0, sizeof(mPosLoop)); } void StaticAudioTrackClientProxy::flush() @@ -501,30 +518,72 @@ void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int // FIXME Should return an error status return; } - StaticAudioTrackState newState; - newState.mLoopStart = (uint32_t) loopStart; - newState.mLoopEnd = (uint32_t) loopEnd; - newState.mLoopCount = loopCount; - size_t bufferPosition; - if (loopCount == 0 || (bufferPosition = getBufferPosition()) >= loopEnd) { - bufferPosition = loopStart; + mState.mLoopStart = (uint32_t) loopStart; + mState.mLoopEnd = (uint32_t) loopEnd; + mState.mLoopCount = loopCount; + mState.mLoopSequence = incrementSequence(mState.mLoopSequence, mState.mPositionSequence); + // set patch-up variables until the mState is acknowledged by the ServerProxy. + // observed buffer position and loop count will freeze until then to give the + // illusion of a synchronous change. + getBufferPositionAndLoopCount(NULL, NULL); + // preserve behavior to restart at mState.mLoopStart if position exceeds mState.mLoopEnd. + if (mState.mLoopCount != 0 && mPosLoop.mBufferPosition >= mState.mLoopEnd) { + mPosLoop.mBufferPosition = mState.mLoopStart; } - mBufferPosition = bufferPosition; // snapshot buffer position until loop is acknowledged. - (void) mMutator.push(newState); + mPosLoop.mLoopCount = mState.mLoopCount; + (void) mMutator.push(mState); +} + +void StaticAudioTrackClientProxy::setBufferPosition(size_t position) +{ + // This can only happen on a 64-bit client + if (position > UINT32_MAX) { + // FIXME Should return an error status + return; + } + mState.mPosition = (uint32_t) position; + mState.mPositionSequence = incrementSequence(mState.mPositionSequence, mState.mLoopSequence); + // set patch-up variables until the mState is acknowledged by the ServerProxy. + // observed buffer position and loop count will freeze until then to give the + // illusion of a synchronous change. + if (mState.mLoopCount > 0) { // only check if loop count is changing + getBufferPositionAndLoopCount(NULL, NULL); // get last position + } + mPosLoop.mBufferPosition = position; + if (position >= mState.mLoopEnd) { + // no ongoing loop is possible if position is greater than loopEnd. + mPosLoop.mLoopCount = 0; + } + (void) mMutator.push(mState); +} + +void StaticAudioTrackClientProxy::setBufferPositionAndLoop(size_t position, size_t loopStart, + size_t loopEnd, int loopCount) +{ + setLoop(loopStart, loopEnd, loopCount); + setBufferPosition(position); } size_t StaticAudioTrackClientProxy::getBufferPosition() { - size_t bufferPosition; - if (mMutator.ack()) { - bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition; - if (bufferPosition > mFrameCount) { - bufferPosition = mFrameCount; - } - } else { - bufferPosition = mBufferPosition; + getBufferPositionAndLoopCount(NULL, NULL); + return mPosLoop.mBufferPosition; +} + +void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount( + size_t *position, int *loopCount) +{ + if (mMutator.ack() == StaticAudioTrackSingleStateQueue::SSQ_DONE) { + if (mPosLoopObserver.poll(mPosLoop)) { + ; // a valid mPosLoop should be available if ackDone is true. + } + } + if (position != NULL) { + *position = mPosLoop.mBufferPosition; + } + if (loopCount != NULL) { + *loopCount = mPosLoop.mLoopCount; } - return bufferPosition; } // --------------------------------------------------------------------------- @@ -560,7 +619,8 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush) ssize_t filled = rear - newFront; // Rather than shutting down on a corrupt flush, just treat it as a full flush if (!(0 <= filled && (size_t) filled <= mFrameCount)) { - ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, filled %d=%#x", + ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, " + "filled %d=%#x", mFlush, flush, front, rear, mask, newFront, filled, filled); newFront = rear; } @@ -739,13 +799,12 @@ void AudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount) StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize), - mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0), + mObserver(&cblk->u.mStatic.mSingleStateQueue), + mPosLoopMutator(&cblk->u.mStatic.mPosLoopQueue), mFramesReadySafe(frameCount), mFramesReady(frameCount), mFramesReadyIsCalledByMultipleThreads(false) { - mState.mLoopStart = 0; - mState.mLoopEnd = 0; - mState.mLoopCount = 0; + memset(&mState, 0, sizeof(mState)); } void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads() @@ -762,55 +821,97 @@ size_t StaticAudioTrackServerProxy::framesReady() return mFramesReadySafe; } -ssize_t StaticAudioTrackServerProxy::pollPosition() +status_t StaticAudioTrackServerProxy::updateStateWithLoop( + StaticAudioTrackState *localState, const StaticAudioTrackState &update) const { - size_t position = mPosition; - StaticAudioTrackState state; - if (mObserver.poll(state)) { + if (localState->mLoopSequence != update.mLoopSequence) { bool valid = false; - size_t loopStart = state.mLoopStart; - size_t loopEnd = state.mLoopEnd; - if (state.mLoopCount == 0) { - if (loopStart > mFrameCount) { - loopStart = mFrameCount; - } - // ignore loopEnd - mPosition = position = loopStart; - mFramesReady = mFrameCount - mPosition; - mState.mLoopCount = 0; + const size_t loopStart = update.mLoopStart; + const size_t loopEnd = update.mLoopEnd; + size_t position = localState->mPosition; + if (update.mLoopCount == 0) { valid = true; - } else if (state.mLoopCount >= -1) { + } else if (update.mLoopCount >= -1) { if (loopStart < loopEnd && loopEnd <= mFrameCount && loopEnd - loopStart >= MIN_LOOP) { // If the current position is greater than the end of the loop // we "wrap" to the loop start. This might cause an audible pop. if (position >= loopEnd) { - mPosition = position = loopStart; - } - if (state.mLoopCount == -1) { - mFramesReady = INT64_MAX; - } else { - // mFramesReady is 64 bits to handle the effective number of frames - // that the static audio track contains, including loops. - // TODO: Later consider fixing overflow, but does not seem needed now - // as will not overflow if loopStart and loopEnd are Java "ints". - mFramesReady = int64_t(state.mLoopCount) * (loopEnd - loopStart) - + mFrameCount - mPosition; + position = loopStart; } - mState = state; valid = true; } } - if (!valid || mPosition > mFrameCount) { + if (!valid || position > mFrameCount) { + return NO_INIT; + } + localState->mPosition = position; + localState->mLoopCount = update.mLoopCount; + localState->mLoopEnd = loopEnd; + localState->mLoopStart = loopStart; + localState->mLoopSequence = update.mLoopSequence; + } + return OK; +} + +status_t StaticAudioTrackServerProxy::updateStateWithPosition( + StaticAudioTrackState *localState, const StaticAudioTrackState &update) const +{ + if (localState->mPositionSequence != update.mPositionSequence) { + if (update.mPosition > mFrameCount) { + return NO_INIT; + } else if (localState->mLoopCount != 0 && update.mPosition >= localState->mLoopEnd) { + localState->mLoopCount = 0; // disable loop count if position is beyond loop end. + } + localState->mPosition = update.mPosition; + localState->mPositionSequence = update.mPositionSequence; + } + return OK; +} + +ssize_t StaticAudioTrackServerProxy::pollPosition() +{ + StaticAudioTrackState state; + if (mObserver.poll(state)) { + StaticAudioTrackState trystate = mState; + bool result; + const int32_t diffSeq = state.mLoopSequence - state.mPositionSequence; + + if (diffSeq < 0) { + result = updateStateWithLoop(&trystate, state) == OK && + updateStateWithPosition(&trystate, state) == OK; + } else { + result = updateStateWithPosition(&trystate, state) == OK && + updateStateWithLoop(&trystate, state) == OK; + } + if (!result) { + mObserver.done(); + // caution: no update occurs so server state will be inconsistent with client state. ALOGE("%s client pushed an invalid state, shutting down", __func__); mIsShutdown = true; return (ssize_t) NO_INIT; } + mState = trystate; + if (mState.mLoopCount == -1) { + mFramesReady = INT64_MAX; + } else if (mState.mLoopCount == 0) { + mFramesReady = mFrameCount - mState.mPosition; + } else if (mState.mLoopCount > 0) { + // TODO: Later consider fixing overflow, but does not seem needed now + // as will not overflow if loopStart and loopEnd are Java "ints". + mFramesReady = int64_t(mState.mLoopCount) * (mState.mLoopEnd - mState.mLoopStart) + + mFrameCount - mState.mPosition; + } mFramesReadySafe = clampToSize(mFramesReady); // This may overflow, but client is not supposed to rely on it - mCblk->u.mStatic.mBufferPosition = (uint32_t) position; + StaticAudioTrackPosLoop posLoop; + + posLoop.mLoopCount = (int32_t) mState.mLoopCount; + posLoop.mBufferPosition = (uint32_t) mState.mPosition; + mPosLoopMutator.push(posLoop); + mObserver.done(); // safe to read mStatic variables. } - return (ssize_t) position; + return (ssize_t) mState.mPosition; } status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused) @@ -849,7 +950,7 @@ status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush } // As mFramesReady is the total remaining frames in the static audio track, // it is always larger or equal to avail. - LOG_ALWAYS_FATAL_IF(mFramesReady < avail); + LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail); buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail); mUnreleased = avail; return NO_ERROR; @@ -858,7 +959,7 @@ status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) { size_t stepCount = buffer->mFrameCount; - LOG_ALWAYS_FATAL_IF(!(stepCount <= mFramesReady)); + LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady)); LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased)); if (stepCount == 0) { // prevent accidental re-use of buffer @@ -868,11 +969,12 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) } mUnreleased -= stepCount; audio_track_cblk_t* cblk = mCblk; - size_t position = mPosition; + size_t position = mState.mPosition; size_t newPosition = position + stepCount; int32_t setFlags = 0; if (!(position <= newPosition && newPosition <= mFrameCount)) { - ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount); + ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, + mFrameCount); newPosition = mFrameCount; } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) { newPosition = mState.mLoopStart; @@ -885,7 +987,7 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) if (newPosition == mFrameCount) { setFlags |= CBLK_BUFFER_END; } - mPosition = newPosition; + mState.mPosition = newPosition; if (mFramesReady != INT64_MAX) { mFramesReady -= stepCount; } @@ -893,7 +995,10 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) cblk->mServer += stepCount; // This may overflow, but client is not supposed to rely on it - cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition; + StaticAudioTrackPosLoop posLoop; + posLoop.mBufferPosition = mState.mPosition; + posLoop.mLoopCount = mState.mLoopCount; + mPosLoopMutator.push(posLoop); if (setFlags != 0) { (void) android_atomic_or(setFlags, &cblk->mFlags); // this would be a good place to wake a futex diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp index a733b68..9181f86 100644 --- a/media/libmedia/IMediaRecorder.cpp +++ b/media/libmedia/IMediaRecorder.cpp @@ -46,7 +46,6 @@ enum { SET_OUTPUT_FORMAT, SET_VIDEO_ENCODER, SET_AUDIO_ENCODER, - SET_OUTPUT_FILE_PATH, SET_OUTPUT_FILE_FD, SET_VIDEO_SIZE, SET_VIDEO_FRAMERATE, @@ -158,16 +157,6 @@ public: return reply.readInt32(); } - status_t setOutputFile(const char* path) - { - ALOGV("setOutputFile(%s)", path); - Parcel data, reply; - data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor()); - data.writeCString(path); - remote()->transact(SET_OUTPUT_FILE_PATH, data, &reply); - return reply.readInt32(); - } - status_t setOutputFile(int fd, int64_t offset, int64_t length) { ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length); Parcel data, reply; @@ -300,7 +289,8 @@ IMPLEMENT_META_INTERFACE(MediaRecorder, "android.media.IMediaRecorder"); // ---------------------------------------------------------------------- status_t BnMediaRecorder::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) + uint32_t code, const Parcel& data, Parcel* reply, + uint32_t flags) { switch (code) { case RELEASE: { @@ -390,13 +380,6 @@ status_t BnMediaRecorder::onTransact( return NO_ERROR; } break; - case SET_OUTPUT_FILE_PATH: { - ALOGV("SET_OUTPUT_FILE_PATH"); - CHECK_INTERFACE(IMediaRecorder, data, reply); - const char* path = data.readCString(); - reply->writeInt32(setOutputFile(path)); - return NO_ERROR; - } break; case SET_OUTPUT_FILE_FD: { ALOGV("SET_OUTPUT_FILE_FD"); CHECK_INTERFACE(IMediaRecorder, data, reply); @@ -445,7 +428,8 @@ status_t BnMediaRecorder::onTransact( case SET_PREVIEW_SURFACE: { ALOGV("SET_PREVIEW_SURFACE"); CHECK_INTERFACE(IMediaRecorder, data, reply); - sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(data.readStrongBinder()); + sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>( + data.readStrongBinder()); reply->writeInt32(setPreviewSurface(surface)); return NO_ERROR; } break; diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp index 721d8d7..271be0c 100644 --- a/media/libmedia/JetPlayer.cpp +++ b/media/libmedia/JetPlayer.cpp @@ -408,7 +408,8 @@ int JetPlayer::queueSegment(int segmentNum, int libNum, int repeatCount, int tra ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d", segmentNum, libNum, repeatCount, transpose); Mutex::Autolock lock(mMutex); - return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags, userID); + return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags, + userID); } //------------------------------------------------------------------------------------------------- @@ -449,7 +450,8 @@ void JetPlayer::dump() void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus) { if (pJetStatus!=NULL) - ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d paused=%d", + ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d " + "paused=%d", pJetStatus->currentUserID, pJetStatus->segmentRepeatCount, pJetStatus->numQueuedSegments, pJetStatus->paused); else diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp index e2e6042..47f9258 100644 --- a/media/libmedia/MediaProfiles.cpp +++ b/media/libmedia/MediaProfiles.cpp @@ -163,7 +163,8 @@ MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap UNUSED } /*static*/ int -MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings, const char *name) +MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings, + const char *name) { int tag = -1; for (size_t i = 0; i < nMappings; ++i) { @@ -295,9 +296,8 @@ MediaProfiles::createAudioEncoderCap(const char **atts) CHECK(codec != -1); MediaProfiles::AudioEncoderCap *cap = - new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]), atoi(atts[7]), - atoi(atts[9]), atoi(atts[11]), atoi(atts[13]), - atoi(atts[15])); + new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]), + atoi(atts[7]), atoi(atts[9]), atoi(atts[11]), atoi(atts[13]), atoi(atts[15])); logAudioEncoderCap(*cap); return cap; } @@ -330,7 +330,8 @@ MediaProfiles::createCamcorderProfile(int cameraId, const char **atts, Vector<in !strcmp("fileFormat", atts[2]) && !strcmp("duration", atts[4])); - const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/sizeof(sCamcorderQualityNameMap[0]); + const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/ + sizeof(sCamcorderQualityNameMap[0]); const int quality = findTagForName(sCamcorderQualityNameMap, nProfileMappings, atts[1]); CHECK(quality != -1); @@ -722,16 +723,20 @@ MediaProfiles::createDefaultCamcorderTimeLapse480pProfile(camcorder_quality qual MediaProfiles::createDefaultCamcorderTimeLapseLowProfiles( MediaProfiles::CamcorderProfile **lowTimeLapseProfile, MediaProfiles::CamcorderProfile **lowSpecificTimeLapseProfile) { - *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_LOW); - *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_QCIF); + *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile( + CAMCORDER_QUALITY_TIME_LAPSE_LOW); + *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile( + CAMCORDER_QUALITY_TIME_LAPSE_QCIF); } /*static*/ void MediaProfiles::createDefaultCamcorderTimeLapseHighProfiles( MediaProfiles::CamcorderProfile **highTimeLapseProfile, MediaProfiles::CamcorderProfile **highSpecificTimeLapseProfile) { - *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_HIGH); - *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_480P); + *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile( + CAMCORDER_QUALITY_TIME_LAPSE_HIGH); + *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile( + CAMCORDER_QUALITY_TIME_LAPSE_480P); } /*static*/ MediaProfiles::CamcorderProfile* @@ -809,7 +814,8 @@ MediaProfiles::createDefaultCamcorderProfiles(MediaProfiles *profiles) // high camcorder time lapse profiles. MediaProfiles::CamcorderProfile *highTimeLapseProfile, *highSpecificTimeLapseProfile; - createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile, &highSpecificTimeLapseProfile); + createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile, + &highSpecificTimeLapseProfile); profiles->mCamcorderProfiles.add(highTimeLapseProfile); profiles->mCamcorderProfiles.add(highSpecificTimeLapseProfile); diff --git a/media/libmedia/SingleStateQueue.cpp b/media/libmedia/SingleStateQueue.cpp deleted file mode 100644 index c241184..0000000 --- a/media/libmedia/SingleStateQueue.cpp +++ /dev/null @@ -1,106 +0,0 @@ -/* - * Copyright (C) 2012 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. - */ - -#include <new> -#include <cutils/atomic.h> -#include <media/SingleStateQueue.h> - -namespace android { - -template<typename T> SingleStateQueue<T>::Mutator::Mutator(Shared *shared) - : mSequence(0), mShared((Shared *) shared) -{ - // exactly one of Mutator and Observer must initialize, currently it is Observer - //shared->init(); -} - -template<typename T> int32_t SingleStateQueue<T>::Mutator::push(const T& value) -{ - Shared *shared = mShared; - int32_t sequence = mSequence; - sequence++; - android_atomic_acquire_store(sequence, &shared->mSequence); - shared->mValue = value; - sequence++; - android_atomic_release_store(sequence, &shared->mSequence); - mSequence = sequence; - // consider signalling a futex here, if we know that observer is waiting - return sequence; -} - -template<typename T> bool SingleStateQueue<T>::Mutator::ack() -{ - return mShared->mAck - mSequence == 0; -} - -template<typename T> bool SingleStateQueue<T>::Mutator::ack(int32_t sequence) -{ - // this relies on 2's complement rollover to detect an ancient sequence number - return mShared->mAck - sequence >= 0; -} - -template<typename T> SingleStateQueue<T>::Observer::Observer(Shared *shared) - : mSequence(0), mSeed(1), mShared((Shared *) shared) -{ - // exactly one of Mutator and Observer must initialize, currently it is Observer - shared->init(); -} - -template<typename T> bool SingleStateQueue<T>::Observer::poll(T& value) -{ - Shared *shared = mShared; - int32_t before = shared->mSequence; - if (before == mSequence) { - return false; - } - for (int tries = 0; ; ) { - const int MAX_TRIES = 5; - if (before & 1) { - if (++tries >= MAX_TRIES) { - return false; - } - before = shared->mSequence; - } else { - android_memory_barrier(); - T temp = shared->mValue; - int32_t after = android_atomic_release_load(&shared->mSequence); - if (after == before) { - value = temp; - shared->mAck = before; - mSequence = before; - return true; - } - if (++tries >= MAX_TRIES) { - return false; - } - before = after; - } - } -} - -#if 0 -template<typename T> SingleStateQueue<T>::SingleStateQueue(void /*Shared*/ *shared) -{ - ((Shared *) shared)->init(); -} -#endif - -} // namespace android - -// hack for gcc -#ifdef SINGLE_STATE_QUEUE_INSTANTIATIONS -#include SINGLE_STATE_QUEUE_INSTANTIATIONS -#endif diff --git a/media/libmedia/SingleStateQueueInstantiations.cpp b/media/libmedia/SingleStateQueueInstantiations.cpp deleted file mode 100644 index 0265c8c..0000000 --- a/media/libmedia/SingleStateQueueInstantiations.cpp +++ /dev/null @@ -1,28 +0,0 @@ -/* - * Copyright (C) 2012 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. - */ - -#include <media/SingleStateQueue.h> -#include <private/media/StaticAudioTrackState.h> -#include <media/AudioTimestamp.h> - -// FIXME hack for gcc - -namespace android { - -template class SingleStateQueue<StaticAudioTrackState>; // typedef StaticAudioTrackSingleStateQueue -template class SingleStateQueue<AudioTimestamp>; // typedef AudioTimestampSingleStateQueue - -} diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp index 5f5b57a..477e3fd 100644 --- a/media/libmedia/StringArray.cpp +++ b/media/libmedia/StringArray.cpp @@ -16,7 +16,7 @@ // // Sortable array of strings. STL-ish, but STL-free. -// +// #include <stdlib.h> #include <string.h> diff --git a/media/libmedia/docs/Makefile b/media/libmedia/docs/Makefile new file mode 100644 index 0000000..bddbc9b --- /dev/null +++ b/media/libmedia/docs/Makefile @@ -0,0 +1,2 @@ +paused.png : paused.dot + dot -Tpng < $< > $@ diff --git a/media/libmedia/docs/paused.dot b/media/libmedia/docs/paused.dot new file mode 100644 index 0000000..11e1777 --- /dev/null +++ b/media/libmedia/docs/paused.dot @@ -0,0 +1,85 @@ +digraph paused { +initial [label="INITIAL\n\ +mIgnoreNextPausedInt = false\n\ +mPaused = false\n\ +mPausedInt = false"]; + +resume_body [label="mIgnoreNextPausedInt = true\nif (mPaused || mPausedInt)"]; +resume_paused [label="mPaused = false\nmPausedInt = false\nsignal()"]; +resume_paused -> resume_merged; +resume_merged [label="return"]; + +Application -> ATstop; +ATstop [label="AudioTrack::stop()"]; +ATstop -> pause; +Application -> ATpause; +ATpause [label="AudioTrack::pause()"]; +ATpause -> pause; +ATstart -> resume; +ATstart [label="AudioTrack::start()"]; +destructor [label="~AudioTrack()"]; +destructor -> requestExit; +requestExit [label="AudioTrackThread::requestExit()"]; +requestExit -> resume; +Application -> ATsetMarkerPosition +ATsetMarkerPosition [label="AudioTrack::setMarkerPosition()\n[sets marker variables]"]; +ATsetMarkerPosition -> ATTwake +Application -> ATsetPositionUpdatePeriod +ATsetPositionUpdatePeriod [label="AudioTrack::setPositionUpdatePeriod()\n[sets update period variables]"]; +ATsetPositionUpdatePeriod -> ATTwake +Application -> ATstart; + +resume [label="AudioTrackThread::resume()"]; +resume -> resume_body; + +resume_body -> resume_paused [label="true"]; +resume_body -> resume_merged [label="false"]; + +ATTwake [label="AudioTrackThread::wake()\nif (!mPaused && mPausedInt && mPausedNs > 0)"]; +ATTwake-> ATTWake_wakeable [label="true"]; +ATTWake_wakeable [label="mIgnoreNextPausedInt = true\nmPausedInt = false\nsignal()"]; +ATTwake-> ATTWake_cannotwake [label="false"] +ATTWake_cannotwake [label="ignore"]; + +pause [label="mPaused = true"]; +pause -> return; + +threadLoop [label="AudioTrackThread::threadLoop()\nENTRY"]; +threadLoop -> threadLoop_1; +threadLoop_1 [label="if (mPaused)"]; +threadLoop_1 -> threadLoop_1_true [label="true"]; +threadLoop_1 -> threadLoop_2 [label="false"]; +threadLoop_1_true [label="wait()\nreturn true"]; +threadLoop_2 [label="if (mIgnoreNextPausedInt)"]; +threadLoop_2 -> threadLoop_2_true [label="true"]; +threadLoop_2 -> threadLoop_3 [label="false"]; +threadLoop_2_true [label="mIgnoreNextPausedInt = false\nmPausedInt = false"]; +threadLoop_2_true -> threadLoop_3; +threadLoop_3 [label="if (mPausedInt)"]; +threadLoop_3 -> threadLoop_3_true [label="true"]; +threadLoop_3 -> threadLoop_4 [label="false"]; +threadLoop_3_true [label="wait()\nmPausedInt = false\nreturn true"]; +threadLoop_4 [label="if (exitPending)"]; +threadLoop_4 -> threadLoop_4_true [label="true"]; +threadLoop_4 -> threadLoop_5 [label="false"]; +threadLoop_4_true [label="return false"]; +threadLoop_5 [label="ns = processAudioBuffer()"]; +threadLoop_5 -> threadLoop_6; +threadLoop_6 [label="case ns"]; +threadLoop_6 -> threadLoop_6_0 [label="0"]; +threadLoop_6 -> threadLoop_6_NS_INACTIVE [label="NS_INACTIVE"]; +threadLoop_6 -> threadLoop_6_NS_NEVER [label="NS_NEVER"]; +threadLoop_6 -> threadLoop_6_NS_WHENEVER [label="NS_WHENEVER"]; +threadLoop_6 -> threadLoop_6_default [label="default"]; +threadLoop_6_default [label="if (ns < 0)"]; +threadLoop_6_default -> threadLoop_6_default_true [label="true"]; +threadLoop_6_default -> threadLoop_6_default_false [label="false"]; +threadLoop_6_default_true [label="FATAL"]; +threadLoop_6_default_false [label="pauseInternal(ns) [wake()-able]\nmPausedInternal = true\nmPausedNs = ns\nreturn true"]; +threadLoop_6_0 [label="return true"]; +threadLoop_6_NS_INACTIVE [label="pauseInternal()\nmPausedInternal = true\nmPausedNs = 0\nreturn true"]; +threadLoop_6_NS_NEVER [label="return false"]; +threadLoop_6_NS_WHENEVER [label="ns = 1s"]; +threadLoop_6_NS_WHENEVER -> threadLoop_6_default_false; + +} diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp index 05c89ed..432ecda 100644 --- a/media/libmedia/mediaplayer.cpp +++ b/media/libmedia/mediaplayer.cpp @@ -240,7 +240,7 @@ status_t MediaPlayer::setVideoSurfaceTexture( // must call with lock held status_t MediaPlayer::prepareAsync_l() { - if ( (mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) { + if ( (mPlayer != 0) && ( mCurrentState & (MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) { mPlayer->setAudioStreamType(mStreamType); if (mAudioAttributesParcel != NULL) { mPlayer->setParameter(KEY_PARAMETER_AUDIO_ATTRIBUTES, *mAudioAttributesParcel); @@ -414,7 +414,8 @@ status_t MediaPlayer::getCurrentPosition(int *msec) status_t MediaPlayer::getDuration_l(int *msec) { ALOGV("getDuration_l"); - bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE)); + bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED | + MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE)); if (mPlayer != 0 && isValidState) { int durationMs; status_t ret = mPlayer->getDuration(&durationMs); @@ -443,7 +444,8 @@ status_t MediaPlayer::getDuration(int *msec) status_t MediaPlayer::seekTo_l(int msec) { ALOGV("seekTo %d", msec); - if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) { + if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED | + MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) { if ( msec < 0 ) { ALOGW("Attempt to seek to invalid position: %d", msec); msec = 0; @@ -477,7 +479,8 @@ status_t MediaPlayer::seekTo_l(int msec) return NO_ERROR; } } - ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(), mCurrentState); + ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(), + mCurrentState); return INVALID_OPERATION; } diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp index 1952b86..973e156 100644 --- a/media/libmedia/mediarecorder.cpp +++ b/media/libmedia/mediarecorder.cpp @@ -264,32 +264,6 @@ status_t MediaRecorder::setAudioEncoder(int ae) return ret; } -status_t MediaRecorder::setOutputFile(const char* path) -{ - ALOGV("setOutputFile(%s)", path); - if (mMediaRecorder == NULL) { - ALOGE("media recorder is not initialized yet"); - return INVALID_OPERATION; - } - if (mIsOutputFileSet) { - ALOGE("output file has already been set"); - return INVALID_OPERATION; - } - if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) { - ALOGE("setOutputFile called in an invalid state(%d)", mCurrentState); - return INVALID_OPERATION; - } - - status_t ret = mMediaRecorder->setOutputFile(path); - if (OK != ret) { - ALOGV("setOutputFile failed: %d", ret); - mCurrentState = MEDIA_RECORDER_ERROR; - return ret; - } - mIsOutputFileSet = true; - return ret; -} - status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length) { ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length); diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp index 194abbb..4d4de9b 100644 --- a/media/libmediaplayerservice/MediaRecorderClient.cpp +++ b/media/libmediaplayerservice/MediaRecorderClient.cpp @@ -154,17 +154,6 @@ status_t MediaRecorderClient::setAudioEncoder(int ae) return mRecorder->setAudioEncoder((audio_encoder)ae); } -status_t MediaRecorderClient::setOutputFile(const char* path) -{ - ALOGV("setOutputFile(%s)", path); - Mutex::Autolock lock(mLock); - if (mRecorder == NULL) { - ALOGE("recorder is not initialized"); - return NO_INIT; - } - return mRecorder->setOutputFile(path); -} - status_t MediaRecorderClient::setOutputFile(int fd, int64_t offset, int64_t length) { ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length); diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h index a65ec9f..a444b6c 100644 --- a/media/libmediaplayerservice/MediaRecorderClient.h +++ b/media/libmediaplayerservice/MediaRecorderClient.h @@ -38,7 +38,6 @@ public: virtual status_t setOutputFormat(int of); virtual status_t setVideoEncoder(int ve); virtual status_t setAudioEncoder(int ae); - virtual status_t setOutputFile(const char* path); virtual status_t setOutputFile(int fd, int64_t offset, int64_t length); virtual status_t setVideoSize(int width, int height); diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp index 86639cb..5c16920 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.cpp +++ b/media/libmediaplayerservice/StagefrightRecorder.cpp @@ -241,14 +241,6 @@ status_t StagefrightRecorder::setPreviewSurface(const sp<IGraphicBufferProducer> return OK; } -status_t StagefrightRecorder::setOutputFile(const char * /* path */) { - ALOGE("setOutputFile(const char*) must not be called"); - // We don't actually support this at all, as the media_server process - // no longer has permissions to create files. - - return -EPERM; -} - status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) { ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length); // These don't make any sense, do they? diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h index 54c38d3..a6eba39 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.h +++ b/media/libmediaplayerservice/StagefrightRecorder.h @@ -53,7 +53,6 @@ struct StagefrightRecorder : public MediaRecorderBase { virtual status_t setVideoFrameRate(int frames_per_second); virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy); virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface); - virtual status_t setOutputFile(const char *path); virtual status_t setOutputFile(int fd, int64_t offset, int64_t length); virtual status_t setParameters(const String8& params); virtual status_t setListener(const sp<IMediaRecorderClient>& listener); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp index bc79fdb..abfa4d3 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp @@ -351,6 +351,14 @@ status_t NuPlayerDriver::seekTo(int msec) { case STATE_PREPARED: case STATE_STOPPED_AND_PREPARED: { + int curpos = 0; + if (mPositionUs > 0) { + curpos = (mPositionUs + 500ll) / 1000; + } + if (curpos == msec) { + // nothing to do, and doing something anyway could result in deadlock (b/15323063) + break; + } mStartupSeekTimeUs = seekTimeUs; // pretend that the seek completed. It will actually happen when starting playback. // TODO: actually perform the seek here, so the player is ready to go at the new diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk index 9707c4a..1353f28 100644 --- a/media/libnbaio/Android.mk +++ b/media/libnbaio/Android.mk @@ -11,7 +11,6 @@ LOCAL_SRC_FILES := \ MonoPipeReader.cpp \ Pipe.cpp \ PipeReader.cpp \ - roundup.c \ SourceAudioBufferProvider.cpp LOCAL_SRC_FILES += NBLog.cpp @@ -27,12 +26,13 @@ LOCAL_SRC_FILES += NBLog.cpp LOCAL_MODULE := libnbaio LOCAL_SHARED_LIBRARIES := \ + libaudioutils \ libbinder \ libcommon_time_client \ libcutils \ libutils \ liblog -LOCAL_STATIC_LIBRARIES += libinstantssq +LOCAL_C_INCLUDES := $(call include-path-for, audio-utils) include $(BUILD_SHARED_LIBRARY) diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp index 0b65861..129e9ef 100644 --- a/media/libnbaio/MonoPipe.cpp +++ b/media/libnbaio/MonoPipe.cpp @@ -27,7 +27,7 @@ #include <utils/Trace.h> #include <media/AudioBufferProvider.h> #include <media/nbaio/MonoPipe.h> -#include <media/nbaio/roundup.h> +#include <audio_utils/roundup.h> namespace android { diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp index de82229..e4d3ed8 100644 --- a/media/libnbaio/MonoPipeReader.cpp +++ b/media/libnbaio/MonoPipeReader.cpp @@ -39,7 +39,7 @@ ssize_t MonoPipeReader::availableToRead() return NEGOTIATE; } ssize_t ret = android_atomic_acquire_load(&mPipe->mRear) - mPipe->mFront; - ALOG_ASSERT((0 <= ret) && (ret <= mMaxFrames)); + ALOG_ASSERT((0 <= ret) && ((size_t) ret <= mPipe->mMaxFrames)); return ret; } diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp index 6e0ec8c..13f211d 100644 --- a/media/libnbaio/Pipe.cpp +++ b/media/libnbaio/Pipe.cpp @@ -21,7 +21,7 @@ #include <cutils/compiler.h> #include <utils/Log.h> #include <media/nbaio/Pipe.h> -#include <media/nbaio/roundup.h> +#include <audio_utils/roundup.h> namespace android { diff --git a/media/libnbaio/roundup.c b/media/libnbaio/roundup.c deleted file mode 100644 index 1d552d1..0000000 --- a/media/libnbaio/roundup.c +++ /dev/null @@ -1,32 +0,0 @@ -/* - * Copyright (C) 2012 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. - */ - -#include <media/nbaio/roundup.h> - -unsigned roundup(unsigned v) -{ - // __builtin_clz is undefined for zero input - if (v == 0) { - v = 1; - } - int lz = __builtin_clz((int) v); - unsigned rounded = ((unsigned) 0x80000000) >> lz; - // 0x800000001 and higher are actually rounded _down_ to prevent overflow - if (v > rounded && lz > 0) { - rounded <<= 1; - } - return rounded; -} diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp index 2e41d80..9d90dbd 100644 --- a/media/libstagefright/AACWriter.cpp +++ b/media/libstagefright/AACWriter.cpp @@ -36,33 +36,19 @@ namespace android { -AACWriter::AACWriter(const char *filename) - : mFd(-1), - mInitCheck(NO_INIT), - mStarted(false), - mPaused(false), - mResumed(false), - mChannelCount(-1), - mSampleRate(-1), - mAACProfile(OMX_AUDIO_AACObjectLC) { - - ALOGV("AACWriter Constructor"); - - mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); - if (mFd >= 0) { - mInitCheck = OK; - } -} - AACWriter::AACWriter(int fd) : mFd(dup(fd)), mInitCheck(mFd < 0? NO_INIT: OK), mStarted(false), mPaused(false), mResumed(false), + mThread(0), + mEstimatedSizeBytes(0), + mEstimatedDurationUs(0), mChannelCount(-1), mSampleRate(-1), - mAACProfile(OMX_AUDIO_AACObjectLC) { + mAACProfile(OMX_AUDIO_AACObjectLC), + mFrameDurationUs(0) { } AACWriter::~AACWriter() { diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp index 9aa7d95..f53d7f0 100644 --- a/media/libstagefright/AMRWriter.cpp +++ b/media/libstagefright/AMRWriter.cpp @@ -31,19 +31,6 @@ namespace android { -AMRWriter::AMRWriter(const char *filename) - : mFd(-1), - mInitCheck(NO_INIT), - mStarted(false), - mPaused(false), - mResumed(false) { - - mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); - if (mFd >= 0) { - mInitCheck = OK; - } -} - AMRWriter::AMRWriter(int fd) : mFd(dup(fd)), mInitCheck(mFd < 0? NO_INIT: OK), diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index 2629afc..6d9bbae 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -101,6 +101,7 @@ LOCAL_STATIC_LIBRARIES := \ libstagefright_color_conversion \ libstagefright_aacenc \ libstagefright_matroska \ + libstagefright_mediafilter \ libstagefright_webm \ libstagefright_timedtext \ libvpx \ @@ -108,13 +109,14 @@ LOCAL_STATIC_LIBRARIES := \ libstagefright_mpeg2ts \ libstagefright_id3 \ libFLAC \ - libmedia_helper + libmedia_helper \ LOCAL_SHARED_LIBRARIES += \ libstagefright_enc_common \ libstagefright_avc_common \ libstagefright_foundation \ - libdl + libdl \ + libRScpp \ LOCAL_CFLAGS += -Wno-multichar diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp index a7ca3da..f0db76b 100644 --- a/media/libstagefright/FileSource.cpp +++ b/media/libstagefright/FileSource.cpp @@ -14,6 +14,10 @@ * limitations under the License. */ +//#define LOG_NDEBUG 0 +#define LOG_TAG "FileSource" +#include <utils/Log.h> + #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/FileSource.h> #include <sys/types.h> diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp index 9856f92..4359fb9 100644 --- a/media/libstagefright/MPEG2TSWriter.cpp +++ b/media/libstagefright/MPEG2TSWriter.cpp @@ -480,19 +480,6 @@ MPEG2TSWriter::MPEG2TSWriter(int fd) init(); } -MPEG2TSWriter::MPEG2TSWriter(const char *filename) - : mFile(fopen(filename, "wb")), - mWriteCookie(NULL), - mWriteFunc(NULL), - mStarted(false), - mNumSourcesDone(0), - mNumTSPacketsWritten(0), - mNumTSPacketsBeforeMeta(0), - mPATContinuityCounter(0), - mPMTContinuityCounter(0) { - init(); -} - MPEG2TSWriter::MPEG2TSWriter( void *cookie, ssize_t (*write)(void *cookie, const void *data, size_t size)) diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp index 9f20b1d..beb6f20 100644 --- a/media/libstagefright/MPEG4Writer.cpp +++ b/media/libstagefright/MPEG4Writer.cpp @@ -345,31 +345,6 @@ private: Track &operator=(const Track &); }; -MPEG4Writer::MPEG4Writer(const char *filename) - : mFd(-1), - mInitCheck(NO_INIT), - mIsRealTimeRecording(true), - mUse4ByteNalLength(true), - mUse32BitOffset(true), - mIsFileSizeLimitExplicitlyRequested(false), - mPaused(false), - mStarted(false), - mWriterThreadStarted(false), - mOffset(0), - mMdatOffset(0), - mEstimatedMoovBoxSize(0), - mInterleaveDurationUs(1000000), - mLatitudex10000(0), - mLongitudex10000(0), - mAreGeoTagsAvailable(false), - mStartTimeOffsetMs(-1) { - - mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); - if (mFd >= 0) { - mInitCheck = OK; - } -} - MPEG4Writer::MPEG4Writer(int fd) : mFd(dup(fd)), mInitCheck(mFd < 0? NO_INIT: OK), diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp index c2381b4..8fa07b2 100644 --- a/media/libstagefright/MediaCodec.cpp +++ b/media/libstagefright/MediaCodec.cpp @@ -36,6 +36,7 @@ #include <media/stagefright/MediaCodecList.h> #include <media/stagefright/MediaDefs.h> #include <media/stagefright/MediaErrors.h> +#include <media/stagefright/MediaFilter.h> #include <media/stagefright/MetaData.h> #include <media/stagefright/NativeWindowWrapper.h> #include <private/android_filesystem_config.h> @@ -189,7 +190,16 @@ status_t MediaCodec::init(const AString &name, bool nameIsType, bool encoder) { // quickly, violating the OpenMAX specs, until that is remedied // we need to invest in an extra looper to free the main event // queue. - mCodec = new ACodec; + + if (nameIsType || !strncasecmp(name.c_str(), "omx.", 4)) { + mCodec = new ACodec; + } else if (!nameIsType + && !strncasecmp(name.c_str(), "android.filter.", 15)) { + mCodec = new MediaFilter; + } else { + return NAME_NOT_FOUND; + } + bool needDedicatedLooper = false; if (nameIsType && !strncasecmp(name.c_str(), "video/", 6)) { needDedicatedLooper = true; diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp index c7c6f34..b13877d 100644 --- a/media/libstagefright/MediaMuxer.cpp +++ b/media/libstagefright/MediaMuxer.cpp @@ -38,21 +38,6 @@ namespace android { -MediaMuxer::MediaMuxer(const char *path, OutputFormat format) - : mFormat(format), - mState(UNINITIALIZED) { - if (format == OUTPUT_FORMAT_MPEG_4) { - mWriter = new MPEG4Writer(path); - } else if (format == OUTPUT_FORMAT_WEBM) { - mWriter = new WebmWriter(path); - } - - if (mWriter != NULL) { - mFileMeta = new MetaData; - mState = INITIALIZED; - } -} - MediaMuxer::MediaMuxer(int fd, OutputFormat format) : mFormat(format), mState(UNINITIALIZED) { diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp index 8a95643..6e6a78a 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp @@ -38,7 +38,10 @@ SoftVPX::SoftVPX( NULL /* profileLevels */, 0 /* numProfileLevels */, 320 /* width */, 240 /* height */, callbacks, appData, component), mMode(codingType == OMX_VIDEO_CodingVP8 ? MODE_VP8 : MODE_VP9), + mEOSStatus(INPUT_DATA_AVAILABLE), mCtx(NULL), + mFrameParallelMode(false), + mTimeStampIdx(0), mImg(NULL) { // arbitrary from avc/hevc as vpx does not specify a min compression ratio const size_t kMinCompressionRatio = mMode == MODE_VP8 ? 2 : 4; @@ -51,9 +54,7 @@ SoftVPX::SoftVPX( } SoftVPX::~SoftVPX() { - vpx_codec_destroy((vpx_codec_ctx_t *)mCtx); - delete (vpx_codec_ctx_t *)mCtx; - mCtx = NULL; + destroyDecoder(); } static int GetCPUCoreCount() { @@ -73,12 +74,19 @@ status_t SoftVPX::initDecoder() { mCtx = new vpx_codec_ctx_t; vpx_codec_err_t vpx_err; vpx_codec_dec_cfg_t cfg; + vpx_codec_flags_t flags; memset(&cfg, 0, sizeof(vpx_codec_dec_cfg_t)); + memset(&flags, 0, sizeof(vpx_codec_flags_t)); cfg.threads = GetCPUCoreCount(); + + if (mFrameParallelMode) { + flags |= VPX_CODEC_USE_FRAME_THREADING; + } + if ((vpx_err = vpx_codec_dec_init( (vpx_codec_ctx_t *)mCtx, mMode == MODE_VP8 ? &vpx_codec_vp8_dx_algo : &vpx_codec_vp9_dx_algo, - &cfg, 0))) { + &cfg, flags))) { ALOGE("on2 decoder failed to initialize. (%d)", vpx_err); return UNKNOWN_ERROR; } @@ -86,86 +94,155 @@ status_t SoftVPX::initDecoder() { return OK; } +status_t SoftVPX::destroyDecoder() { + vpx_codec_destroy((vpx_codec_ctx_t *)mCtx); + delete (vpx_codec_ctx_t *)mCtx; + mCtx = NULL; + return OK; +} + +bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset) { + List<BufferInfo *> &inQueue = getPortQueue(0); + List<BufferInfo *> &outQueue = getPortQueue(1); + BufferInfo *outInfo = NULL; + OMX_BUFFERHEADERTYPE *outHeader = NULL; + vpx_codec_iter_t iter = NULL; + + if (flushDecoder && mFrameParallelMode) { + // Flush decoder by passing NULL data ptr and 0 size. + // Ideally, this should never fail. + if (vpx_codec_decode((vpx_codec_ctx_t *)mCtx, NULL, 0, NULL, 0)) { + ALOGE("Failed to flush on2 decoder."); + return false; + } + } + + if (!display) { + if (!flushDecoder) { + ALOGE("Invalid operation."); + return false; + } + // Drop all the decoded frames in decoder. + while ((mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter))) { + } + return true; + } + + while (!outQueue.empty()) { + if (mImg == NULL) { + mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter); + if (mImg == NULL) { + break; + } + } + uint32_t width = mImg->d_w; + uint32_t height = mImg->d_h; + outInfo = *outQueue.begin(); + outHeader = outInfo->mHeader; + CHECK_EQ(mImg->fmt, IMG_FMT_I420); + handlePortSettingsChange(portWillReset, width, height); + if (*portWillReset) { + return true; + } + + outHeader->nOffset = 0; + outHeader->nFilledLen = (width * height * 3) / 2; + outHeader->nFlags = 0; + outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv; + + uint8_t *dst = outHeader->pBuffer; + const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y]; + const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U]; + const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V]; + size_t srcYStride = mImg->stride[PLANE_Y]; + size_t srcUStride = mImg->stride[PLANE_U]; + size_t srcVStride = mImg->stride[PLANE_V]; + copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride); + + mImg = NULL; + outInfo->mOwnedByUs = false; + outQueue.erase(outQueue.begin()); + outInfo = NULL; + notifyFillBufferDone(outHeader); + outHeader = NULL; + } + + if (!eos) { + return true; + } + + if (!outQueue.empty()) { + outInfo = *outQueue.begin(); + outQueue.erase(outQueue.begin()); + outHeader = outInfo->mHeader; + outHeader->nTimeStamp = 0; + outHeader->nFilledLen = 0; + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + outInfo->mOwnedByUs = false; + notifyFillBufferDone(outHeader); + mEOSStatus = OUTPUT_FRAMES_FLUSHED; + } + return true; +} + void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) { - if (mOutputPortSettingsChange != NONE) { + if (mOutputPortSettingsChange != NONE || mEOSStatus == OUTPUT_FRAMES_FLUSHED) { return; } List<BufferInfo *> &inQueue = getPortQueue(0); List<BufferInfo *> &outQueue = getPortQueue(1); bool EOSseen = false; + vpx_codec_err_t err; + bool portWillReset = false; + + while ((mEOSStatus == INPUT_EOS_SEEN || !inQueue.empty()) + && !outQueue.empty()) { + // Output the pending frames that left from last port reset or decoder flush. + if (mEOSStatus == INPUT_EOS_SEEN || mImg != NULL) { + if (!outputBuffers( + mEOSStatus == INPUT_EOS_SEEN, true /* display */, + mEOSStatus == INPUT_EOS_SEEN, &portWillReset)) { + ALOGE("on2 decoder failed to output frame."); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + if (portWillReset || mEOSStatus == OUTPUT_FRAMES_FLUSHED || + mEOSStatus == INPUT_EOS_SEEN) { + return; + } + } - while (!inQueue.empty() && !outQueue.empty()) { BufferInfo *inInfo = *inQueue.begin(); OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + mTimeStamps[mTimeStampIdx] = inHeader->nTimeStamp; BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mEOSStatus = INPUT_EOS_SEEN; EOSseen = true; - if (inHeader->nFilledLen == 0) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - outHeader->nFilledLen = 0; - outHeader->nFlags = OMX_BUFFERFLAG_EOS; - - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } } - if (mImg == NULL) { - if (vpx_codec_decode( - (vpx_codec_ctx_t *)mCtx, - inHeader->pBuffer + inHeader->nOffset, - inHeader->nFilledLen, - NULL, - 0)) { - ALOGE("on2 decoder failed to decode frame."); - - notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); - return; - } - vpx_codec_iter_t iter = NULL; - mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter); + if (inHeader->nFilledLen > 0 && + vpx_codec_decode((vpx_codec_ctx_t *)mCtx, + inHeader->pBuffer + inHeader->nOffset, + inHeader->nFilledLen, + &mTimeStamps[mTimeStampIdx], 0)) { + ALOGE("on2 decoder failed to decode frame."); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; } + mTimeStampIdx = (mTimeStampIdx + 1) % kNumBuffers; - if (mImg != NULL) { - CHECK_EQ(mImg->fmt, IMG_FMT_I420); - - uint32_t width = mImg->d_w; - uint32_t height = mImg->d_h; - bool portWillReset = false; - handlePortSettingsChange(&portWillReset, width, height); - if (portWillReset) { - return; - } - - outHeader->nOffset = 0; - outHeader->nFilledLen = (width * height * 3) / 2; - outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0; - outHeader->nTimeStamp = inHeader->nTimeStamp; - - uint8_t *dst = outHeader->pBuffer; - const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y]; - const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U]; - const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V]; - size_t srcYStride = mImg->stride[PLANE_Y]; - size_t srcUStride = mImg->stride[PLANE_U]; - size_t srcVStride = mImg->stride[PLANE_V]; - copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride); - - mImg = NULL; - outInfo->mOwnedByUs = false; - outQueue.erase(outQueue.begin()); - outInfo = NULL; - notifyFillBufferDone(outHeader); - outHeader = NULL; + if (!outputBuffers( + EOSseen /* flushDecoder */, true /* display */, EOSseen, &portWillReset)) { + ALOGE("on2 decoder failed to output frame."); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + if (portWillReset) { + return; } inInfo->mOwnedByUs = false; @@ -176,6 +253,30 @@ void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) { } } +void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) { + if (portIndex == kInputPortIndex) { + bool portWillReset = false; + if (!outputBuffers( + true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) { + ALOGE("Failed to flush decoder."); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + return; + } + mEOSStatus = INPUT_DATA_AVAILABLE; + } +} + +void SoftVPX::onReset() { + bool portWillReset = false; + if (!outputBuffers( + true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) { + ALOGW("Failed to flush decoder. Try to hard reset decoder"); + destroyDecoder(); + initDecoder(); + } + mEOSStatus = INPUT_DATA_AVAILABLE; +} + } // namespace android android::SoftOMXComponent *createSoftOMXComponent( diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h index 8f68693..8ccbae2 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.h +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h @@ -38,6 +38,8 @@ protected: virtual ~SoftVPX(); virtual void onQueueFilled(OMX_U32 portIndex); + virtual void onPortFlushCompleted(OMX_U32 portIndex); + virtual void onReset(); private: enum { @@ -49,11 +51,21 @@ private: MODE_VP9 } mMode; - void *mCtx; + enum { + INPUT_DATA_AVAILABLE, // VPX component is ready to decode data. + INPUT_EOS_SEEN, // VPX component saw EOS and is flushing On2 decoder. + OUTPUT_FRAMES_FLUSHED // VPX component finished flushing On2 decoder. + } mEOSStatus; + void *mCtx; + bool mFrameParallelMode; // Frame parallel is only supported by VP9 decoder. + OMX_TICKS mTimeStamps[kNumBuffers]; + uint8_t mTimeStampIdx; vpx_image_t *mImg; status_t initDecoder(); + status_t destroyDecoder(); + bool outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset); DISALLOW_EVIL_CONSTRUCTORS(SoftVPX); }; diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp index 1899b40..6474906 100644 --- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp +++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp @@ -98,32 +98,48 @@ void SoftwareRenderer::resetFormatIfChanged(const sp<AMessage> &format) { mCropWidth = mCropRight - mCropLeft + 1; mCropHeight = mCropBottom - mCropTop + 1; - int halFormat; - size_t bufWidth, bufHeight; - - switch (mColorFormat) { - case OMX_COLOR_FormatYUV420Planar: - case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar: - { - if (!runningInEmulator()) { + // by default convert everything to RGB565 + int halFormat = HAL_PIXEL_FORMAT_RGB_565; + size_t bufWidth = mCropWidth; + size_t bufHeight = mCropHeight; + + // hardware has YUV12 and RGBA8888 support, so convert known formats + if (!runningInEmulator()) { + switch (mColorFormat) { + case OMX_COLOR_FormatYUV420Planar: + case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar: + { halFormat = HAL_PIXEL_FORMAT_YV12; bufWidth = (mCropWidth + 1) & ~1; bufHeight = (mCropHeight + 1) & ~1; break; } - - // fall through. + case OMX_COLOR_Format24bitRGB888: + { + halFormat = HAL_PIXEL_FORMAT_RGB_888; + bufWidth = (mCropWidth + 1) & ~1; + bufHeight = (mCropHeight + 1) & ~1; + break; + } + case OMX_COLOR_Format32bitARGB8888: + case OMX_COLOR_Format32BitRGBA8888: + { + halFormat = HAL_PIXEL_FORMAT_RGBA_8888; + bufWidth = (mCropWidth + 1) & ~1; + bufHeight = (mCropHeight + 1) & ~1; + break; + } + default: + { + break; + } } + } - default: - halFormat = HAL_PIXEL_FORMAT_RGB_565; - bufWidth = mCropWidth; - bufHeight = mCropHeight; - - mConverter = new ColorConverter( - mColorFormat, OMX_COLOR_Format16bitRGB565); - CHECK(mConverter->isValid()); - break; + if (halFormat == HAL_PIXEL_FORMAT_RGB_565) { + mConverter = new ColorConverter( + mColorFormat, OMX_COLOR_Format16bitRGB565); + CHECK(mConverter->isValid()); } CHECK(mNativeWindow != NULL); @@ -200,6 +216,8 @@ void SoftwareRenderer::render( CHECK_EQ(0, mapper.lock( buf->handle, GRALLOC_USAGE_SW_WRITE_OFTEN, bounds, &dst)); + // TODO move the other conversions also into ColorConverter, and + // fix cropping issues (when mCropLeft/Top != 0 or mWidth != mCropWidth) if (mConverter) { mConverter->convert( data, @@ -210,7 +228,8 @@ void SoftwareRenderer::render( 0, 0, mCropWidth - 1, mCropHeight - 1); } else if (mColorFormat == OMX_COLOR_FormatYUV420Planar) { const uint8_t *src_y = (const uint8_t *)data; - const uint8_t *src_u = (const uint8_t *)data + mWidth * mHeight; + const uint8_t *src_u = + (const uint8_t *)data + mWidth * mHeight; const uint8_t *src_v = src_u + (mWidth / 2 * mHeight / 2); uint8_t *dst_y = (uint8_t *)dst; @@ -236,14 +255,10 @@ void SoftwareRenderer::render( dst_u += dst_c_stride; dst_v += dst_c_stride; } - } else { - CHECK_EQ(mColorFormat, OMX_TI_COLOR_FormatYUV420PackedSemiPlanar); - - const uint8_t *src_y = - (const uint8_t *)data; - - const uint8_t *src_uv = - (const uint8_t *)data + mWidth * (mHeight - mCropTop / 2); + } else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar) { + const uint8_t *src_y = (const uint8_t *)data; + const uint8_t *src_uv = (const uint8_t *)data + + mWidth * (mHeight - mCropTop / 2); uint8_t *dst_y = (uint8_t *)dst; @@ -271,6 +286,40 @@ void SoftwareRenderer::render( dst_u += dst_c_stride; dst_v += dst_c_stride; } + } else if (mColorFormat == OMX_COLOR_Format24bitRGB888) { + uint8_t* srcPtr = (uint8_t*)data; + uint8_t* dstPtr = (uint8_t*)dst; + + for (size_t y = 0; y < (size_t)mCropHeight; ++y) { + memcpy(dstPtr, srcPtr, mCropWidth * 3); + srcPtr += mWidth * 3; + dstPtr += buf->stride * 3; + } + } else if (mColorFormat == OMX_COLOR_Format32bitARGB8888) { + uint8_t *srcPtr, *dstPtr; + + for (size_t y = 0; y < (size_t)mCropHeight; ++y) { + srcPtr = (uint8_t*)data + mWidth * 4 * y; + dstPtr = (uint8_t*)dst + buf->stride * 4 * y; + for (size_t x = 0; x < (size_t)mCropWidth; ++x) { + uint8_t a = *srcPtr++; + for (size_t i = 0; i < 3; ++i) { // copy RGB + *dstPtr++ = *srcPtr++; + } + *dstPtr++ = a; // alpha last (ARGB to RGBA) + } + } + } else if (mColorFormat == OMX_COLOR_Format32BitRGBA8888) { + uint8_t* srcPtr = (uint8_t*)data; + uint8_t* dstPtr = (uint8_t*)dst; + + for (size_t y = 0; y < (size_t)mCropHeight; ++y) { + memcpy(dstPtr, srcPtr, mCropWidth * 4); + srcPtr += mWidth * 4; + dstPtr += buf->stride * 4; + } + } else { + LOG_ALWAYS_FATAL("bad color format %#x", mColorFormat); } CHECK_EQ(0, mapper.unlock(buf->handle)); diff --git a/media/libstagefright/filters/Android.mk b/media/libstagefright/filters/Android.mk new file mode 100644 index 0000000..36ab444 --- /dev/null +++ b/media/libstagefright/filters/Android.mk @@ -0,0 +1,27 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_SRC_FILES := \ + ColorConvert.cpp \ + GraphicBufferListener.cpp \ + IntrinsicBlurFilter.cpp \ + MediaFilter.cpp \ + RSFilter.cpp \ + SaturationFilter.cpp \ + saturationARGB.rs \ + SimpleFilter.cpp \ + ZeroFilter.cpp + +LOCAL_C_INCLUDES := \ + $(TOP)/frameworks/native/include/media/openmax \ + $(TOP)/frameworks/rs/cpp \ + $(TOP)/frameworks/rs \ + +intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,) +LOCAL_C_INCLUDES += $(intermediates) + +LOCAL_CFLAGS += -Wno-multichar + +LOCAL_MODULE:= libstagefright_mediafilter + +include $(BUILD_STATIC_LIBRARY) diff --git a/media/libstagefright/filters/ColorConvert.cpp b/media/libstagefright/filters/ColorConvert.cpp new file mode 100644 index 0000000..a5039f9 --- /dev/null +++ b/media/libstagefright/filters/ColorConvert.cpp @@ -0,0 +1,111 @@ +/* + * 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. + */ + +#include "ColorConvert.h" + +#ifndef max +#define max(a,b) ((a) > (b) ? (a) : (b)) +#endif +#ifndef min +#define min(a,b) ((a) < (b) ? (a) : (b)) +#endif + +namespace android { + +void YUVToRGB( + int32_t y, int32_t u, int32_t v, + int32_t* r, int32_t* g, int32_t* b) { + y -= 16; + u -= 128; + v -= 128; + + *b = 1192 * y + 2066 * u; + *g = 1192 * y - 833 * v - 400 * u; + *r = 1192 * y + 1634 * v; + + *r = min(262143, max(0, *r)); + *g = min(262143, max(0, *g)); + *b = min(262143, max(0, *b)); + + *r >>= 10; + *g >>= 10; + *b >>= 10; +} + +void convertYUV420spToARGB( + uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height, + uint8_t *dest) { + const int32_t bytes_per_pixel = 2; + + for (int32_t i = 0; i < height; i++) { + for (int32_t j = 0; j < width; j++) { + int32_t y = *(pY + i * width + j); + int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2)); + int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1); + + int32_t r, g, b; + YUVToRGB(y, u, v, &r, &g, &b); + + *dest++ = 0xFF; + *dest++ = r; + *dest++ = g; + *dest++ = b; + } + } +} + +void convertYUV420spToRGB888( + uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height, + uint8_t *dest) { + const int32_t bytes_per_pixel = 2; + + for (int32_t i = 0; i < height; i++) { + for (int32_t j = 0; j < width; j++) { + int32_t y = *(pY + i * width + j); + int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2)); + int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1); + + int32_t r, g, b; + YUVToRGB(y, u, v, &r, &g, &b); + + *dest++ = r; + *dest++ = g; + *dest++ = b; + } + } +} + +// HACK - not even slightly optimized +// TODO: remove when RGBA support is added to SoftwareRenderer +void convertRGBAToARGB( + uint8_t *src, int32_t width, int32_t height, uint32_t stride, + uint8_t *dest) { + for (size_t i = 0; i < height; ++i) { + for (size_t j = 0; j < width; ++j) { + uint8_t r = *src++; + uint8_t g = *src++; + uint8_t b = *src++; + uint8_t a = *src++; + *dest++ = a; + *dest++ = r; + *dest++ = g; + *dest++ = b; + } + src += (stride - width) * 4; + } +} + +} // namespace android diff --git a/media/libstagefright/filters/ColorConvert.h b/media/libstagefright/filters/ColorConvert.h new file mode 100644 index 0000000..13faa02 --- /dev/null +++ b/media/libstagefright/filters/ColorConvert.h @@ -0,0 +1,43 @@ +/* + * 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 COLOR_CONVERT_H_ +#define COLOR_CONVERT_H_ + +#include <inttypes.h> + +namespace android { + +void YUVToRGB( + int32_t y, int32_t u, int32_t v, + int32_t* r, int32_t* g, int32_t* b); + +void convertYUV420spToARGB( + uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height, + uint8_t *dest); + +void convertYUV420spToRGB888( + uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height, + uint8_t *dest); + +// TODO: remove when RGBA support is added to SoftwareRenderer +void convertRGBAToARGB( + uint8_t *src, int32_t width, int32_t height, uint32_t stride, + uint8_t *dest); + +} // namespace android + +#endif // COLOR_CONVERT_H_ diff --git a/media/libstagefright/filters/GraphicBufferListener.cpp b/media/libstagefright/filters/GraphicBufferListener.cpp new file mode 100644 index 0000000..fa38192 --- /dev/null +++ b/media/libstagefright/filters/GraphicBufferListener.cpp @@ -0,0 +1,154 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "GraphicBufferListener" + +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MediaErrors.h> + +#include "GraphicBufferListener.h" + +namespace android { + +status_t GraphicBufferListener::init( + const sp<AMessage> ¬ify, + size_t bufferWidth, size_t bufferHeight, size_t bufferCount) { + mNotify = notify; + + String8 name("GraphicBufferListener"); + BufferQueue::createBufferQueue(&mProducer, &mConsumer); + mConsumer->setConsumerName(name); + mConsumer->setDefaultBufferSize(bufferWidth, bufferHeight); + mConsumer->setConsumerUsageBits(GRALLOC_USAGE_SW_READ_OFTEN); + + status_t err = mConsumer->setMaxAcquiredBufferCount(bufferCount); + if (err != NO_ERROR) { + ALOGE("Unable to set BQ max acquired buffer count to %u: %d", + bufferCount, err); + return err; + } + + wp<BufferQueue::ConsumerListener> listener = + static_cast<BufferQueue::ConsumerListener*>(this); + sp<BufferQueue::ProxyConsumerListener> proxy = + new BufferQueue::ProxyConsumerListener(listener); + + err = mConsumer->consumerConnect(proxy, false); + if (err != NO_ERROR) { + ALOGE("Error connecting to BufferQueue: %s (%d)", + strerror(-err), err); + return err; + } + + ALOGV("init() successful."); + + return OK; +} + +void GraphicBufferListener::onFrameAvailable(const BufferItem& /* item */) { + ALOGV("onFrameAvailable() called"); + + { + Mutex::Autolock autoLock(mMutex); + mNumFramesAvailable++; + } + + sp<AMessage> notify = mNotify->dup(); + mNotify->setWhat(kWhatFrameAvailable); + mNotify->post(); +} + +void GraphicBufferListener::onBuffersReleased() { + ALOGV("onBuffersReleased() called"); + // nothing to do +} + +void GraphicBufferListener::onSidebandStreamChanged() { + ALOGW("GraphicBufferListener cannot consume sideband streams."); + // nothing to do +} + +BufferQueue::BufferItem GraphicBufferListener::getBufferItem() { + BufferQueue::BufferItem item; + + { + Mutex::Autolock autoLock(mMutex); + if (mNumFramesAvailable <= 0) { + ALOGE("getBuffer() called with no frames available"); + return item; + } + mNumFramesAvailable--; + } + + status_t err = mConsumer->acquireBuffer(&item, 0); + if (err == BufferQueue::NO_BUFFER_AVAILABLE) { + // shouldn't happen, since we track num frames available + ALOGE("frame was not available"); + item.mBuf = -1; + return item; + } else if (err != OK) { + ALOGE("acquireBuffer returned err=%d", err); + item.mBuf = -1; + return item; + } + + // Wait for it to become available. + err = item.mFence->waitForever("GraphicBufferListener::getBufferItem"); + if (err != OK) { + ALOGW("failed to wait for buffer fence: %d", err); + // keep going + } + + // If this is the first time we're seeing this buffer, add it to our + // slot table. + if (item.mGraphicBuffer != NULL) { + ALOGV("setting mBufferSlot %d", item.mBuf); + mBufferSlot[item.mBuf] = item.mGraphicBuffer; + } + + return item; +} + +sp<GraphicBuffer> GraphicBufferListener::getBuffer( + BufferQueue::BufferItem item) { + sp<GraphicBuffer> buf; + if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) { + ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf); + return buf; + } + + buf = mBufferSlot[item.mBuf]; + CHECK(buf.get() != NULL); + + return buf; +} + +status_t GraphicBufferListener::releaseBuffer( + BufferQueue::BufferItem item) { + if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) { + ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf); + return ERROR_OUT_OF_RANGE; + } + + mConsumer->releaseBuffer(item.mBuf, item.mFrameNumber, + EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE); + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/GraphicBufferListener.h b/media/libstagefright/filters/GraphicBufferListener.h new file mode 100644 index 0000000..b3e0ee3 --- /dev/null +++ b/media/libstagefright/filters/GraphicBufferListener.h @@ -0,0 +1,70 @@ +/* + * 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 GRAPHIC_BUFFER_LISTENER_H_ +#define GRAPHIC_BUFFER_LISTENER_H_ + +#include <gui/BufferQueue.h> + +namespace android { + +struct AMessage; + +struct GraphicBufferListener : public BufferQueue::ConsumerListener { +public: + GraphicBufferListener() {}; + + status_t init( + const sp<AMessage> ¬ify, + size_t bufferWidth, size_t bufferHeight, size_t bufferCount); + + virtual void onFrameAvailable(const BufferItem& item); + virtual void onBuffersReleased(); + virtual void onSidebandStreamChanged(); + + // Returns the handle to the producer side of the BufferQueue. Buffers + // queued on this will be received by GraphicBufferListener. + sp<IGraphicBufferProducer> getIGraphicBufferProducer() const { + return mProducer; + } + + BufferQueue::BufferItem getBufferItem(); + sp<GraphicBuffer> getBuffer(BufferQueue::BufferItem item); + status_t releaseBuffer(BufferQueue::BufferItem item); + + enum { + kWhatFrameAvailable = 'frav', + }; + +private: + sp<AMessage> mNotify; + size_t mNumFramesAvailable; + + mutable Mutex mMutex; + + // Our BufferQueue interfaces. mProducer is passed to the producer through + // getIGraphicBufferProducer, and mConsumer is used internally to retrieve + // the buffers queued by the producer. + sp<IGraphicBufferProducer> mProducer; + sp<IGraphicBufferConsumer> mConsumer; + + // Cache of GraphicBuffers from the buffer queue. + sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS]; +}; + +} // namespace android + +#endif // GRAPHIC_BUFFER_LISTENER_H diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.cpp b/media/libstagefright/filters/IntrinsicBlurFilter.cpp new file mode 100644 index 0000000..cbcf699 --- /dev/null +++ b/media/libstagefright/filters/IntrinsicBlurFilter.cpp @@ -0,0 +1,99 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "IntrinsicBlurFilter" + +#include <utils/Log.h> + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include "IntrinsicBlurFilter.h" + +namespace android { + +status_t IntrinsicBlurFilter::configure(const sp<AMessage> &msg) { + status_t err = SimpleFilter::configure(msg); + if (err != OK) { + return err; + } + + if (!msg->findString("cacheDir", &mCacheDir)) { + ALOGE("Failed to find cache directory in config message."); + return NAME_NOT_FOUND; + } + + return OK; +} + +status_t IntrinsicBlurFilter::start() { + // TODO: use a single RS context object for entire application + mRS = new RSC::RS(); + + if (!mRS->init(mCacheDir.c_str())) { + ALOGE("Failed to initialize RenderScript context."); + return NO_INIT; + } + + // 32-bit elements for ARGB8888 + RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS); + + RSC::Type::Builder tb(mRS, e); + tb.setX(mWidth); + tb.setY(mHeight); + RSC::sp<const RSC::Type> t = tb.create(); + + mAllocIn = RSC::Allocation::createTyped(mRS, t); + mAllocOut = RSC::Allocation::createTyped(mRS, t); + + mBlur = RSC::ScriptIntrinsicBlur::create(mRS, e); + mBlur->setRadius(mBlurRadius); + mBlur->setInput(mAllocIn); + + return OK; +} + +void IntrinsicBlurFilter::reset() { + mBlur.clear(); + mAllocOut.clear(); + mAllocIn.clear(); + mRS.clear(); +} + +status_t IntrinsicBlurFilter::setParameters(const sp<AMessage> &msg) { + sp<AMessage> params; + CHECK(msg->findMessage("params", ¶ms)); + + float blurRadius; + if (params->findFloat("blur-radius", &blurRadius)) { + mBlurRadius = blurRadius; + } + + return OK; +} + +status_t IntrinsicBlurFilter::processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) { + mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data()); + mBlur->forEach(mAllocOut); + mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data()); + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.h b/media/libstagefright/filters/IntrinsicBlurFilter.h new file mode 100644 index 0000000..4707ab7 --- /dev/null +++ b/media/libstagefright/filters/IntrinsicBlurFilter.h @@ -0,0 +1,50 @@ +/* + * 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 INTRINSIC_BLUR_FILTER_H_ +#define INTRINSIC_BLUR_FILTER_H_ + +#include "RenderScript.h" +#include "SimpleFilter.h" + +namespace android { + +struct IntrinsicBlurFilter : public SimpleFilter { +public: + IntrinsicBlurFilter() : mBlurRadius(1.f) {}; + + virtual status_t configure(const sp<AMessage> &msg); + virtual status_t start(); + virtual void reset(); + virtual status_t setParameters(const sp<AMessage> &msg); + virtual status_t processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer); + +protected: + virtual ~IntrinsicBlurFilter() {}; + +private: + AString mCacheDir; + RSC::sp<RSC::RS> mRS; + RSC::sp<RSC::Allocation> mAllocIn; + RSC::sp<RSC::Allocation> mAllocOut; + RSC::sp<RSC::ScriptIntrinsicBlur> mBlur; + float mBlurRadius; +}; + +} // namespace android + +#endif // INTRINSIC_BLUR_FILTER_H_ diff --git a/media/libstagefright/filters/MediaFilter.cpp b/media/libstagefright/filters/MediaFilter.cpp new file mode 100644 index 0000000..c5289b6 --- /dev/null +++ b/media/libstagefright/filters/MediaFilter.cpp @@ -0,0 +1,816 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "MediaFilter" + +#include <inttypes.h> +#include <utils/Trace.h> + +#include <binder/MemoryDealer.h> + +#include <media/stagefright/BufferProducerWrapper.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include <media/stagefright/MediaDefs.h> +#include <media/stagefright/MediaErrors.h> +#include <media/stagefright/MediaFilter.h> + +#include "ColorConvert.h" +#include "GraphicBufferListener.h" +#include "IntrinsicBlurFilter.h" +#include "RSFilter.h" +#include "SaturationFilter.h" +#include "ZeroFilter.h" + +namespace android { + +// parameter: number of input and output buffers +static const size_t kBufferCountActual = 4; + +MediaFilter::MediaFilter() + : mState(UNINITIALIZED), + mGeneration(0), + mGraphicBufferListener(NULL) { +} + +MediaFilter::~MediaFilter() { +} + +//////////////////// PUBLIC FUNCTIONS ////////////////////////////////////////// + +void MediaFilter::setNotificationMessage(const sp<AMessage> &msg) { + mNotify = msg; +} + +void MediaFilter::initiateAllocateComponent(const sp<AMessage> &msg) { + msg->setWhat(kWhatAllocateComponent); + msg->setTarget(id()); + msg->post(); +} + +void MediaFilter::initiateConfigureComponent(const sp<AMessage> &msg) { + msg->setWhat(kWhatConfigureComponent); + msg->setTarget(id()); + msg->post(); +} + +void MediaFilter::initiateCreateInputSurface() { + (new AMessage(kWhatCreateInputSurface, id()))->post(); +} + +void MediaFilter::initiateStart() { + (new AMessage(kWhatStart, id()))->post(); +} + +void MediaFilter::initiateShutdown(bool keepComponentAllocated) { + sp<AMessage> msg = new AMessage(kWhatShutdown, id()); + msg->setInt32("keepComponentAllocated", keepComponentAllocated); + msg->post(); +} + +void MediaFilter::signalFlush() { + (new AMessage(kWhatFlush, id()))->post(); +} + +void MediaFilter::signalResume() { + (new AMessage(kWhatResume, id()))->post(); +} + +// nothing to do +void MediaFilter::signalRequestIDRFrame() { + return; +} + +void MediaFilter::signalSetParameters(const sp<AMessage> ¶ms) { + sp<AMessage> msg = new AMessage(kWhatSetParameters, id()); + msg->setMessage("params", params); + msg->post(); +} + +void MediaFilter::signalEndOfInputStream() { + (new AMessage(kWhatSignalEndOfInputStream, id()))->post(); +} + +void MediaFilter::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatAllocateComponent: + { + onAllocateComponent(msg); + break; + } + case kWhatConfigureComponent: + { + onConfigureComponent(msg); + break; + } + case kWhatStart: + { + onStart(); + break; + } + case kWhatProcessBuffers: + { + processBuffers(); + break; + } + case kWhatInputBufferFilled: + { + onInputBufferFilled(msg); + break; + } + case kWhatOutputBufferDrained: + { + onOutputBufferDrained(msg); + break; + } + case kWhatShutdown: + { + onShutdown(msg); + break; + } + case kWhatFlush: + { + onFlush(); + break; + } + case kWhatResume: + { + // nothing to do + break; + } + case kWhatSetParameters: + { + onSetParameters(msg); + break; + } + case kWhatCreateInputSurface: + { + onCreateInputSurface(); + break; + } + case GraphicBufferListener::kWhatFrameAvailable: + { + onInputFrameAvailable(); + break; + } + case kWhatSignalEndOfInputStream: + { + onSignalEndOfInputStream(); + break; + } + default: + { + ALOGE("Message not handled:\n%s", msg->debugString().c_str()); + break; + } + } +} + +//////////////////// PORT DESCRIPTION ////////////////////////////////////////// + +MediaFilter::PortDescription::PortDescription() { +} + +void MediaFilter::PortDescription::addBuffer( + IOMX::buffer_id id, const sp<ABuffer> &buffer) { + mBufferIDs.push_back(id); + mBuffers.push_back(buffer); +} + +size_t MediaFilter::PortDescription::countBuffers() { + return mBufferIDs.size(); +} + +IOMX::buffer_id MediaFilter::PortDescription::bufferIDAt(size_t index) const { + return mBufferIDs.itemAt(index); +} + +sp<ABuffer> MediaFilter::PortDescription::bufferAt(size_t index) const { + return mBuffers.itemAt(index); +} + +//////////////////// HELPER FUNCTIONS ////////////////////////////////////////// + +void MediaFilter::signalProcessBuffers() { + (new AMessage(kWhatProcessBuffers, id()))->post(); +} + +void MediaFilter::signalError(status_t error) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatError); + notify->setInt32("err", error); + notify->post(); +} + +status_t MediaFilter::allocateBuffersOnPort(OMX_U32 portIndex) { + CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput); + const bool isInput = portIndex == kPortIndexInput; + const size_t bufferSize = isInput ? mMaxInputSize : mMaxOutputSize; + + CHECK(mDealer[portIndex] == NULL); + CHECK(mBuffers[portIndex].isEmpty()); + + ALOGV("Allocating %zu buffers of size %zu on %s port", + kBufferCountActual, bufferSize, + isInput ? "input" : "output"); + + size_t totalSize = kBufferCountActual * bufferSize; + + mDealer[portIndex] = new MemoryDealer(totalSize, "MediaFilter"); + + for (size_t i = 0; i < kBufferCountActual; ++i) { + sp<IMemory> mem = mDealer[portIndex]->allocate(bufferSize); + CHECK(mem.get() != NULL); + + BufferInfo info; + info.mStatus = BufferInfo::OWNED_BY_US; + info.mBufferID = i; + info.mGeneration = mGeneration; + info.mOutputFlags = 0; + info.mData = new ABuffer(mem->pointer(), bufferSize); + info.mData->meta()->setInt64("timeUs", 0); + + mBuffers[portIndex].push_back(info); + + if (!isInput) { + mAvailableOutputBuffers.push( + &mBuffers[portIndex].editItemAt(i)); + } + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatBuffersAllocated); + + notify->setInt32("portIndex", portIndex); + + sp<PortDescription> desc = new PortDescription; + + for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) { + const BufferInfo &info = mBuffers[portIndex][i]; + + desc->addBuffer(info.mBufferID, info.mData); + } + + notify->setObject("portDesc", desc); + notify->post(); + + return OK; +} + +MediaFilter::BufferInfo* MediaFilter::findBufferByID( + uint32_t portIndex, IOMX::buffer_id bufferID, + ssize_t *index) { + for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) { + BufferInfo *info = &mBuffers[portIndex].editItemAt(i); + + if (info->mBufferID == bufferID) { + if (index != NULL) { + *index = i; + } + return info; + } + } + + TRESPASS(); + + return NULL; +} + +void MediaFilter::postFillThisBuffer(BufferInfo *info) { + ALOGV("postFillThisBuffer on buffer %d", info->mBufferID); + if (mPortEOS[kPortIndexInput]) { + return; + } + + CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US); + + info->mGeneration = mGeneration; + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatFillThisBuffer); + notify->setInt32("buffer-id", info->mBufferID); + + info->mData->meta()->clear(); + notify->setBuffer("buffer", info->mData); + + sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, id()); + reply->setInt32("buffer-id", info->mBufferID); + + notify->setMessage("reply", reply); + + info->mStatus = BufferInfo::OWNED_BY_UPSTREAM; + notify->post(); +} + +void MediaFilter::postDrainThisBuffer(BufferInfo *info) { + CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US); + + info->mGeneration = mGeneration; + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatDrainThisBuffer); + notify->setInt32("buffer-id", info->mBufferID); + notify->setInt32("flags", info->mOutputFlags); + notify->setBuffer("buffer", info->mData); + + sp<AMessage> reply = new AMessage(kWhatOutputBufferDrained, id()); + reply->setInt32("buffer-id", info->mBufferID); + + notify->setMessage("reply", reply); + + notify->post(); + + info->mStatus = BufferInfo::OWNED_BY_UPSTREAM; +} + +void MediaFilter::postEOS() { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatEOS); + notify->setInt32("err", ERROR_END_OF_STREAM); + notify->post(); + + ALOGV("Sent kWhatEOS."); +} + +void MediaFilter::sendFormatChange() { + sp<AMessage> notify = mNotify->dup(); + + notify->setInt32("what", kWhatOutputFormatChanged); + + AString mime; + CHECK(mOutputFormat->findString("mime", &mime)); + notify->setString("mime", mime.c_str()); + + notify->setInt32("stride", mStride); + notify->setInt32("slice-height", mSliceHeight); + notify->setInt32("color-format", mColorFormatOut); + notify->setRect("crop", 0, 0, mStride - 1, mSliceHeight - 1); + notify->setInt32("width", mWidth); + notify->setInt32("height", mHeight); + + notify->post(); +} + +void MediaFilter::requestFillEmptyInput() { + if (mPortEOS[kPortIndexInput]) { + return; + } + + for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) { + BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i); + + if (info->mStatus == BufferInfo::OWNED_BY_US) { + postFillThisBuffer(info); + } + } +} + +void MediaFilter::processBuffers() { + if (mAvailableInputBuffers.empty() || mAvailableOutputBuffers.empty()) { + ALOGV("Skipping process (buffers unavailable)"); + return; + } + + if (mPortEOS[kPortIndexOutput]) { + // TODO notify caller of queueInput error when it is supported + // in MediaCodec + ALOGW("Tried to process a buffer after EOS."); + return; + } + + BufferInfo *inputInfo = mAvailableInputBuffers[0]; + mAvailableInputBuffers.removeAt(0); + BufferInfo *outputInfo = mAvailableOutputBuffers[0]; + mAvailableOutputBuffers.removeAt(0); + + status_t err; + err = mFilter->processBuffers(inputInfo->mData, outputInfo->mData); + if (err != (status_t)OK) { + outputInfo->mData->meta()->setInt32("err", err); + } + + int64_t timeUs; + CHECK(inputInfo->mData->meta()->findInt64("timeUs", &timeUs)); + outputInfo->mData->meta()->setInt64("timeUs", timeUs); + outputInfo->mOutputFlags = 0; + int32_t eos = 0; + if (inputInfo->mData->meta()->findInt32("eos", &eos) && eos != 0) { + outputInfo->mOutputFlags |= OMX_BUFFERFLAG_EOS; + mPortEOS[kPortIndexOutput] = true; + outputInfo->mData->meta()->setInt32("eos", eos); + postEOS(); + ALOGV("Output stream saw EOS."); + } + + ALOGV("Processed input buffer %u [%zu], output buffer %u [%zu]", + inputInfo->mBufferID, inputInfo->mData->size(), + outputInfo->mBufferID, outputInfo->mData->size()); + + if (mGraphicBufferListener != NULL) { + delete inputInfo; + } else { + postFillThisBuffer(inputInfo); + } + postDrainThisBuffer(outputInfo); + + // prevent any corner case where buffers could get stuck in queue + signalProcessBuffers(); +} + +void MediaFilter::onAllocateComponent(const sp<AMessage> &msg) { + CHECK_EQ(mState, UNINITIALIZED); + + CHECK(msg->findString("componentName", &mComponentName)); + const char* name = mComponentName.c_str(); + if (!strcasecmp(name, "android.filter.zerofilter")) { + mFilter = new ZeroFilter; + } else if (!strcasecmp(name, "android.filter.saturation")) { + mFilter = new SaturationFilter; + } else if (!strcasecmp(name, "android.filter.intrinsicblur")) { + mFilter = new IntrinsicBlurFilter; + } else if (!strcasecmp(name, "android.filter.RenderScript")) { + mFilter = new RSFilter; + } else { + ALOGE("Unrecognized filter name: %s", name); + signalError(NAME_NOT_FOUND); + return; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatComponentAllocated); + // HACK - need "OMX.google" to use MediaCodec's software renderer + notify->setString("componentName", "OMX.google.MediaFilter"); + notify->post(); + mState = INITIALIZED; + ALOGV("Handled kWhatAllocateComponent."); +} + +void MediaFilter::onConfigureComponent(const sp<AMessage> &msg) { + // TODO: generalize to allow audio filters as well as video + + CHECK_EQ(mState, INITIALIZED); + + // get params - at least mime, width & height + AString mime; + CHECK(msg->findString("mime", &mime)); + if (strcasecmp(mime.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) { + ALOGE("Bad mime: %s", mime.c_str()); + signalError(BAD_VALUE); + return; + } + + CHECK(msg->findInt32("width", &mWidth)); + CHECK(msg->findInt32("height", &mHeight)); + if (!msg->findInt32("stride", &mStride)) { + mStride = mWidth; + } + if (!msg->findInt32("slice-height", &mSliceHeight)) { + mSliceHeight = mHeight; + } + + mMaxInputSize = mWidth * mHeight * 4; // room for ARGB8888 + int32_t maxInputSize; + if (msg->findInt32("max-input-size", &maxInputSize) + && (size_t)maxInputSize > mMaxInputSize) { + mMaxInputSize = maxInputSize; + } + + if (!msg->findInt32("color-format", &mColorFormatIn)) { + // default to OMX_COLOR_Format32bitARGB8888 + mColorFormatIn = OMX_COLOR_Format32bitARGB8888; + msg->setInt32("color-format", mColorFormatIn); + } + mColorFormatOut = mColorFormatIn; + + mMaxOutputSize = mWidth * mHeight * 4; // room for ARGB8888 + + AString cacheDir; + if (!msg->findString("cacheDir", &cacheDir)) { + ALOGE("Failed to find cache directory in config message."); + signalError(NAME_NOT_FOUND); + return; + } + + status_t err; + err = mFilter->configure(msg); + if (err != (status_t)OK) { + ALOGE("Failed to configure filter component, err %d", err); + signalError(err); + return; + } + + mInputFormat = new AMessage(); + mInputFormat->setString("mime", mime.c_str()); + mInputFormat->setInt32("stride", mStride); + mInputFormat->setInt32("slice-height", mSliceHeight); + mInputFormat->setInt32("color-format", mColorFormatIn); + mInputFormat->setRect("crop", 0, 0, mStride, mSliceHeight); + mInputFormat->setInt32("width", mWidth); + mInputFormat->setInt32("height", mHeight); + + mOutputFormat = new AMessage(); + mOutputFormat->setString("mime", mime.c_str()); + mOutputFormat->setInt32("stride", mStride); + mOutputFormat->setInt32("slice-height", mSliceHeight); + mOutputFormat->setInt32("color-format", mColorFormatOut); + mOutputFormat->setRect("crop", 0, 0, mStride, mSliceHeight); + mOutputFormat->setInt32("width", mWidth); + mOutputFormat->setInt32("height", mHeight); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatComponentConfigured); + notify->setString("componentName", "MediaFilter"); + notify->setMessage("input-format", mInputFormat); + notify->setMessage("output-format", mOutputFormat); + notify->post(); + mState = CONFIGURED; + ALOGV("Handled kWhatConfigureComponent."); + + sendFormatChange(); +} + +void MediaFilter::onStart() { + CHECK_EQ(mState, CONFIGURED); + + allocateBuffersOnPort(kPortIndexInput); + + allocateBuffersOnPort(kPortIndexOutput); + + status_t err = mFilter->start(); + if (err != (status_t)OK) { + ALOGE("Failed to start filter component, err %d", err); + signalError(err); + return; + } + + mPortEOS[kPortIndexInput] = false; + mPortEOS[kPortIndexOutput] = false; + mInputEOSResult = OK; + mState = STARTED; + + requestFillEmptyInput(); + ALOGV("Handled kWhatStart."); +} + +void MediaFilter::onInputBufferFilled(const sp<AMessage> &msg) { + IOMX::buffer_id bufferID; + CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID)); + BufferInfo *info = findBufferByID(kPortIndexInput, bufferID); + + if (mState != STARTED) { + // we're not running, so we'll just keep that buffer... + info->mStatus = BufferInfo::OWNED_BY_US; + return; + } + + if (info->mGeneration != mGeneration) { + ALOGV("Caught a stale input buffer [ID %d]", bufferID); + // buffer is stale (taken before a flush/shutdown) - repost it + CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US); + postFillThisBuffer(info); + return; + } + + CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM); + info->mStatus = BufferInfo::OWNED_BY_US; + + sp<ABuffer> buffer; + int32_t err = OK; + bool eos = false; + + if (!msg->findBuffer("buffer", &buffer)) { + // these are unfilled buffers returned by client + CHECK(msg->findInt32("err", &err)); + + if (err == OK) { + // buffers with no errors are returned on MediaCodec.flush + ALOGV("saw unfilled buffer (MediaCodec.flush)"); + postFillThisBuffer(info); + return; + } else { + ALOGV("saw error %d instead of an input buffer", err); + eos = true; + } + + buffer.clear(); + } + + int32_t isCSD; + if (buffer != NULL && buffer->meta()->findInt32("csd", &isCSD) + && isCSD != 0) { + // ignore codec-specific data buffers + ALOGW("MediaFilter received a codec-specific data buffer"); + postFillThisBuffer(info); + return; + } + + int32_t tmp; + if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) { + eos = true; + err = ERROR_END_OF_STREAM; + } + + mAvailableInputBuffers.push_back(info); + processBuffers(); + + if (eos) { + mPortEOS[kPortIndexInput] = true; + mInputEOSResult = err; + } + + ALOGV("Handled kWhatInputBufferFilled. [ID %u]", bufferID); +} + +void MediaFilter::onOutputBufferDrained(const sp<AMessage> &msg) { + IOMX::buffer_id bufferID; + CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID)); + BufferInfo *info = findBufferByID(kPortIndexOutput, bufferID); + + if (mState != STARTED) { + // we're not running, so we'll just keep that buffer... + info->mStatus = BufferInfo::OWNED_BY_US; + return; + } + + if (info->mGeneration != mGeneration) { + ALOGV("Caught a stale output buffer [ID %d]", bufferID); + // buffer is stale (taken before a flush/shutdown) - keep it + CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US); + return; + } + + CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM); + info->mStatus = BufferInfo::OWNED_BY_US; + + mAvailableOutputBuffers.push_back(info); + + processBuffers(); + + ALOGV("Handled kWhatOutputBufferDrained. [ID %u]", + bufferID); +} + +void MediaFilter::onShutdown(const sp<AMessage> &msg) { + mGeneration++; + + if (mState != UNINITIALIZED) { + mFilter->reset(); + } + + int32_t keepComponentAllocated; + CHECK(msg->findInt32("keepComponentAllocated", &keepComponentAllocated)); + if (!keepComponentAllocated || mState == UNINITIALIZED) { + mState = UNINITIALIZED; + } else { + mState = INITIALIZED; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatShutdownCompleted); + notify->post(); +} + +void MediaFilter::onFlush() { + mGeneration++; + + mAvailableInputBuffers.clear(); + for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) { + BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i); + info->mStatus = BufferInfo::OWNED_BY_US; + } + mAvailableOutputBuffers.clear(); + for (size_t i = 0; i < mBuffers[kPortIndexOutput].size(); ++i) { + BufferInfo *info = &mBuffers[kPortIndexOutput].editItemAt(i); + info->mStatus = BufferInfo::OWNED_BY_US; + mAvailableOutputBuffers.push_back(info); + } + + mPortEOS[kPortIndexInput] = false; + mPortEOS[kPortIndexOutput] = false; + mInputEOSResult = OK; + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatFlushCompleted); + notify->post(); + ALOGV("Posted kWhatFlushCompleted"); + + // MediaCodec returns all input buffers after flush, so in + // onInputBufferFilled we call postFillThisBuffer on them +} + +void MediaFilter::onSetParameters(const sp<AMessage> &msg) { + CHECK(mState != STARTED); + + status_t err = mFilter->setParameters(msg); + if (err != (status_t)OK) { + ALOGE("setParameters returned err %d", err); + } +} + +void MediaFilter::onCreateInputSurface() { + CHECK(mState == CONFIGURED); + + mGraphicBufferListener = new GraphicBufferListener; + + sp<AMessage> notify = new AMessage(); + notify->setTarget(id()); + status_t err = mGraphicBufferListener->init( + notify, mStride, mSliceHeight, kBufferCountActual); + + if (err != OK) { + ALOGE("Failed to init mGraphicBufferListener: %d", err); + signalError(err); + return; + } + + sp<AMessage> reply = mNotify->dup(); + reply->setInt32("what", CodecBase::kWhatInputSurfaceCreated); + reply->setObject( + "input-surface", + new BufferProducerWrapper( + mGraphicBufferListener->getIGraphicBufferProducer())); + reply->post(); +} + +void MediaFilter::onInputFrameAvailable() { + BufferQueue::BufferItem item = mGraphicBufferListener->getBufferItem(); + sp<GraphicBuffer> buf = mGraphicBufferListener->getBuffer(item); + + // get pointer to graphic buffer + void* bufPtr; + buf->lock(GraphicBuffer::USAGE_SW_READ_OFTEN, &bufPtr); + + // HACK - there is no OMX_COLOR_FORMATTYPE value for RGBA, so the format + // conversion is hardcoded until we add this. + // TODO: check input format and convert only if necessary + // copy RGBA graphic buffer into temporary ARGB input buffer + BufferInfo *inputInfo = new BufferInfo; + inputInfo->mData = new ABuffer(buf->getWidth() * buf->getHeight() * 4); + ALOGV("Copying surface data into temp buffer."); + convertRGBAToARGB( + (uint8_t*)bufPtr, buf->getWidth(), buf->getHeight(), + buf->getStride(), inputInfo->mData->data()); + inputInfo->mBufferID = item.mBuf; + inputInfo->mGeneration = mGeneration; + inputInfo->mOutputFlags = 0; + inputInfo->mStatus = BufferInfo::OWNED_BY_US; + inputInfo->mData->meta()->setInt64("timeUs", item.mTimestamp / 1000); + + mAvailableInputBuffers.push_back(inputInfo); + + mGraphicBufferListener->releaseBuffer(item); + + signalProcessBuffers(); +} + +void MediaFilter::onSignalEndOfInputStream() { + // if using input surface, need to send an EOS output buffer + if (mGraphicBufferListener != NULL) { + Vector<BufferInfo> *outputBufs = &mBuffers[kPortIndexOutput]; + BufferInfo* eosBuf; + bool foundBuf = false; + for (size_t i = 0; i < kBufferCountActual; i++) { + eosBuf = &outputBufs->editItemAt(i); + if (eosBuf->mStatus == BufferInfo::OWNED_BY_US) { + foundBuf = true; + break; + } + } + + if (!foundBuf) { + ALOGE("onSignalEndOfInputStream failed to find an output buffer"); + return; + } + + eosBuf->mOutputFlags = OMX_BUFFERFLAG_EOS; + eosBuf->mGeneration = mGeneration; + eosBuf->mData->setRange(0, 0); + postDrainThisBuffer(eosBuf); + ALOGV("Posted EOS on output buffer %zu", eosBuf->mBufferID); + } + + mPortEOS[kPortIndexOutput] = true; + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", CodecBase::kWhatSignaledInputEOS); + notify->post(); + + ALOGV("Output stream saw EOS."); +} + +} // namespace android diff --git a/media/libstagefright/filters/RSFilter.cpp b/media/libstagefright/filters/RSFilter.cpp new file mode 100644 index 0000000..b569945 --- /dev/null +++ b/media/libstagefright/filters/RSFilter.cpp @@ -0,0 +1,96 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "RSFilter" + +#include <utils/Log.h> + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include "RSFilter.h" + +namespace android { + +RSFilter::RSFilter() { + +} + +RSFilter::~RSFilter() { + +} + +status_t RSFilter::configure(const sp<AMessage> &msg) { + status_t err = SimpleFilter::configure(msg); + if (err != OK) { + return err; + } + + if (!msg->findString("cacheDir", &mCacheDir)) { + ALOGE("Failed to find cache directory in config message."); + return NAME_NOT_FOUND; + } + + sp<RenderScriptWrapper> wrapper; + if (!msg->findObject("rs-wrapper", (sp<RefBase>*)&wrapper)) { + ALOGE("Failed to find RenderScriptWrapper in config message."); + return NAME_NOT_FOUND; + } + + mRS = wrapper->mContext; + mCallback = wrapper->mCallback; + + return OK; +} + +status_t RSFilter::start() { + // 32-bit elements for ARGB8888 + RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS); + + RSC::Type::Builder tb(mRS, e); + tb.setX(mWidth); + tb.setY(mHeight); + RSC::sp<const RSC::Type> t = tb.create(); + + mAllocIn = RSC::Allocation::createTyped(mRS, t); + mAllocOut = RSC::Allocation::createTyped(mRS, t); + + return OK; +} + +void RSFilter::reset() { + mCallback.clear(); + mAllocOut.clear(); + mAllocIn.clear(); + mRS.clear(); +} + +status_t RSFilter::setParameters(const sp<AMessage> &msg) { + return mCallback->handleSetParameters(msg); +} + +status_t RSFilter::processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) { + mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data()); + mCallback->processBuffers(mAllocIn.get(), mAllocOut.get()); + mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data()); + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/RSFilter.h b/media/libstagefright/filters/RSFilter.h new file mode 100644 index 0000000..c5b5074 --- /dev/null +++ b/media/libstagefright/filters/RSFilter.h @@ -0,0 +1,53 @@ +/* + * 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 RS_FILTER_H_ +#define RS_FILTER_H_ + +#include <media/stagefright/RenderScriptWrapper.h> +#include <RenderScript.h> + +#include "SimpleFilter.h" + +namespace android { + +struct AString; + +struct RSFilter : public SimpleFilter { +public: + RSFilter(); + + virtual status_t configure(const sp<AMessage> &msg); + virtual status_t start(); + virtual void reset(); + virtual status_t setParameters(const sp<AMessage> &msg); + virtual status_t processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer); + +protected: + virtual ~RSFilter(); + +private: + AString mCacheDir; + sp<RenderScriptWrapper::RSFilterCallback> mCallback; + RSC::sp<RSC::RS> mRS; + RSC::sp<RSC::Allocation> mAllocIn; + RSC::sp<RSC::Allocation> mAllocOut; +}; + +} // namespace android + +#endif // RS_FILTER_H_ diff --git a/media/libstagefright/filters/SaturationFilter.cpp b/media/libstagefright/filters/SaturationFilter.cpp new file mode 100644 index 0000000..ba5f75a --- /dev/null +++ b/media/libstagefright/filters/SaturationFilter.cpp @@ -0,0 +1,99 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "SaturationFilter" + +#include <utils/Log.h> + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include "SaturationFilter.h" + +namespace android { + +status_t SaturationFilter::configure(const sp<AMessage> &msg) { + status_t err = SimpleFilter::configure(msg); + if (err != OK) { + return err; + } + + if (!msg->findString("cacheDir", &mCacheDir)) { + ALOGE("Failed to find cache directory in config message."); + return NAME_NOT_FOUND; + } + + return OK; +} + +status_t SaturationFilter::start() { + // TODO: use a single RS context object for entire application + mRS = new RSC::RS(); + + if (!mRS->init(mCacheDir.c_str())) { + ALOGE("Failed to initialize RenderScript context."); + return NO_INIT; + } + + // 32-bit elements for ARGB8888 + RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS); + + RSC::Type::Builder tb(mRS, e); + tb.setX(mWidth); + tb.setY(mHeight); + RSC::sp<const RSC::Type> t = tb.create(); + + mAllocIn = RSC::Allocation::createTyped(mRS, t); + mAllocOut = RSC::Allocation::createTyped(mRS, t); + + mScript = new ScriptC_saturationARGB(mRS); + + mScript->set_gSaturation(mSaturation); + + return OK; +} + +void SaturationFilter::reset() { + mScript.clear(); + mAllocOut.clear(); + mAllocIn.clear(); + mRS.clear(); +} + +status_t SaturationFilter::setParameters(const sp<AMessage> &msg) { + sp<AMessage> params; + CHECK(msg->findMessage("params", ¶ms)); + + float saturation; + if (params->findFloat("saturation", &saturation)) { + mSaturation = saturation; + } + + return OK; +} + +status_t SaturationFilter::processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) { + mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data()); + mScript->forEach_root(mAllocIn, mAllocOut); + mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data()); + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/SaturationFilter.h b/media/libstagefright/filters/SaturationFilter.h new file mode 100644 index 0000000..0545021 --- /dev/null +++ b/media/libstagefright/filters/SaturationFilter.h @@ -0,0 +1,52 @@ +/* + * 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 SATURATION_FILTER_H_ +#define SATURATION_FILTER_H_ + +#include <RenderScript.h> + +#include "ScriptC_saturationARGB.h" +#include "SimpleFilter.h" + +namespace android { + +struct SaturationFilter : public SimpleFilter { +public: + SaturationFilter() : mSaturation(1.f) {}; + + virtual status_t configure(const sp<AMessage> &msg); + virtual status_t start(); + virtual void reset(); + virtual status_t setParameters(const sp<AMessage> &msg); + virtual status_t processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer); + +protected: + virtual ~SaturationFilter() {}; + +private: + AString mCacheDir; + RSC::sp<RSC::RS> mRS; + RSC::sp<RSC::Allocation> mAllocIn; + RSC::sp<RSC::Allocation> mAllocOut; + RSC::sp<ScriptC_saturationARGB> mScript; + float mSaturation; +}; + +} // namespace android + +#endif // SATURATION_FILTER_H_ diff --git a/media/libstagefright/filters/SimpleFilter.cpp b/media/libstagefright/filters/SimpleFilter.cpp new file mode 100644 index 0000000..6c1ca2c --- /dev/null +++ b/media/libstagefright/filters/SimpleFilter.cpp @@ -0,0 +1,39 @@ +/* + * 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. + */ + +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include "SimpleFilter.h" + +namespace android { + +status_t SimpleFilter::configure(const sp<AMessage> &msg) { + CHECK(msg->findInt32("width", &mWidth)); + CHECK(msg->findInt32("height", &mHeight)); + if (!msg->findInt32("stride", &mStride)) { + mStride = mWidth; + } + if (!msg->findInt32("slice-height", &mSliceHeight)) { + mSliceHeight = mHeight; + } + CHECK(msg->findInt32("color-format", &mColorFormatIn)); + mColorFormatOut = mColorFormatIn; + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/SimpleFilter.h b/media/libstagefright/filters/SimpleFilter.h new file mode 100644 index 0000000..4cd37ef --- /dev/null +++ b/media/libstagefright/filters/SimpleFilter.h @@ -0,0 +1,52 @@ +/* + * 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 SIMPLE_FILTER_H_ +#define SIMPLE_FILTER_H_ + +#include <stdint.h> +#include <utils/Errors.h> +#include <utils/RefBase.h> + +struct ABuffer; +struct AMessage; + +namespace android { + +struct SimpleFilter : public RefBase { +public: + SimpleFilter() : mWidth(0), mHeight(0), mStride(0), mSliceHeight(0), + mColorFormatIn(0), mColorFormatOut(0) {}; + + virtual status_t configure(const sp<AMessage> &msg); + + virtual status_t start() = 0; + virtual void reset() = 0; + virtual status_t setParameters(const sp<AMessage> &msg) = 0; + virtual status_t processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) = 0; + +protected: + int32_t mWidth, mHeight; + int32_t mStride, mSliceHeight; + int32_t mColorFormatIn, mColorFormatOut; + + virtual ~SimpleFilter() {}; +}; + +} // namespace android + +#endif // SIMPLE_FILTER_H_ diff --git a/media/libstagefright/filters/ZeroFilter.cpp b/media/libstagefright/filters/ZeroFilter.cpp new file mode 100644 index 0000000..3f1243c --- /dev/null +++ b/media/libstagefright/filters/ZeroFilter.cpp @@ -0,0 +1,57 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "ZeroFilter" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> + +#include "ZeroFilter.h" + +namespace android { + +status_t ZeroFilter::setParameters(const sp<AMessage> &msg) { + sp<AMessage> params; + CHECK(msg->findMessage("params", ¶ms)); + + int32_t invert; + if (params->findInt32("invert", &invert)) { + mInvertData = (invert != 0); + } + + return OK; +} + +status_t ZeroFilter::processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) { + // assuming identical input & output buffers, since we're a copy filter + if (mInvertData) { + uint32_t* src = (uint32_t*)srcBuffer->data(); + uint32_t* dest = (uint32_t*)outBuffer->data(); + for (size_t i = 0; i < srcBuffer->size() / 4; ++i) { + *(dest++) = *(src++) ^ 0xFFFFFFFF; + } + } else { + memcpy(outBuffer->data(), srcBuffer->data(), srcBuffer->size()); + } + outBuffer->setRange(0, srcBuffer->size()); + + return OK; +} + +} // namespace android diff --git a/media/libstagefright/filters/ZeroFilter.h b/media/libstagefright/filters/ZeroFilter.h new file mode 100644 index 0000000..bd34dfb --- /dev/null +++ b/media/libstagefright/filters/ZeroFilter.h @@ -0,0 +1,43 @@ +/* + * 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 ZERO_FILTER_H_ +#define ZERO_FILTER_H_ + +#include "SimpleFilter.h" + +namespace android { + +struct ZeroFilter : public SimpleFilter { +public: + ZeroFilter() : mInvertData(false) {}; + + virtual status_t start() { return OK; }; + virtual void reset() {}; + virtual status_t setParameters(const sp<AMessage> &msg); + virtual status_t processBuffers( + const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer); + +protected: + virtual ~ZeroFilter() {}; + +private: + bool mInvertData; +}; + +} // namespace android + +#endif // ZERO_FILTER_H_ diff --git a/media/libstagefright/filters/saturation.rs b/media/libstagefright/filters/saturation.rs new file mode 100644 index 0000000..2c867ac --- /dev/null +++ b/media/libstagefright/filters/saturation.rs @@ -0,0 +1,40 @@ +// Sample script for RGB888 support (compare to saturationARGB.rs) +/* + * 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. + */ + +#pragma version(1) +#pragma rs java_package_name(com.android.rs.cppbasic) +#pragma rs_fp_relaxed + +const static float3 gMonoMult = {0.299f, 0.587f, 0.114f}; + +// global variables (parameters accessible to application code) +float gSaturation = 1.0f; + +void root(const uchar3 *v_in, uchar3 *v_out) { + // scale 0-255 uchar to 0-1.0 float + float3 in = {v_in->r * 0.003921569f, v_in->g * 0.003921569f, + v_in->b * 0.003921569f}; + + // apply saturation filter + float3 result = dot(in, gMonoMult); + result = mix(result, in, gSaturation); + + // convert to uchar, copied from rsPackColorTo8888 + v_out->x = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f); + v_out->y = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f); + v_out->z = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f); +} diff --git a/media/libstagefright/filters/saturationARGB.rs b/media/libstagefright/filters/saturationARGB.rs new file mode 100644 index 0000000..1de9dd8 --- /dev/null +++ b/media/libstagefright/filters/saturationARGB.rs @@ -0,0 +1,40 @@ +/* + * 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. + */ + +#pragma version(1) +#pragma rs java_package_name(com.android.rs.cppbasic) +#pragma rs_fp_relaxed + +const static float3 gMonoMult = {0.299f, 0.587f, 0.114f}; + +// global variables (parameters accessible to application code) +float gSaturation = 1.0f; + +void root(const uchar4 *v_in, uchar4 *v_out) { + v_out->x = v_in->x; // don't modify A + + // get RGB, scale 0-255 uchar to 0-1.0 float + float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f, + v_in->w * 0.003921569f}; + + // apply saturation filter + float3 result = dot(rgb, gMonoMult); + result = mix(result, rgb, gSaturation); + + v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f); + v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f); + v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f); +} diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp index a605595..4cf3819 100644 --- a/media/libstagefright/mpeg2ts/ESQueue.cpp +++ b/media/libstagefright/mpeg2ts/ESQueue.cpp @@ -617,8 +617,6 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAAC() { // having to interpolate. // The final AAC frame may well extend into the next RangeInfo but // that's ok. - // TODO: the logic commented above is skipped because codec cannot take - // arbitrary sized input buffers; size_t offset = 0; while (offset < info.mLength) { if (offset + 7 > mBuffer->size()) { @@ -683,12 +681,9 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAAC() { size_t headerSize __unused = protection_absent ? 7 : 9; offset += aac_frame_length; - // TODO: move back to concatenation when codec can support arbitrary input buffers. - // For now only queue a single buffer - break; } - int64_t timeUs = fetchTimestampAAC(offset); + int64_t timeUs = fetchTimestamp(offset); sp<ABuffer> accessUnit = new ABuffer(offset); memcpy(accessUnit->data(), mBuffer->data(), offset); @@ -735,45 +730,6 @@ int64_t ElementaryStreamQueue::fetchTimestamp(size_t size) { return timeUs; } -// TODO: avoid interpolating timestamps once codec supports arbitrary sized input buffers -int64_t ElementaryStreamQueue::fetchTimestampAAC(size_t size) { - int64_t timeUs = -1; - bool first = true; - - size_t samplesize = size; - while (size > 0) { - CHECK(!mRangeInfos.empty()); - - RangeInfo *info = &*mRangeInfos.begin(); - - if (first) { - timeUs = info->mTimestampUs; - first = false; - } - - if (info->mLength > size) { - int32_t sampleRate; - CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate)); - info->mLength -= size; - size_t numSamples = 1024 * size / samplesize; - info->mTimestampUs += numSamples * 1000000ll / sampleRate; - size = 0; - } else { - size -= info->mLength; - - mRangeInfos.erase(mRangeInfos.begin()); - info = NULL; - } - - } - - if (timeUs == 0ll) { - ALOGV("Returning 0 timestamp"); - } - - return timeUs; -} - struct NALPosition { size_t nalOffset; size_t nalSize; diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h index eb4b1c9..45b4624 100644 --- a/media/libstagefright/mpeg2ts/ESQueue.h +++ b/media/libstagefright/mpeg2ts/ESQueue.h @@ -77,7 +77,6 @@ private: // consume a logical (compressed) access unit of size "size", // returns its timestamp in us (or -1 if no time information). int64_t fetchTimestamp(size_t size); - int64_t fetchTimestampAAC(size_t size); DISALLOW_EVIL_CONSTRUCTORS(ElementaryStreamQueue); }; diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp index 069961b..737f144 100644 --- a/media/libstagefright/webm/WebmWriter.cpp +++ b/media/libstagefright/webm/WebmWriter.cpp @@ -80,38 +80,6 @@ WebmWriter::WebmWriter(int fd) mCuePoints); } -WebmWriter::WebmWriter(const char *filename) - : mInitCheck(NO_INIT), - mTimeCodeScale(1000000), - mStartTimestampUs(0), - mStartTimeOffsetMs(0), - mSegmentOffset(0), - mSegmentDataStart(0), - mInfoOffset(0), - mInfoSize(0), - mTracksOffset(0), - mCuesOffset(0), - mPaused(false), - mStarted(false), - mIsFileSizeLimitExplicitlyRequested(false), - mIsRealTimeRecording(false), - mStreamableFile(true), - mEstimatedCuesSize(0) { - mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR); - if (mFd >= 0) { - ALOGV("fd %d; flags: %o", mFd, fcntl(mFd, F_GETFL, 0)); - mInitCheck = OK; - } - mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack); - mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack); - mSinkThread = new WebmFrameSinkThread( - mFd, - mSegmentDataStart, - mStreams[kVideoIndex].mSink, - mStreams[kAudioIndex].mSink, - mCuePoints); -} - // static sp<WebmElement> WebmWriter::videoTrack(const sp<MetaData>& md) { int32_t width, height; diff --git a/media/libstagefright/webm/WebmWriter.h b/media/libstagefright/webm/WebmWriter.h index 36b6965..4ad770e 100644 --- a/media/libstagefright/webm/WebmWriter.h +++ b/media/libstagefright/webm/WebmWriter.h @@ -37,7 +37,6 @@ namespace android { class WebmWriter : public MediaWriter { public: WebmWriter(int fd); - WebmWriter(const char *filename); ~WebmWriter() { reset(); } diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk index 3a280f0..f1b84ad 100644 --- a/media/mediaserver/Android.mk +++ b/media/mediaserver/Android.mk @@ -11,7 +11,7 @@ endif include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - main_mediaserver.cpp + main_mediaserver.cpp LOCAL_SHARED_LIBRARIES := \ libaudioflinger \ diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp index ed00b72..3124e4a 100644 --- a/media/ndk/NdkMediaCodec.cpp +++ b/media/ndk/NdkMediaCodec.cpp @@ -352,7 +352,8 @@ media_status_t AMediaCodec_releaseOutputBufferAtTime( } //EXPORT -media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback, void *userdata) { +media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback, + void *userdata) { mData->mCallback = callback; mData->mCallbackUserData = userdata; return AMEDIA_OK; diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp index db57d0b..0ecd64f 100644 --- a/media/ndk/NdkMediaExtractor.cpp +++ b/media/ndk/NdkMediaExtractor.cpp @@ -70,7 +70,8 @@ media_status_t AMediaExtractor_delete(AMediaExtractor *mData) { } EXPORT -media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset, off64_t length) { +media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset, + off64_t length) { ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length); return translate_error(mData->mImpl->setDataSource(fd, offset, length)); } diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 993db73..9ad437a 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -185,7 +185,8 @@ AudioFlinger::AudioFlinger() char value[PROPERTY_VALUE_MAX]; bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); if (doLog) { - mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); + mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", + MemoryHeapBase::READ_ONLY); } #ifdef TEE_SINK @@ -401,6 +402,9 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args) String8 result(kClientLockedString); write(fd, result.string(), result.size()); } + + EffectDumpEffects(fd); + dumpClients(fd, args); if (clientLocked) { mClientLock.unlock(); diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index fd28ea1..0d4b358 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -430,6 +430,10 @@ void AudioMixer::setLog(NBLog::Writer *log) mState.mLog = log; } +static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) { + return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; +} + int AudioMixer::getTrackName(audio_channel_mask_t channelMask, audio_format_t format, int sessionId) { @@ -492,24 +496,23 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, t->mInputBufferProvider = NULL; t->mReformatBufferProvider = NULL; t->downmixerBufferProvider = NULL; + t->mPostDownmixReformatBufferProvider = NULL; t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT; t->mFormat = format; - t->mMixerInFormat = kUseFloat && kUseNewMixer - ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; + t->mMixerInFormat = selectMixerInFormat(format); + t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits( AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO); t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask); // Check the downmixing (or upmixing) requirements. - status_t status = initTrackDownmix(t, n); + status_t status = t->prepareForDownmix(); if (status != OK) { ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask); return -1; } - // initTrackDownmix() may change the input format requirement. - // If you desire floating point input to the mixer, it may change - // to integer because the downmixer requires integer to process. + // prepareForDownmix() may change mDownmixRequiresFormat ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat); - prepareTrackForReformat(t, n); + t->prepareForReformat(); mTrackNames |= 1 << n; return TRACK0 + n; } @@ -526,7 +529,7 @@ void AudioMixer::invalidateState(uint32_t mask) } // Called when channel masks have changed for a track name -// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format, +// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format, // which will simplify this logic. bool AudioMixer::setChannelMasks(int name, audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) { @@ -551,21 +554,18 @@ bool AudioMixer::setChannelMasks(int name, // channel masks have changed, does this track need a downmixer? // update to try using our desired format (if we aren't already using it) - const audio_format_t prevMixerInFormat = track.mMixerInFormat; - track.mMixerInFormat = kUseFloat && kUseNewMixer - ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; - const status_t status = initTrackDownmix(&mState.tracks[name], name); + const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat; + const status_t status = mState.tracks[name].prepareForDownmix(); ALOGE_IF(status != OK, - "initTrackDownmix error %d, track channel mask %#x, mixer channel mask %#x", + "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x", status, track.channelMask, track.mMixerChannelMask); - const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat; - if (mixerInFormatChanged) { - prepareTrackForReformat(&track, name); // because of downmixer, track format may change! + if (prevDownmixerFormat != track.mDownmixRequiresFormat) { + track.prepareForReformat(); // because of downmixer, track format may change! } - if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) { - // resampler input format or channels may have changed. + if (track.resampler && mixerChannelCountChanged) { + // resampler channels may have changed. const uint32_t resetToSampleRate = track.sampleRate; delete track.resampler; track.resampler = NULL; @@ -576,99 +576,122 @@ bool AudioMixer::setChannelMasks(int name, return true; } -status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackName) -{ - // Only remix (upmix or downmix) if the track and mixer/device channel masks - // are not the same and not handled internally, as mono -> stereo currently is. - if (pTrack->channelMask != pTrack->mMixerChannelMask - && !(pTrack->channelMask == AUDIO_CHANNEL_OUT_MONO - && pTrack->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) { - return prepareTrackForDownmix(pTrack, trackName); - } - // no remix necessary - unprepareTrackForDownmix(pTrack, trackName); - return NO_ERROR; -} - -void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) { - ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName); +void AudioMixer::track_t::unprepareForDownmix() { + ALOGV("AudioMixer::unprepareForDownmix(%p)", this); - if (pTrack->downmixerBufferProvider != NULL) { + mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; + if (downmixerBufferProvider != NULL) { // this track had previously been configured with a downmixer, delete it ALOGV(" deleting old downmixer"); - delete pTrack->downmixerBufferProvider; - pTrack->downmixerBufferProvider = NULL; - reconfigureBufferProviders(pTrack); + delete downmixerBufferProvider; + downmixerBufferProvider = NULL; + reconfigureBufferProviders(); } else { ALOGV(" nothing to do, no downmixer to delete"); } } -status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName) +status_t AudioMixer::track_t::prepareForDownmix() { - ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask); + ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x", + this, channelMask); // discard the previous downmixer if there was one - unprepareTrackForDownmix(pTrack, trackName); + unprepareForDownmix(); + // Only remix (upmix or downmix) if the track and mixer/device channel masks + // are not the same and not handled internally, as mono -> stereo currently is. + if (channelMask == mMixerChannelMask + || (channelMask == AUDIO_CHANNEL_OUT_MONO + && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) { + return NO_ERROR; + } if (DownmixerBufferProvider::isMultichannelCapable()) { - DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(pTrack->channelMask, - pTrack->mMixerChannelMask, - AUDIO_FORMAT_PCM_16_BIT /* TODO: use pTrack->mMixerInFormat, now only PCM 16 */, - pTrack->sampleRate, pTrack->sessionId, kCopyBufferFrameCount); + DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask, + mMixerChannelMask, + AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */, + sampleRate, sessionId, kCopyBufferFrameCount); if (pDbp->isValid()) { // if constructor completed properly - pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix - pTrack->downmixerBufferProvider = pDbp; - reconfigureBufferProviders(pTrack); + mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix + downmixerBufferProvider = pDbp; + reconfigureBufferProviders(); return NO_ERROR; } delete pDbp; } // Effect downmixer does not accept the channel conversion. Let's use our remixer. - RemixBufferProvider* pRbp = new RemixBufferProvider(pTrack->channelMask, - pTrack->mMixerChannelMask, pTrack->mMixerInFormat, kCopyBufferFrameCount); + RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask, + mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount); // Remix always finds a conversion whereas Downmixer effect above may fail. - pTrack->downmixerBufferProvider = pRbp; - reconfigureBufferProviders(pTrack); + downmixerBufferProvider = pRbp; + reconfigureBufferProviders(); return NO_ERROR; } -void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) { - ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName); - if (pTrack->mReformatBufferProvider != NULL) { - delete pTrack->mReformatBufferProvider; - pTrack->mReformatBufferProvider = NULL; - reconfigureBufferProviders(pTrack); +void AudioMixer::track_t::unprepareForReformat() { + ALOGV("AudioMixer::unprepareForReformat(%p)", this); + bool requiresReconfigure = false; + if (mReformatBufferProvider != NULL) { + delete mReformatBufferProvider; + mReformatBufferProvider = NULL; + requiresReconfigure = true; + } + if (mPostDownmixReformatBufferProvider != NULL) { + delete mPostDownmixReformatBufferProvider; + mPostDownmixReformatBufferProvider = NULL; + requiresReconfigure = true; + } + if (requiresReconfigure) { + reconfigureBufferProviders(); } } -status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName) +status_t AudioMixer::track_t::prepareForReformat() { - ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat); - // discard the previous reformatter if there was one - unprepareTrackForReformat(pTrack, trackName); - // only configure reformatter if needed - if (pTrack->mFormat != pTrack->mMixerInFormat) { - pTrack->mReformatBufferProvider = new ReformatBufferProvider( - audio_channel_count_from_out_mask(pTrack->channelMask), - pTrack->mFormat, pTrack->mMixerInFormat, + ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat); + // discard previous reformatters + unprepareForReformat(); + // only configure reformatters as needed + const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID + ? mDownmixRequiresFormat : mMixerInFormat; + bool requiresReconfigure = false; + if (mFormat != targetFormat) { + mReformatBufferProvider = new ReformatBufferProvider( + audio_channel_count_from_out_mask(channelMask), + mFormat, + targetFormat, kCopyBufferFrameCount); - reconfigureBufferProviders(pTrack); + requiresReconfigure = true; + } + if (targetFormat != mMixerInFormat) { + mPostDownmixReformatBufferProvider = new ReformatBufferProvider( + audio_channel_count_from_out_mask(mMixerChannelMask), + targetFormat, + mMixerInFormat, + kCopyBufferFrameCount); + requiresReconfigure = true; + } + if (requiresReconfigure) { + reconfigureBufferProviders(); } return NO_ERROR; } -void AudioMixer::reconfigureBufferProviders(track_t* pTrack) +void AudioMixer::track_t::reconfigureBufferProviders() { - pTrack->bufferProvider = pTrack->mInputBufferProvider; - if (pTrack->mReformatBufferProvider) { - pTrack->mReformatBufferProvider->setBufferProvider(pTrack->bufferProvider); - pTrack->bufferProvider = pTrack->mReformatBufferProvider; + bufferProvider = mInputBufferProvider; + if (mReformatBufferProvider) { + mReformatBufferProvider->setBufferProvider(bufferProvider); + bufferProvider = mReformatBufferProvider; + } + if (downmixerBufferProvider) { + downmixerBufferProvider->setBufferProvider(bufferProvider); + bufferProvider = downmixerBufferProvider; } - if (pTrack->downmixerBufferProvider) { - pTrack->downmixerBufferProvider->setBufferProvider(pTrack->bufferProvider); - pTrack->bufferProvider = pTrack->downmixerBufferProvider; + if (mPostDownmixReformatBufferProvider) { + mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider); + bufferProvider = mPostDownmixReformatBufferProvider; } } @@ -687,9 +710,9 @@ void AudioMixer::deleteTrackName(int name) delete track.resampler; track.resampler = NULL; // delete the downmixer - unprepareTrackForDownmix(&mState.tracks[name], name); + mState.tracks[name].unprepareForDownmix(); // delete the reformatter - unprepareTrackForReformat(&mState.tracks[name], name); + mState.tracks[name].unprepareForReformat(); mTrackNames &= ~(1<<name); } @@ -828,7 +851,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format); track.mFormat = format; ALOGV("setParameter(TRACK, FORMAT, %#x)", format); - prepareTrackForReformat(&track, name); + track.prepareForReformat(); invalidateState(1 << name); } } break; @@ -1032,10 +1055,13 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider if (mState.tracks[name].mReformatBufferProvider != NULL) { mState.tracks[name].mReformatBufferProvider->reset(); } else if (mState.tracks[name].downmixerBufferProvider != NULL) { + mState.tracks[name].downmixerBufferProvider->reset(); + } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) { + mState.tracks[name].mPostDownmixReformatBufferProvider->reset(); } mState.tracks[name].mInputBufferProvider = bufferProvider; - reconfigureBufferProviders(&mState.tracks[name]); + mState.tracks[name].reconfigureBufferProviders(); } diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index f4f142b..c5df08a 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -127,10 +127,16 @@ public: size_t getUnreleasedFrames(int name) const; static inline bool isValidPcmTrackFormat(audio_format_t format) { - return format == AUDIO_FORMAT_PCM_16_BIT || - format == AUDIO_FORMAT_PCM_24_BIT_PACKED || - format == AUDIO_FORMAT_PCM_32_BIT || - format == AUDIO_FORMAT_PCM_FLOAT; + switch (format) { + case AUDIO_FORMAT_PCM_8_BIT: + case AUDIO_FORMAT_PCM_16_BIT: + case AUDIO_FORMAT_PCM_24_BIT_PACKED: + case AUDIO_FORMAT_PCM_32_BIT: + case AUDIO_FORMAT_PCM_FLOAT: + return true; + default: + return false; + } } private: @@ -205,17 +211,34 @@ private: int32_t* auxBuffer; // 16-byte boundary + + /* Buffer providers are constructed to translate the track input data as needed. + * + * 1) mInputBufferProvider: The AudioTrack buffer provider. + * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to + * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer + * requires reformat. For example, it may convert floating point input to + * PCM_16_bit if that's required by the downmixer. + * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match + * the number of channels required by the mixer sink. + * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from + * the downmixer requirements to the mixer engine input requirements. + */ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. + CopyBufferProvider* mPostDownmixReformatBufferProvider; + // 16-byte boundary int32_t sessionId; - // 16-byte boundary audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) audio_format_t mFormat; // input track format audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) // each track must be converted to this format. + audio_format_t mDownmixRequiresFormat; // required downmixer format + // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary + // AUDIO_FORMAT_INVALID if no required format float mVolume[MAX_NUM_VOLUMES]; // floating point set volume float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume @@ -225,7 +248,6 @@ private: float mPrevAuxLevel; // floating point prev aux level float mAuxInc; // floating point aux increment - // 16-byte boundary audio_channel_mask_t mMixerChannelMask; uint32_t mMixerChannelCount; @@ -236,6 +258,12 @@ private: void adjustVolumeRamp(bool aux, bool useFloat = false); size_t getUnreleasedFrames() const { return resampler != NULL ? resampler->getUnreleasedFrames() : 0; }; + + status_t prepareForDownmix(); + void unprepareForDownmix(); + status_t prepareForReformat(); + void unprepareForReformat(); + void reconfigureBufferProviders(); }; typedef void (*process_hook_t)(state_t* state, int64_t pts); @@ -382,14 +410,6 @@ private: bool setChannelMasks(int name, audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask); - // TODO: remove unused trackName/trackNum from functions below. - static status_t initTrackDownmix(track_t* pTrack, int trackName); - static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); - static void unprepareTrackForDownmix(track_t* pTrack, int trackName); - static status_t prepareTrackForReformat(track_t* pTrack, int trackNum); - static void unprepareTrackForReformat(track_t* pTrack, int trackName); - static void reconfigureBufferProviders(track_t* pTrack); - static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h index f3718b6..a9c84de 100644 --- a/services/audioflinger/AudioResamplerFirGen.h +++ b/services/audioflinger/AudioResamplerFirGen.h @@ -204,7 +204,8 @@ struct I0ATerm { template <> struct I0ATerm<0> { // 1/sqrt(2*PI); - static const CONSTEXPR double value = 0.398942280401432677939946059934381868475858631164934657665925; + static const CONSTEXPR double value = + 0.398942280401432677939946059934381868475858631164934657665925; }; #if USE_HORNERS_METHOD diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h index efc8055..1118bf8 100644 --- a/services/audioflinger/AudioResamplerFirProcess.h +++ b/services/audioflinger/AudioResamplerFirProcess.h @@ -174,7 +174,8 @@ struct InterpNull { * Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase. */ -template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP> +template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, + typename TINTERP> static inline void ProcessBase(TO* const out, size_t count, @@ -268,7 +269,8 @@ void Process(TO* const out, TINTERP lerpP, const TO* const volumeLR) { - ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR); + ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, + volumeLR); } /* diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp index 2678cbf..141a79e 100644 --- a/services/audioflinger/FastMixer.cpp +++ b/services/audioflinger/FastMixer.cpp @@ -629,7 +629,8 @@ void FastMixerDumpState::dump(int fd) const left.sample(tail[i]); right.sample(tail[n - (i + 1)]); } - dprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n" + dprintf(fd, " Distribution of mix cycle times in ms for the tails " + "(> ~3 stddev outliers):\n" " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n" " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n", left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6, diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index ee48276..902d5e4 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -255,7 +255,7 @@ public: class Buffer : public AudioBufferProvider::Buffer { public: - int16_t *mBuffer; + void *mBuffer; }; OutputTrack(PlaybackThread *thread, @@ -271,7 +271,7 @@ public: AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); virtual void stop(); - bool write(int16_t* data, uint32_t frames); + bool write(void* data, uint32_t frames); bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } bool isActive() const { return mActive; } const wp<ThreadBase>& thread() const { return mThread; } diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 51025fe..15dd408 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -172,6 +172,18 @@ static int sFastTrackMultiplier = kFastTrackMultiplier; // and that all "fast" AudioRecord clients read from. In either case, the size can be small. static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; +// Returns the source frames needed to resample to destination frames. This is not a precise +// value and depends on the resampler (and possibly how it handles rounding internally). +// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which +// may not be a true if the resampler is asynchronous. +static inline size_t sourceFramesNeeded( + uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) { + // +1 for rounding - always do this even if matched ratio + // +1 for additional sample needed for interpolation + return srcSampleRate == dstSampleRate ? dstFramesRequired : + size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); +} + // ---------------------------------------------------------------------------- static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; @@ -314,6 +326,64 @@ void CpuStats::sample(const String8 &title // ThreadBase // ---------------------------------------------------------------------------- +// static +const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) +{ + switch (type) { + case MIXER: + return "MIXER"; + case DIRECT: + return "DIRECT"; + case DUPLICATING: + return "DUPLICATING"; + case RECORD: + return "RECORD"; + case OFFLOAD: + return "OFFLOAD"; + default: + return "unknown"; + } +} + +static String8 outputFlagsToString(audio_output_flags_t flags) +{ + static const struct mapping { + audio_output_flags_t mFlag; + const char * mString; + } mappings[] = { + AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", + AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", + AUDIO_OUTPUT_FLAG_FAST, "FAST", + AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD", + AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", + AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", + AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last + }; + String8 result; + audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; + const mapping *entry; + for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { + allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); + if (flags & entry->mFlag) { + if (!result.isEmpty()) { + result.append("|"); + } + result.append(entry->mString); + } + } + if (flags & ~allFlags) { + if (!result.isEmpty()) { + result.append("|"); + } + result.appendFormat("0x%X", flags & ~allFlags); + } + if (result.isEmpty()) { + result.append(entry->mString); + } + return result; +} + AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, type_t type) : Thread(false /*canCallJava*/), @@ -577,20 +647,21 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u bool locked = AudioFlinger::dumpTryLock(mLock); if (!locked) { - dprintf(fd, "thread %p maybe dead locked\n", this); + dprintf(fd, "thread %p may be deadlocked\n", this); } dprintf(fd, " I/O handle: %d\n", mId); dprintf(fd, " TID: %d\n", getTid()); dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); - dprintf(fd, " Sample rate: %u\n", mSampleRate); + dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); dprintf(fd, " HAL frame count: %zu\n", mFrameCount); + dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); - dprintf(fd, " Channel Count: %u\n", mChannelCount); - dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, + dprintf(fd, " Channel count: %u\n", mChannelCount); + dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, channelMaskToString(mChannelMask, mType != RECORD).string()); - dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); - dprintf(fd, " Frame size: %zu\n", mFrameSize); + dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); + dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); dprintf(fd, " Pending config events:"); size_t numConfig = mConfigEvents.size(); if (numConfig) { @@ -1315,7 +1386,7 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) { - dprintf(fd, "\nOutput thread %p:\n", this); + dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); dprintf(fd, " Total writes: %d\n", mNumWrites); @@ -1326,6 +1397,10 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); + AudioStreamOut *output = mOutput; + audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; + String8 flagsAsString = outputFlagsToString(flags); + dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); dumpBase(fd, args); } @@ -1861,6 +1936,22 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() } } + if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { + // For best precision, we use float instead of the associated output + // device format (typically PCM 16 bit). + + mFormat = AUDIO_FORMAT_PCM_FLOAT; + mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); + mBufferSize = mFrameSize * mFrameCount; + + // TODO: We currently use the associated output device channel mask and sample rate. + // (1) Perhaps use the ORed channel mask of all downstream MixerThreads + // (if a valid mask) to avoid premature downmix. + // (2) Perhaps use the maximum sample rate of all downstream MixerThreads + // instead of the output device sample rate to avoid loss of high frequency information. + // This may need to be updated as MixerThread/OutputTracks are added and not here. + } + // Calculate size of normal sink buffer relative to the HAL output buffer size double multiplier = 1.0; if (mType == MIXER && (kUseFastMixer == FastMixer_Static || @@ -2137,6 +2228,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() } else { bytesWritten = framesWritten; } + mLatchDValid = false; status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); if (status == NO_ERROR) { size_t totalFramesWritten = mNormalSink->framesWritten(); @@ -2640,7 +2732,9 @@ bool AudioFlinger::PlaybackThread::threadLoop() } } else { + ATRACE_BEGIN("sleep"); usleep(sleepTime); + ATRACE_END(); } } @@ -2800,6 +2894,12 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud mNormalFrameCount); mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); + if (type == DUPLICATING) { + // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks + // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). + // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. + return; + } // create an NBAIO sink for the HAL output stream, and negotiate mOutputSink = new AudioStreamOutSink(output->stream); size_t numCounterOffers = 0; @@ -2841,6 +2941,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud NBAIO_Format format = mOutputSink->format(); NBAIO_Format origformat = format; // adjust format to match that of the Fast Mixer + ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); format.mFormat = fastMixerFormat; format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; @@ -3386,8 +3487,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if (sr == mSampleRate) { desiredFrames = mNormalFrameCount; } else { - // +1 for rounding and +1 for additional sample needed for interpolation - desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; + desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); // add frames already consumed but not yet released by the resampler // because mAudioTrackServerProxy->framesReady() will include these frames desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); @@ -3405,6 +3505,23 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } size_t framesReady = track->framesReady(); + if (ATRACE_ENABLED()) { + // I wish we had formatted trace names + char traceName[16]; + strcpy(traceName, "nRdy"); + int name = track->name(); + if (AudioMixer::TRACK0 <= name && + name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { + name -= AudioMixer::TRACK0; + traceName[4] = (name / 10) + '0'; + traceName[5] = (name % 10) + '0'; + } else { + traceName[4] = '?'; + traceName[5] = '?'; + } + traceName[6] = '\0'; + ATRACE_INT(traceName, framesReady); + } if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { @@ -4797,16 +4914,8 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() { - // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT - // for delivery downstream as needed. This in-place conversion is safe as - // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format - // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). - if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { - memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, - mSinkBuffer, mFormat, writeFrames * mChannelCount); - } for (size_t i = 0; i < outputTracks.size(); i++) { - outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); + outputTracks[i]->write(mSinkBuffer, writeFrames); } mStandby = false; return (ssize_t)mSinkBufferSize; @@ -4833,25 +4942,26 @@ void AudioFlinger::DuplicatingThread::clearOutputTracks() void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) { Mutex::Autolock _l(mLock); - // FIXME explain this formula - size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); - // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat - // due to current usage case and restrictions on the AudioBufferProvider. - // Actual buffer conversion is done in threadLoop_write(). - // - // TODO: This may change in the future, depending on multichannel - // (and non int16_t*) support on AF::PlaybackThread::OutputTrack - OutputTrack *outputTrack = new OutputTrack(thread, + // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. + // Adjust for thread->sampleRate() to determine minimum buffer frame count. + // Then triple buffer because Threads do not run synchronously and may not be clock locked. + const size_t frameCount = + 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); + // TODO: Consider asynchronous sample rate conversion to handle clock disparity + // from different OutputTracks and their associated MixerThreads (e.g. one may + // nearly empty and the other may be dropping data). + + sp<OutputTrack> outputTrack = new OutputTrack(thread, this, mSampleRate, - AUDIO_FORMAT_PCM_16_BIT, + mFormat, mChannelMask, frameCount, IPCThreadState::self()->getCallingUid()); if (outputTrack->cblk() != NULL) { thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); mOutputTracks.add(outputTrack); - ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); + ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); updateWaitTime_l(); } } @@ -5135,7 +5245,9 @@ reacquire_wakelock: // sleep with mutex unlocked if (sleepUs > 0) { + ATRACE_BEGIN("sleep"); usleep(sleepUs); + ATRACE_END(); sleepUs = 0; } @@ -5279,7 +5391,8 @@ reacquire_wakelock: state->mCommand = FastCaptureState::READ_WRITE; #if 0 // FIXME mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? - FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); + FastCaptureDumpState::kSamplingNforLowRamDevice : + FastMixerDumpState::kSamplingN); #endif didModify = true; } @@ -5427,8 +5540,8 @@ reacquire_wakelock: upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, part1); } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, - part1); + downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, + (const int16_t *)src, part1); } dst += part1 * activeTrack->mFrameSize; front += part1; diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index 1088843..a1ac42c 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -32,6 +32,8 @@ public: OFFLOAD // Thread class is OffloadThread }; + static const char *threadTypeToString(type_t type); + ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, type_t type); virtual ~ThreadBase(); @@ -406,6 +408,7 @@ protected: audio_channel_mask_t mChannelMask; uint32_t mChannelCount; size_t mFrameSize; + // not HAL frame size, this is for output sink (to pipe to fast mixer) audio_format_t mFormat; // Source format for Recording and // Sink format for Playback. // Sink format may be different than @@ -1167,7 +1170,8 @@ private: const sp<MemoryDealer> mReadOnlyHeap; // one-time initialization, no locks required - sp<FastCapture> mFastCapture; // non-0 if there is also a fast capture + sp<FastCapture> mFastCapture; // non-0 if there is also + // a fast capture // FIXME audio watchdog thread // contents are not guaranteed to be consistent, no locks required diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index e970036..7757ea2 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -859,6 +859,7 @@ void AudioFlinger::PlaybackThread::Track::reset() if (mState == FLUSHED) { mState = IDLE; } + mPreviousValid = false; } } @@ -1709,14 +1710,13 @@ void AudioFlinger::PlaybackThread::OutputTrack::stop() mActive = false; } -bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) +bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames) { Buffer *pInBuffer; Buffer inBuffer; - uint32_t channelCount = mChannelCount; bool outputBufferFull = false; inBuffer.frameCount = frames; - inBuffer.i16 = data; + inBuffer.raw = data; uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); @@ -1726,13 +1726,17 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr if (thread != 0) { MixerThread *mixerThread = (MixerThread *)thread.get(); if (mFrameCount > frames) { + // For the first write after being inactive, ensure that we have + // enough frames to fill mFrameCount (which should be multiples of + // the minimum buffer requirements of the downstream MixerThread). + // This provides enough frames for the downstream mixer to begin + // (see AudioFlinger::PlaybackThread::Track::isReady()). if (mBufferQueue.size() < kMaxOverFlowBuffers) { uint32_t startFrames = (mFrameCount - frames); pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; + pInBuffer->mBuffer = calloc(1, startFrames * mFrameSize); pInBuffer->frameCount = startFrames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); + pInBuffer->raw = pInBuffer->mBuffer; mBufferQueue.add(pInBuffer); } else { ALOGW("OutputTrack::write() %p no more buffers in queue", this); @@ -1773,20 +1777,20 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; - memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); + memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize); Proxy::Buffer buf; buf.mFrameCount = outFrames; buf.mRaw = NULL; mClientProxy->releaseBuffer(&buf); pInBuffer->frameCount -= outFrames; - pInBuffer->i16 += outFrames * channelCount; + pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize; mOutBuffer.frameCount -= outFrames; - mOutBuffer.i16 += outFrames * channelCount; + mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize; if (pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); - delete [] pInBuffer->mBuffer; + free(pInBuffer->mBuffer); delete pInBuffer; ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); @@ -1802,11 +1806,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr if (thread != 0 && !thread->standby()) { if (mBufferQueue.size() < kMaxOverFlowBuffers) { pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; + pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize); pInBuffer->frameCount = inBuffer.frameCount; - pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * - sizeof(int16_t)); + pInBuffer->raw = pInBuffer->mBuffer; + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize); mBufferQueue.add(pInBuffer); ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); @@ -1817,23 +1820,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr } } - // Calling write() with a 0 length buffer, means that no more data will be written: - // If no more buffers are pending, fill output track buffer to make sure it is started - // by output mixer. - if (frames == 0 && mBufferQueue.size() == 0) { - // FIXME borken, replace by getting framesReady() from proxy - size_t user = 0; // was mCblk->user - if (user < mFrameCount) { - frames = mFrameCount - user; - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[frames * channelCount]; - pInBuffer->frameCount = frames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - } else if (mActive) { - stop(); - } + // Calling write() with a 0 length buffer means that no more data will be written: + // We rely on stop() to set the appropriate flags to allow the remaining frames to play out. + if (frames == 0 && mBufferQueue.size() == 0 && mActive) { + stop(); } return outputBufferFull; @@ -1859,7 +1849,7 @@ void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() for (size_t i = 0; i < size; i++) { Buffer *pBuffer = mBufferQueue.itemAt(i); - delete [] pBuffer->mBuffer; + free(pBuffer->mBuffer); delete pBuffer; } mBufferQueue.clear(); diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh index 9b39e77..e60e6d5 100755 --- a/services/audioflinger/tests/mixer_to_wav_tests.sh +++ b/services/audioflinger/tests/mixer_to_wav_tests.sh @@ -63,8 +63,18 @@ function createwav() { # process__genericResampling # track__Resample / track__genericResample adb shell test-mixer $1 -s 48000 \ + -o /sdcard/tm48000grif.wav \ + sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \ + sine:f,6,6000,19000 chirp:i,4,30000 + adb pull /sdcard/tm48000grif.wav $2 + +# Test: +# process__genericResampling +# track__Resample / track__genericResample + adb shell test-mixer $1 -s 48000 \ -o /sdcard/tm48000gr.wav \ - sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 + sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \ + sine:6,6000,19000 adb pull /sdcard/tm48000gr.wav $2 # Test: diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp index 9a4fad6..8da6245 100644 --- a/services/audioflinger/tests/test-mixer.cpp +++ b/services/audioflinger/tests/test-mixer.cpp @@ -39,7 +39,7 @@ static void usage(const char* name) { fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]" " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]" " (<input-file> | <command>)+\n", name); - fprintf(stderr, " -f enable floating point input track\n"); + fprintf(stderr, " -f enable floating point input track by default\n"); fprintf(stderr, " -m enable floating point mixer output\n"); fprintf(stderr, " -c number of mixer output channels\n"); fprintf(stderr, " -s mixer sample-rate\n"); @@ -47,8 +47,8 @@ static void usage(const char* name) { fprintf(stderr, " -a <aux-buffer-file>\n"); fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n"); fprintf(stderr, " <input-file> is a WAV file\n"); - fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n"); - fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n"); + fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n"); + fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n"); } static int writeFile(const char *filename, const void *buffer, @@ -78,6 +78,18 @@ static int writeFile(const char *filename, const void *buffer, return EXIT_SUCCESS; } +const char *parseFormat(const char *s, bool *useFloat) { + if (!strncmp(s, "f,", 2)) { + *useFloat = true; + return s + 2; + } + if (!strncmp(s, "i,", 2)) { + *useFloat = false; + return s + 2; + } + return s; +} + int main(int argc, char* argv[]) { const char* const progname = argv[0]; bool useInputFloat = false; @@ -88,8 +100,9 @@ int main(int argc, char* argv[]) { std::vector<int> Pvalues; const char* outputFilename = NULL; const char* auxFilename = NULL; - std::vector<int32_t> Names; - std::vector<SignalProvider> Providers; + std::vector<int32_t> names; + std::vector<SignalProvider> providers; + std::vector<audio_format_t> formats; for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) { switch (ch) { @@ -138,54 +151,65 @@ int main(int argc, char* argv[]) { size_t outputFrames = 0; // create providers for each track - Providers.resize(argc); + names.resize(argc); + providers.resize(argc); + formats.resize(argc); for (int i = 0; i < argc; ++i) { static const char chirp[] = "chirp:"; static const char sine[] = "sine:"; static const double kSeconds = 1; + bool useFloat = useInputFloat; if (!strncmp(argv[i], chirp, strlen(chirp))) { std::vector<int> v; + const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat); - parseCSV(argv[i] + strlen(chirp), v); + parseCSV(s, v); if (v.size() == 2) { printf("creating chirp(%d %d)\n", v[0], v[1]); - if (useInputFloat) { - Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds); + if (useFloat) { + providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds); + providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + providers[i].setIncr(Pvalues); } else { fprintf(stderr, "malformed input '%s'\n", argv[i]); } } else if (!strncmp(argv[i], sine, strlen(sine))) { std::vector<int> v; + const char *s = parseFormat(argv[i] + strlen(sine), &useFloat); - parseCSV(argv[i] + strlen(sine), v); + parseCSV(s, v); if (v.size() == 3) { printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]); - if (useInputFloat) { - Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds); + if (useFloat) { + providers[i].setSine<float>(v[0], v[1], v[2], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds); + providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + providers[i].setIncr(Pvalues); } else { fprintf(stderr, "malformed input '%s'\n", argv[i]); } } else { printf("creating filename(%s)\n", argv[i]); if (useInputFloat) { - Providers[i].setFile<float>(argv[i]); + providers[i].setFile<float>(argv[i]); + formats[i] = AUDIO_FORMAT_PCM_FLOAT; } else { - Providers[i].setFile<short>(argv[i]); + providers[i].setFile<short>(argv[i]); + formats[i] = AUDIO_FORMAT_PCM_16_BIT; } - Providers[i].setIncr(Pvalues); + providers[i].setIncr(Pvalues); } // calculate the number of output frames - size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate - / Providers[i].getSampleRate(); + size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate + / providers[i].getSampleRate(); if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames outputFrames = nframes; } @@ -213,22 +237,20 @@ int main(int argc, char* argv[]) { // create the mixer. const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960 AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate); - audio_format_t inputFormat = useInputFloat - ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; audio_format_t mixerFormat = useMixerFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; - float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks + float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks static float f0; // zero // set up the tracks. - for (size_t i = 0; i < Providers.size(); ++i) { - //printf("track %d out of %d\n", i, Providers.size()); - uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels()); + for (size_t i = 0; i < providers.size(); ++i) { + //printf("track %d out of %d\n", i, providers.size()); + uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels()); int32_t name = mixer->getTrackName(channelMask, - inputFormat, AUDIO_SESSION_OUTPUT_MIX); + formats[i], AUDIO_SESSION_OUTPUT_MIX); ALOG_ASSERT(name >= 0); - Names.push_back(name); - mixer->setBufferProvider(name, &Providers[i]); + names[i] = name; + mixer->setBufferProvider(name, &providers[i]); mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, (void *)outputAddr); mixer->setParameter( @@ -240,7 +262,7 @@ int main(int argc, char* argv[]) { name, AudioMixer::TRACK, AudioMixer::FORMAT, - (void *)(uintptr_t)inputFormat); + (void *)(uintptr_t)formats[i]); mixer->setParameter( name, AudioMixer::TRACK, @@ -255,7 +277,7 @@ int main(int argc, char* argv[]) { name, AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, - (void *)(uintptr_t)Providers[i].getSampleRate()); + (void *)(uintptr_t)providers[i].getSampleRate()); if (useRamp) { mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0); mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0); @@ -277,11 +299,11 @@ int main(int argc, char* argv[]) { // pump the mixer to process data. size_t i; for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) { - for (size_t j = 0; j < Names.size(); ++j) { - mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, + for (size_t j = 0; j < names.size(); ++j) { + mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, (char *) outputAddr + i * outputFrameSize); if (auxFilename) { - mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER, + mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER, (char *) auxAddr + i * auxFrameSize); } } diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk index 95f2fef..03438bf 100644 --- a/services/medialog/Android.mk +++ b/services/medialog/Android.mk @@ -10,4 +10,6 @@ LOCAL_MODULE:= libmedialogservice LOCAL_32_BIT_ONLY := true +LOCAL_C_INCLUDES := $(call include-path-for, audio-utils) + include $(BUILD_SHARED_LIBRARY) diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk index e8cbe39..b58e4cd 100644 --- a/tools/resampler_tools/Android.mk +++ b/tools/resampler_tools/Android.mk @@ -1,6 +1,6 @@ # Copyright 2005 The Android Open Source Project # -# Android.mk for resampler_tools +# Android.mk for resampler_tools # |