diff options
37 files changed, 2705 insertions, 609 deletions
diff --git a/camera/Android.mk b/camera/Android.mk index e633450..5cedab0 100644 --- a/camera/Android.mk +++ b/camera/Android.mk @@ -8,6 +8,7 @@ LOCAL_SRC_FILES:= \ Camera.cpp \ CameraMetadata.cpp \ CameraParameters.cpp \ + CameraParameters2.cpp \ ICamera.cpp \ ICameraClient.cpp \ ICameraService.cpp \ diff --git a/camera/CameraParameters2.cpp b/camera/CameraParameters2.cpp new file mode 100644 index 0000000..eac79e1 --- /dev/null +++ b/camera/CameraParameters2.cpp @@ -0,0 +1,381 @@ +/* +** +** Copyright 2008, 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_TAG "CameraParams2" +// #define LOG_NDEBUG 0 +#include <utils/Log.h> + +#include <string.h> +#include <stdlib.h> +#include <camera/CameraParameters2.h> + +namespace android { + +CameraParameters2::CameraParameters2() + : mMap() +{ +} + +CameraParameters2::~CameraParameters2() +{ +} + +String8 CameraParameters2::flatten() const +{ + String8 flattened(""); + size_t size = mMap.size(); + + for (size_t i = 0; i < size; i++) { + String8 k, v; + k = mMap.keyAt(i); + v = mMap.valueAt(i); + + flattened += k; + flattened += "="; + flattened += v; + if (i != size-1) + flattened += ";"; + } + + ALOGV("%s: Flattened params = %s", __FUNCTION__, flattened.string()); + + return flattened; +} + +void CameraParameters2::unflatten(const String8 ¶ms) +{ + const char *a = params.string(); + const char *b; + + mMap.clear(); + + for (;;) { + // Find the bounds of the key name. + b = strchr(a, '='); + if (b == 0) + break; + + // Create the key string. + String8 k(a, (size_t)(b-a)); + + // Find the value. + a = b+1; + b = strchr(a, ';'); + if (b == 0) { + // If there's no semicolon, this is the last item. + String8 v(a); + mMap.add(k, v); + break; + } + + String8 v(a, (size_t)(b-a)); + mMap.add(k, v); + a = b+1; + } +} + + +void CameraParameters2::set(const char *key, const char *value) +{ + // XXX i think i can do this with strspn() + if (strchr(key, '=') || strchr(key, ';')) { + //XXX ALOGE("Key \"%s\"contains invalid character (= or ;)", key); + return; + } + + if (strchr(value, '=') || strchr(value, ';')) { + //XXX ALOGE("Value \"%s\"contains invalid character (= or ;)", value); + return; + } + + // Replacing a value updates the key's order to be the new largest order + ssize_t res = mMap.replaceValueFor(String8(key), String8(value)); + LOG_ALWAYS_FATAL_IF(res < 0, "replaceValueFor(%s,%s) failed", key, value); +} + +void CameraParameters2::set(const char *key, int value) +{ + char str[16]; + sprintf(str, "%d", value); + set(key, str); +} + +void CameraParameters2::setFloat(const char *key, float value) +{ + char str[16]; // 14 should be enough. We overestimate to be safe. + snprintf(str, sizeof(str), "%g", value); + set(key, str); +} + +const char *CameraParameters2::get(const char *key) const +{ + ssize_t idx = mMap.indexOfKey(String8(key)); + if (idx < 0) { + return NULL; + } else { + return mMap.valueAt(idx).string(); + } +} + +int CameraParameters2::getInt(const char *key) const +{ + const char *v = get(key); + if (v == 0) + return -1; + return strtol(v, 0, 0); +} + +float CameraParameters2::getFloat(const char *key) const +{ + const char *v = get(key); + if (v == 0) return -1; + return strtof(v, 0); +} + +status_t CameraParameters2::compareSetOrder(const char *key1, const char *key2, + int *order) const { + if (key1 == NULL) { + ALOGE("%s: key1 must not be NULL", __FUNCTION__); + return BAD_VALUE; + } else if (key2 == NULL) { + ALOGE("%s: key2 must not be NULL", __FUNCTION__); + return BAD_VALUE; + } else if (order == NULL) { + ALOGE("%s: order must not be NULL", __FUNCTION__); + return BAD_VALUE; + } + + ssize_t index1 = mMap.indexOfKey(String8(key1)); + ssize_t index2 = mMap.indexOfKey(String8(key2)); + if (index1 < 0) { + ALOGW("%s: Key1 (%s) was not set", __FUNCTION__, key1); + return NAME_NOT_FOUND; + } else if (index2 < 0) { + ALOGW("%s: Key2 (%s) was not set", __FUNCTION__, key2); + return NAME_NOT_FOUND; + } + + *order = (index1 == index2) ? 0 : + (index1 < index2) ? -1 : + 1; + + return OK; +} + +void CameraParameters2::remove(const char *key) +{ + mMap.removeItem(String8(key)); +} + +// Parse string like "640x480" or "10000,20000" +static int parse_pair(const char *str, int *first, int *second, char delim, + char **endptr = NULL) +{ + // Find the first integer. + char *end; + int w = (int)strtol(str, &end, 10); + // If a delimeter does not immediately follow, give up. + if (*end != delim) { + ALOGE("Cannot find delimeter (%c) in str=%s", delim, str); + return -1; + } + + // Find the second integer, immediately after the delimeter. + int h = (int)strtol(end+1, &end, 10); + + *first = w; + *second = h; + + if (endptr) { + *endptr = end; + } + + return 0; +} + +static void parseSizesList(const char *sizesStr, Vector<Size> &sizes) +{ + if (sizesStr == 0) { + return; + } + + char *sizeStartPtr = (char *)sizesStr; + + while (true) { + int width, height; + int success = parse_pair(sizeStartPtr, &width, &height, 'x', + &sizeStartPtr); + if (success == -1 || (*sizeStartPtr != ',' && *sizeStartPtr != '\0')) { + ALOGE("Picture sizes string \"%s\" contains invalid character.", sizesStr); + return; + } + sizes.push(Size(width, height)); + + if (*sizeStartPtr == '\0') { + return; + } + sizeStartPtr++; + } +} + +void CameraParameters2::setPreviewSize(int width, int height) +{ + char str[32]; + sprintf(str, "%dx%d", width, height); + set(CameraParameters::KEY_PREVIEW_SIZE, str); +} + +void CameraParameters2::getPreviewSize(int *width, int *height) const +{ + *width = *height = -1; + // Get the current string, if it doesn't exist, leave the -1x-1 + const char *p = get(CameraParameters::KEY_PREVIEW_SIZE); + if (p == 0) return; + parse_pair(p, width, height, 'x'); +} + +void CameraParameters2::getPreferredPreviewSizeForVideo(int *width, int *height) const +{ + *width = *height = -1; + const char *p = get(CameraParameters::KEY_PREFERRED_PREVIEW_SIZE_FOR_VIDEO); + if (p == 0) return; + parse_pair(p, width, height, 'x'); +} + +void CameraParameters2::getSupportedPreviewSizes(Vector<Size> &sizes) const +{ + const char *previewSizesStr = get(CameraParameters::KEY_SUPPORTED_PREVIEW_SIZES); + parseSizesList(previewSizesStr, sizes); +} + +void CameraParameters2::setVideoSize(int width, int height) +{ + char str[32]; + sprintf(str, "%dx%d", width, height); + set(CameraParameters::KEY_VIDEO_SIZE, str); +} + +void CameraParameters2::getVideoSize(int *width, int *height) const +{ + *width = *height = -1; + const char *p = get(CameraParameters::KEY_VIDEO_SIZE); + if (p == 0) return; + parse_pair(p, width, height, 'x'); +} + +void CameraParameters2::getSupportedVideoSizes(Vector<Size> &sizes) const +{ + const char *videoSizesStr = get(CameraParameters::KEY_SUPPORTED_VIDEO_SIZES); + parseSizesList(videoSizesStr, sizes); +} + +void CameraParameters2::setPreviewFrameRate(int fps) +{ + set(CameraParameters::KEY_PREVIEW_FRAME_RATE, fps); +} + +int CameraParameters2::getPreviewFrameRate() const +{ + return getInt(CameraParameters::KEY_PREVIEW_FRAME_RATE); +} + +void CameraParameters2::getPreviewFpsRange(int *min_fps, int *max_fps) const +{ + *min_fps = *max_fps = -1; + const char *p = get(CameraParameters::KEY_PREVIEW_FPS_RANGE); + if (p == 0) return; + parse_pair(p, min_fps, max_fps, ','); +} + +void CameraParameters2::setPreviewFpsRange(int min_fps, int max_fps) +{ + String8 str = String8::format("%d,%d", min_fps, max_fps); + set(CameraParameters::KEY_PREVIEW_FPS_RANGE, str.string()); +} + +void CameraParameters2::setPreviewFormat(const char *format) +{ + set(CameraParameters::KEY_PREVIEW_FORMAT, format); +} + +const char *CameraParameters2::getPreviewFormat() const +{ + return get(CameraParameters::KEY_PREVIEW_FORMAT); +} + +void CameraParameters2::setPictureSize(int width, int height) +{ + char str[32]; + sprintf(str, "%dx%d", width, height); + set(CameraParameters::KEY_PICTURE_SIZE, str); +} + +void CameraParameters2::getPictureSize(int *width, int *height) const +{ + *width = *height = -1; + // Get the current string, if it doesn't exist, leave the -1x-1 + const char *p = get(CameraParameters::KEY_PICTURE_SIZE); + if (p == 0) return; + parse_pair(p, width, height, 'x'); +} + +void CameraParameters2::getSupportedPictureSizes(Vector<Size> &sizes) const +{ + const char *pictureSizesStr = get(CameraParameters::KEY_SUPPORTED_PICTURE_SIZES); + parseSizesList(pictureSizesStr, sizes); +} + +void CameraParameters2::setPictureFormat(const char *format) +{ + set(CameraParameters::KEY_PICTURE_FORMAT, format); +} + +const char *CameraParameters2::getPictureFormat() const +{ + return get(CameraParameters::KEY_PICTURE_FORMAT); +} + +void CameraParameters2::dump() const +{ + ALOGD("dump: mMap.size = %d", mMap.size()); + for (size_t i = 0; i < mMap.size(); i++) { + String8 k, v; + k = mMap.keyAt(i); + v = mMap.valueAt(i); + ALOGD("%s: %s\n", k.string(), v.string()); + } +} + +status_t CameraParameters2::dump(int fd, const Vector<String16>& args) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + snprintf(buffer, 255, "CameraParameters2::dump: mMap.size = %zu\n", mMap.size()); + result.append(buffer); + for (size_t i = 0; i < mMap.size(); i++) { + String8 k, v; + k = mMap.keyAt(i); + v = mMap.valueAt(i); + snprintf(buffer, 255, "\t%s: %s\n", k.string(), v.string()); + result.append(buffer); + } + write(fd, result.string(), result.size()); + return NO_ERROR; +} + +}; // namespace android diff --git a/include/camera/CameraParameters2.h b/include/camera/CameraParameters2.h new file mode 100644 index 0000000..88ad812 --- /dev/null +++ b/include/camera/CameraParameters2.h @@ -0,0 +1,203 @@ +/* + * Copyright (C) 2014 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_HARDWARE_CAMERA_PARAMETERS2_H +#define ANDROID_HARDWARE_CAMERA_PARAMETERS2_H + +#include <utils/Vector.h> +#include <utils/String8.h> +#include "CameraParameters.h" + +namespace android { + +/** + * A copy of CameraParameters plus ABI-breaking changes. Needed + * because some camera HALs directly link to CameraParameters and cannot + * tolerate an ABI change. + */ +class CameraParameters2 +{ +public: + CameraParameters2(); + CameraParameters2(const String8 ¶ms) { unflatten(params); } + ~CameraParameters2(); + + String8 flatten() const; + void unflatten(const String8 ¶ms); + + void set(const char *key, const char *value); + void set(const char *key, int value); + void setFloat(const char *key, float value); + // Look up string value by key. + // -- The string remains valid until the next set/remove of the same key, + // or until the map gets cleared. + const char *get(const char *key) const; + int getInt(const char *key) const; + float getFloat(const char *key) const; + + // Compare the order that key1 was set vs the order that key2 was set. + // + // Sets the order parameter to an integer less than, equal to, or greater + // than zero if key1's set order was respectively, to be less than, to + // match, or to be greater than key2's set order. + // + // Error codes: + // * NAME_NOT_FOUND - if either key has not been set previously + // * BAD_VALUE - if any of the parameters are NULL + status_t compareSetOrder(const char *key1, const char *key2, + /*out*/ + int *order) const; + + void remove(const char *key); + + void setPreviewSize(int width, int height); + void getPreviewSize(int *width, int *height) const; + void getSupportedPreviewSizes(Vector<Size> &sizes) const; + + // Set the dimensions in pixels to the given width and height + // for video frames. The given width and height must be one + // of the supported dimensions returned from + // getSupportedVideoSizes(). Must not be called if + // getSupportedVideoSizes() returns an empty Vector of Size. + void setVideoSize(int width, int height); + // Retrieve the current dimensions (width and height) + // in pixels for video frames, which must be one of the + // supported dimensions returned from getSupportedVideoSizes(). + // Must not be called if getSupportedVideoSizes() returns an + // empty Vector of Size. + void getVideoSize(int *width, int *height) const; + // Retrieve a Vector of supported dimensions (width and height) + // in pixels for video frames. If sizes returned from the method + // is empty, the camera does not support calls to setVideoSize() + // or getVideoSize(). In adddition, it also indicates that + // the camera only has a single output, and does not have + // separate output for video frames and preview frame. + void getSupportedVideoSizes(Vector<Size> &sizes) const; + // Retrieve the preferred preview size (width and height) in pixels + // for video recording. The given width and height must be one of + // supported preview sizes returned from getSupportedPreviewSizes(). + // Must not be called if getSupportedVideoSizes() returns an empty + // Vector of Size. If getSupportedVideoSizes() returns an empty + // Vector of Size, the width and height returned from this method + // is invalid, and is "-1x-1". + void getPreferredPreviewSizeForVideo(int *width, int *height) const; + + void setPreviewFrameRate(int fps); + int getPreviewFrameRate() const; + void getPreviewFpsRange(int *min_fps, int *max_fps) const; + void setPreviewFpsRange(int min_fps, int max_fps); + void setPreviewFormat(const char *format); + const char *getPreviewFormat() const; + void setPictureSize(int width, int height); + void getPictureSize(int *width, int *height) const; + void getSupportedPictureSizes(Vector<Size> &sizes) const; + void setPictureFormat(const char *format); + const char *getPictureFormat() const; + + void dump() const; + status_t dump(int fd, const Vector<String16>& args) const; + +private: + + // Quick and dirty map that maintains insertion order + template <typename KeyT, typename ValueT> + struct OrderedKeyedVector { + + ssize_t add(const KeyT& key, const ValueT& value) { + return mList.add(Pair(key, value)); + } + + size_t size() const { + return mList.size(); + } + + const KeyT& keyAt(size_t idx) const { + return mList[idx].mKey; + } + + const ValueT& valueAt(size_t idx) const { + return mList[idx].mValue; + } + + const ValueT& valueFor(const KeyT& key) const { + ssize_t i = indexOfKey(key); + LOG_ALWAYS_FATAL_IF(i<0, "%s: key not found", __PRETTY_FUNCTION__); + + return valueAt(i); + } + + ssize_t indexOfKey(const KeyT& key) const { + size_t vectorIdx = 0; + for (; vectorIdx < mList.size(); ++vectorIdx) { + if (mList[vectorIdx].mKey == key) { + return (ssize_t) vectorIdx; + } + } + + return NAME_NOT_FOUND; + } + + ssize_t removeItem(const KeyT& key) { + size_t vectorIdx = (size_t) indexOfKey(key); + + if (vectorIdx < 0) { + return vectorIdx; + } + + return mList.removeAt(vectorIdx); + } + + void clear() { + mList.clear(); + } + + // Same as removing and re-adding. The key's index changes to max. + ssize_t replaceValueFor(const KeyT& key, const ValueT& value) { + removeItem(key); + return add(key, value); + } + + private: + + struct Pair { + Pair() : mKey(), mValue() {} + Pair(const KeyT& key, const ValueT& value) : + mKey(key), + mValue(value) {} + KeyT mKey; + ValueT mValue; + }; + + Vector<Pair> mList; + }; + + /** + * Order matters: Keys that are set() later are stored later in the map. + * + * If two keys have meaning that conflict, then the later-set key + * wins. + * + * For example, preview FPS and preview FPS range conflict since only + * we only want to use the FPS range if that's the last thing that was set. + * So in that case, only use preview FPS range if it was set later than + * the preview FPS. + */ + OrderedKeyedVector<String8,String8> mMap; +}; + +}; // namespace android + +#endif diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index f6646ab..4736369 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -740,6 +740,7 @@ protected: bool mInUnderrun; // whether track is currently in underrun state String8 mName; // server's name for this IAudioTrack + uint32_t mPausedPosition; private: class DeathNotifier : public IBinder::DeathRecipient { diff --git a/include/media/IOMX.h b/include/media/IOMX.h index 9c8451c..6643736 100644 --- a/include/media/IOMX.h +++ b/include/media/IOMX.h @@ -142,6 +142,7 @@ public: enum InternalOptionType { INTERNAL_OPTION_SUSPEND, // data is a bool INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t + INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t }; virtual status_t setInternalOption( node_id node, diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h index e796ab3..46c62dc 100644 --- a/include/media/stagefright/ACodec.h +++ b/include/media/stagefright/ACodec.h @@ -67,6 +67,8 @@ struct ACodec : public AHierarchicalStateMachine { void signalRequestIDRFrame(); + bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; } + struct PortDescription : public RefBase { size_t countBuffers(); IOMX::buffer_id bufferIDAt(size_t index) const; @@ -116,6 +118,7 @@ private: kWhatStart = 'star', kWhatRequestIDRFrame = 'ridr', kWhatSetParameters = 'setP', + kWhatSubmitOutputMetaDataBufferIfEOS = 'subm', }; enum { @@ -186,6 +189,7 @@ private: bool mIsEncoder; bool mUseMetadataOnEncoderOutput; bool mShutdownInProgress; + bool mIsConfiguredForAdaptivePlayback; // If "mKeepComponentAllocated" we only transition back to Loaded state // and do not release the component instance. @@ -199,8 +203,10 @@ private: unsigned mDequeueCounter; bool mStoreMetaDataInOutputBuffers; int32_t mMetaDataBuffersToSubmit; + size_t mNumUndequeuedBuffers; int64_t mRepeatFrameDelayUs; + int64_t mMaxPtsGapUs; status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode); status_t allocateBuffersOnPort(OMX_U32 portIndex); @@ -212,6 +218,7 @@ private: OMX_U32 *nMinUndequeuedBuffers); status_t allocateOutputMetaDataBuffers(); status_t submitOutputMetaDataBuffer(); + void signalSubmitOutputMetaDataBufferIfEOS_workaround(); status_t allocateOutputBuffersFromNativeWindow(); status_t cancelBufferToNativeWindow(BufferInfo *info); status_t freeOutputBuffersNotOwnedByComponent(); diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 666fafa..ccbc5a3 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -545,13 +545,13 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) } const struct timespec *requested; + struct timespec timeout; if (waitCount == -1) { requested = &ClientProxy::kForever; } else if (waitCount == 0) { requested = &ClientProxy::kNonBlocking; } else if (waitCount > 0) { long long ms = WAIT_PERIOD_MS * (long long) waitCount; - struct timespec timeout; timeout.tv_sec = ms / 1000; timeout.tv_nsec = (int) (ms % 1000) * 1000000; requested = &timeout; diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index a9d6993..3f3a88c 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -85,7 +85,8 @@ AudioTrack::AudioTrack() : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { } @@ -106,7 +107,8 @@ AudioTrack::AudioTrack( : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, @@ -131,7 +133,8 @@ AudioTrack::AudioTrack( : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT) + mPreviousSchedulingGroup(SP_DEFAULT), + mPausedPosition(0) { mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, @@ -529,6 +532,16 @@ void AudioTrack::pause() } mProxy->interrupt(); mAudioTrack->pause(); + + if (isOffloaded()) { + if (mOutput != 0) { + uint32_t halFrames; + // OffloadThread sends HAL pause in its threadLoop.. time saved + // here can be slightly off + AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); + ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); + } + } } status_t AudioTrack::setVolume(float left, float right) @@ -747,6 +760,12 @@ status_t AudioTrack::getPosition(uint32_t *position) const if (isOffloaded()) { uint32_t dspFrames = 0; + if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) { + ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); + *position = mPausedPosition; + return NO_ERROR; + } + if (mOutput != 0) { uint32_t halFrames; AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); @@ -1113,13 +1132,13 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) } const struct timespec *requested; + struct timespec timeout; if (waitCount == -1) { requested = &ClientProxy::kForever; } else if (waitCount == 0) { requested = &ClientProxy::kNonBlocking; } else if (waitCount > 0) { long long ms = WAIT_PERIOD_MS * (long long) waitCount; - struct timespec timeout; timeout.tv_sec = ms / 1000; timeout.tv_nsec = (int) (ms % 1000) * 1000000; requested = &timeout; @@ -1451,6 +1470,7 @@ nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) } size_t misalignment = mProxy->getMisalignment(); uint32_t sequence = mSequence; + sp<AudioTrackClientProxy> proxy = mProxy; // These fields don't need to be cached, because they are assigned only by set(): // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags @@ -1459,35 +1479,32 @@ nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) mLock.unlock(); if (waitStreamEnd) { - AutoMutex lock(mLock); - - sp<AudioTrackClientProxy> proxy = mProxy; - sp<IMemory> iMem = mCblkMemory; - struct timespec timeout; timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; timeout.tv_nsec = 0; - mLock.unlock(); - status_t status = mProxy->waitStreamEndDone(&timeout); - mLock.lock(); + status_t status = proxy->waitStreamEndDone(&timeout); switch (status) { case NO_ERROR: case DEAD_OBJECT: case TIMED_OUT: - mLock.unlock(); mCbf(EVENT_STREAM_END, mUserData, NULL); - mLock.lock(); - if (mState == STATE_STOPPING) { - mState = STATE_STOPPED; - if (status != DEAD_OBJECT) { - return NS_INACTIVE; + { + AutoMutex lock(mLock); + // The previously assigned value of waitStreamEnd is no longer valid, + // since the mutex has been unlocked and either the callback handler + // or another thread could have re-started the AudioTrack during that time. + waitStreamEnd = mState == STATE_STOPPING; + if (waitStreamEnd) { + mState = STATE_STOPPED; } } - return 0; - default: - return 0; + if (waitStreamEnd && status != DEAD_OBJECT) { + return NS_INACTIVE; + } + break; } + return 0; } // perform callbacks while unlocked diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp index 3669a5b..25d55a3 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp @@ -1011,7 +1011,14 @@ status_t NuPlayer::feedDecoderInputData(bool audio, const sp<AMessage> &msg) { &NuPlayer::performScanSources)); } - flushDecoder(audio, formatChange); + sp<AMessage> newFormat = mSource->getFormat(audio); + sp<Decoder> &decoder = audio ? mAudioDecoder : mVideoDecoder; + if (formatChange && !decoder->supportsSeamlessFormatChange(newFormat)) { + flushDecoder(audio, /* needShutdown = */ true); + } else { + flushDecoder(audio, /* needShutdown = */ false); + err = OK; + } } else { // This stream is unaffected by the discontinuity diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp index 22f699e..2423fd5 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp @@ -67,6 +67,7 @@ void NuPlayer::Decoder::configure(const sp<AMessage> &format) { // queue. bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6); + mFormat = format; mCodec = new ACodec; if (needDedicatedLooper && mCodecLooper == NULL) { @@ -147,5 +148,65 @@ void NuPlayer::Decoder::initiateShutdown() { } } +bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const { + if (targetFormat == NULL) { + return true; + } + + AString mime; + if (!targetFormat->findString("mime", &mime)) { + return false; + } + + if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) { + // field-by-field comparison + const char * keys[] = { "channel-count", "sample-rate", "is-adts" }; + for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) { + int32_t oldVal, newVal; + if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal) + || oldVal != newVal) { + return false; + } + } + + sp<ABuffer> oldBuf, newBuf; + if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) { + if (oldBuf->size() != newBuf->size()) { + return false; + } + return !memcmp(oldBuf->data(), newBuf->data(), oldBuf->size()); + } + } + return false; +} + +bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const { + if (mFormat == NULL) { + return false; + } + + if (targetFormat == NULL) { + return true; + } + + AString oldMime, newMime; + if (!mFormat->findString("mime", &oldMime) + || !targetFormat->findString("mime", &newMime) + || !(oldMime == newMime)) { + return false; + } + + bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/")); + bool seamless; + if (audio) { + seamless = supportsSeamlessAudioFormatChange(targetFormat); + } else { + seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback(); + } + + ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str()); + return seamless; +} + } // namespace android diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h index a876148..78ea74a 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h @@ -36,6 +36,8 @@ struct NuPlayer::Decoder : public AHandler { void signalResume(); void initiateShutdown(); + bool supportsSeamlessFormatChange(const sp<AMessage> &to) const; + protected: virtual ~Decoder(); @@ -49,6 +51,7 @@ private: sp<AMessage> mNotify; sp<NativeWindowWrapper> mNativeWindow; + sp<AMessage> mFormat; sp<ACodec> mCodec; sp<ALooper> mCodecLooper; @@ -59,6 +62,8 @@ private: void onFillThisBuffer(const sp<AMessage> &msg); + bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const; + DISALLOW_EVIL_CONSTRUCTORS(Decoder); }; diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index dfe6c56..2d04bbb 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -363,6 +363,7 @@ ACodec::ACodec() mIsEncoder(false), mUseMetadataOnEncoderOutput(false), mShutdownInProgress(false), + mIsConfiguredForAdaptivePlayback(false), mEncoderDelay(0), mEncoderPadding(0), mChannelMaskPresent(false), @@ -370,7 +371,8 @@ ACodec::ACodec() mDequeueCounter(0), mStoreMetaDataInOutputBuffers(false), mMetaDataBuffersToSubmit(0), - mRepeatFrameDelayUs(-1ll) { + mRepeatFrameDelayUs(-1ll), + mMaxPtsGapUs(-1l) { mUninitializedState = new UninitializedState(this); mLoadedState = new LoadedState(this); mLoadedToIdleState = new LoadedToIdleState(this); @@ -452,6 +454,18 @@ void ACodec::signalRequestIDRFrame() { (new AMessage(kWhatRequestIDRFrame, id()))->post(); } +// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED *** +// Some codecs may return input buffers before having them processed. +// This causes a halt if we already signaled an EOS on the input +// port. For now keep submitting an output buffer if there was an +// EOS on the input port, but not yet on the output port. +void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() { + if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] && + mMetaDataBuffersToSubmit > 0) { + (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post(); + } +} + status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) { CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput); @@ -624,18 +638,34 @@ status_t ACodec::configureOutputBuffersFromNativeWindow( return err; } - // XXX: Is this the right logic to use? It's not clear to me what the OMX - // buffer counts refer to - how do they account for the renderer holding on - // to buffers? - if (def.nBufferCountActual < def.nBufferCountMin + *minUndequeuedBuffers) { - OMX_U32 newBufferCount = def.nBufferCountMin + *minUndequeuedBuffers; + // FIXME: assume that surface is controlled by app (native window + // returns the number for the case when surface is not controlled by app) + // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported + // For now, try to allocate 1 more buffer, but don't fail if unsuccessful + + // Use conservative allocation while also trying to reduce starvation + // + // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the + // minimum needed for the consumer to be able to work + // 2. try to allocate two (2) additional buffers to reduce starvation from + // the consumer + // plus an extra buffer to account for incorrect minUndequeuedBufs + for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) { + OMX_U32 newBufferCount = + def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - if (err != OK) { - ALOGE("[%s] setting nBufferCountActual to %lu failed: %d", - mComponentName.c_str(), newBufferCount, err); + if (err == OK) { + *minUndequeuedBuffers += extraBuffers; + break; + } + + ALOGW("[%s] setting nBufferCountActual to %lu failed: %d", + mComponentName.c_str(), newBufferCount, err); + /* exit condition */ + if (extraBuffers == 0) { return err; } } @@ -660,6 +690,7 @@ status_t ACodec::allocateOutputBuffersFromNativeWindow() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; + mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu buffers from a native window of size %lu on " "output port", @@ -725,6 +756,7 @@ status_t ACodec::allocateOutputMetaDataBuffers() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; + mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu meta buffers on output port", mComponentName.c_str(), bufferCount); @@ -1096,6 +1128,10 @@ status_t ACodec::configureCodec( &mRepeatFrameDelayUs)) { mRepeatFrameDelayUs = -1ll; } + + if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) { + mMaxPtsGapUs = -1l; + } } // Always try to enable dynamic output buffers on native surface @@ -1103,6 +1139,7 @@ status_t ACodec::configureCodec( int32_t haveNativeWindow = msg->findObject("native-window", &obj) && obj != NULL; mStoreMetaDataInOutputBuffers = false; + mIsConfiguredForAdaptivePlayback = false; if (!encoder && video && haveNativeWindow) { err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE); if (err != OK) { @@ -1147,12 +1184,14 @@ status_t ACodec::configureCodec( ALOGW_IF(err != OK, "[%s] prepareForAdaptivePlayback failed w/ err %d", mComponentName.c_str(), err); + mIsConfiguredForAdaptivePlayback = (err == OK); } // allow failure err = OK; } else { ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str()); mStoreMetaDataInOutputBuffers = true; + mIsConfiguredForAdaptivePlayback = true; } int32_t push; @@ -2408,19 +2447,7 @@ void ACodec::waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs() { return; } - int minUndequeuedBufs = 0; - status_t err = mNativeWindow->query( - mNativeWindow.get(), NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, - &minUndequeuedBufs); - - if (err != OK) { - ALOGE("[%s] NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS query failed: %s (%d)", - mComponentName.c_str(), strerror(-err), -err); - - minUndequeuedBufs = 0; - } - - while (countBuffersOwnedByNativeWindow() > (size_t)minUndequeuedBufs + while (countBuffersOwnedByNativeWindow() > mNumUndequeuedBuffers && dequeueBufferFromNativeWindow() != NULL) { // these buffers will be submitted as regular buffers; account for this if (mStoreMetaDataInOutputBuffers && mMetaDataBuffersToSubmit > 0) { @@ -3235,11 +3262,11 @@ void ACodec::BaseState::onInputBufferFilled(const sp<AMessage> &msg) { mCodec->mInputEOSResult = err; } break; - - default: - CHECK_EQ((int)mode, (int)FREE_BUFFERS); - break; } + + default: + CHECK_EQ((int)mode, (int)FREE_BUFFERS); + break; } } @@ -3690,6 +3717,7 @@ void ACodec::LoadedState::stateEntered() { mCodec->mDequeueCounter = 0; mCodec->mMetaDataBuffersToSubmit = 0; mCodec->mRepeatFrameDelayUs = -1ll; + mCodec->mIsConfiguredForAdaptivePlayback = false; if (mCodec->mShutdownInProgress) { bool keepComponentAllocated = mCodec->mKeepComponentAllocated; @@ -3838,6 +3866,21 @@ void ACodec::LoadedState::onCreateInputSurface( } } + if (err == OK && mCodec->mMaxPtsGapUs > 0l) { + err = mCodec->mOMX->setInternalOption( + mCodec->mNode, + kPortIndexInput, + IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP, + &mCodec->mMaxPtsGapUs, + sizeof(mCodec->mMaxPtsGapUs)); + + if (err != OK) { + ALOGE("[%s] Unable to configure max timestamp gap (err %d)", + mCodec->mComponentName.c_str(), + err); + } + } + if (err == OK) { notify->setObject("input-surface", new BufferProducerWrapper(bufferProducer)); @@ -4036,6 +4079,9 @@ void ACodec::ExecutingState::submitOutputMetaBuffers() { break; } } + + // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED *** + mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround(); } void ACodec::ExecutingState::submitRegularOutputBuffers() { @@ -4184,6 +4230,19 @@ bool ACodec::ExecutingState::onMessageReceived(const sp<AMessage> &msg) { break; } + // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED *** + case kWhatSubmitOutputMetaDataBufferIfEOS: + { + if (mCodec->mPortEOS[kPortIndexInput] && + !mCodec->mPortEOS[kPortIndexOutput]) { + status_t err = mCodec->submitOutputMetaDataBuffer(); + if (err == OK) { + mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround(); + } + } + return true; + } + default: handled = BaseState::onMessageReceived(msg); break; diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp index 900b160..58a4487 100644 --- a/media/libstagefright/MPEG4Writer.cpp +++ b/media/libstagefright/MPEG4Writer.cpp @@ -44,7 +44,9 @@ namespace android { static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024; -static const int64_t kMax32BitFileSize = 0x007fffffffLL; +static const int64_t kMax32BitFileSize = 0x00ffffffffLL; // 2^32-1 : max FAT32 + // filesystem file size + // used by most SD cards static const uint8_t kNalUnitTypeSeqParamSet = 0x07; static const uint8_t kNalUnitTypePicParamSet = 0x08; static const int64_t kInitialDelayTimeUs = 700000LL; @@ -861,11 +863,11 @@ status_t MPEG4Writer::reset() { // Fix up the size of the 'mdat' chunk. if (mUse32BitOffset) { lseek64(mFd, mMdatOffset, SEEK_SET); - int32_t size = htonl(static_cast<int32_t>(mOffset - mMdatOffset)); + uint32_t size = htonl(static_cast<uint32_t>(mOffset - mMdatOffset)); ::write(mFd, &size, 4); } else { lseek64(mFd, mMdatOffset + 8, SEEK_SET); - int64_t size = mOffset - mMdatOffset; + uint64_t size = mOffset - mMdatOffset; size = hton64(size); ::write(mFd, &size, 8); } diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp index a711e43..450fb3b 100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -94,6 +94,7 @@ static sp<MediaSource> InstantiateSoftwareEncoder( #define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__) +#define CODEC_LOGW(x, ...) ALOGW("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGE(x, ...) ALOGE("[%s] "x, mComponentName, ##__VA_ARGS__) struct OMXCodecObserver : public BnOMXObserver { @@ -1779,21 +1780,42 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() { strerror(-err), -err); return err; } - - // XXX: Is this the right logic to use? It's not clear to me what the OMX - // buffer counts refer to - how do they account for the renderer holding on - // to buffers? - if (def.nBufferCountActual < def.nBufferCountMin + minUndequeuedBufs) { - OMX_U32 newBufferCount = def.nBufferCountMin + minUndequeuedBufs; + // FIXME: assume that surface is controlled by app (native window + // returns the number for the case when surface is not controlled by app) + // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported + // For now, try to allocate 1 more buffer, but don't fail if unsuccessful + + // Use conservative allocation while also trying to reduce starvation + // + // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the + // minimum needed for the consumer to be able to work + // 2. try to allocate two (2) additional buffers to reduce starvation from + // the consumer + // plus an extra buffer to account for incorrect minUndequeuedBufs + CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1", + def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); + + for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) { + OMX_U32 newBufferCount = + def.nBufferCountMin + minUndequeuedBufs + extraBuffers; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - if (err != OK) { - CODEC_LOGE("setting nBufferCountActual to %lu failed: %d", - newBufferCount, err); + + if (err == OK) { + minUndequeuedBufs += extraBuffers; + break; + } + + CODEC_LOGW("setting nBufferCountActual to %lu failed: %d", + newBufferCount, err); + /* exit condition */ + if (extraBuffers == 0) { return err; } } + CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1", + def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); err = native_window_set_buffer_count( mNativeWindow.get(), def.nBufferCountActual); diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk index f26f386..109e3fe 100644 --- a/media/libstagefright/chromium_http/Android.mk +++ b/media/libstagefright/chromium_http/Android.mk @@ -18,9 +18,11 @@ LOCAL_C_INCLUDES:= \ LOCAL_CFLAGS += -Wno-multichar LOCAL_SHARED_LIBRARIES += \ + libbinder \ libstlport \ libchromium_net \ libutils \ + libbinder \ libcutils \ liblog \ libstagefright_foundation \ diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp index 3b33212..3de4877 100644 --- a/media/libstagefright/chromium_http/support.cpp +++ b/media/libstagefright/chromium_http/support.cpp @@ -34,14 +34,97 @@ #include "net/proxy/proxy_config_service_android.h" #include "include/ChromiumHTTPDataSource.h" - +#include <arpa/inet.h> +#include <binder/Parcel.h> #include <cutils/log.h> #include <media/stagefright/MediaErrors.h> #include <media/stagefright/Utils.h> #include <string> +#include <utils/Errors.h> +#include <binder/IInterface.h> +#include <binder/IServiceManager.h> + namespace android { +// must be kept in sync with interface defined in IAudioService.aidl +class IAudioService : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioService); + + virtual int verifyX509CertChain( + const std::vector<std::string>& cert_chain, + const std::string& hostname, + const std::string& auth_type) = 0; +}; + +class BpAudioService : public BpInterface<IAudioService> +{ +public: + BpAudioService(const sp<IBinder>& impl) + : BpInterface<IAudioService>(impl) + { + } + + virtual int verifyX509CertChain( + const std::vector<std::string>& cert_chain, + const std::string& hostname, + const std::string& auth_type) + { + Parcel data, reply; + data.writeInterfaceToken(IAudioService::getInterfaceDescriptor()); + + // The vector of std::string we get isn't really a vector of strings, + // but rather a vector of binary certificate data. If we try to pass + // it to Java language code as a string, it ends up mangled on the other + // side, so send them as bytes instead. + // Since we can't send an array of byte arrays, send a single array, + // which will be split out by the recipient. + + int numcerts = cert_chain.size(); + data.writeInt32(numcerts); + size_t total = 0; + for (int i = 0; i < numcerts; i++) { + total += cert_chain[i].size(); + } + size_t bytesize = total + numcerts * 4; + uint8_t *bytes = (uint8_t*) malloc(bytesize); + if (!bytes) { + return 5; // SSL_INVALID + } + ALOGV("%d certs: %d -> %d", numcerts, total, bytesize); + + int offset = 0; + for (int i = 0; i < numcerts; i++) { + int32_t certsize = cert_chain[i].size(); + // store this in a known order, which just happens to match the default + // byte order of a java ByteBuffer + int32_t bigsize = htonl(certsize); + ALOGV("cert %d, size %d", i, certsize); + memcpy(bytes + offset, &bigsize, sizeof(bigsize)); + offset += sizeof(bigsize); + memcpy(bytes + offset, cert_chain[i].data(), certsize); + offset += certsize; + } + data.writeByteArray(bytesize, bytes); + free(bytes); + data.writeString16(String16(hostname.c_str())); + data.writeString16(String16(auth_type.c_str())); + + int32_t result; + if (remote()->transact(IBinder::FIRST_CALL_TRANSACTION, data, &reply) != NO_ERROR + || reply.readExceptionCode() < 0 || reply.readInt32(&result) != NO_ERROR) { + return 5; // SSL_INVALID; + } + return result; + } + +}; + +IMPLEMENT_META_INTERFACE(AudioService, "android.media.IAudioService"); + + static Mutex gNetworkThreadLock; static base::Thread *gNetworkThread = NULL; static scoped_refptr<SfRequestContext> gReqContext; @@ -226,7 +309,24 @@ SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain( const std::vector<std::string>& cert_chain, const std::string& hostname, const std::string& auth_type) { - return VERIFY_OK; + + sp<IBinder> binder = + defaultServiceManager()->checkService(String16("audio")); + if (binder == 0) { + ALOGW("Thread cannot connect to the audio service"); + } else { + sp<IAudioService> service = interface_cast<IAudioService>(binder); + int code = service->verifyX509CertChain(cert_chain, hostname, auth_type); + ALOGV("verified: %d", code); + if (code == -1) { + return VERIFY_OK; + } else if (code == 2) { // SSL_IDMISMATCH + return VERIFY_BAD_HOSTNAME; + } else if (code == 3) { // SSL_UNTRUSTED + return VERIFY_NO_TRUSTED_ROOT; + } + } + return VERIFY_INVOCATION_ERROR; } //////////////////////////////////////////////////////////////////////////////// diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 033d981..6d48ab7 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -37,6 +37,8 @@ #include <media/stagefright/MetaData.h> #include <media/stagefright/Utils.h> +#include <utils/Mutex.h> + #include <ctype.h> #include <inttypes.h> #include <openssl/aes.h> @@ -58,32 +60,57 @@ LiveSession::LiveSession( : 0)), mPrevBandwidthIndex(-1), mStreamMask(0), + mNewStreamMask(0), + mSwapMask(0), mCheckBandwidthGeneration(0), + mSwitchGeneration(0), mLastDequeuedTimeUs(0ll), mRealTimeBaseUs(0ll), mReconfigurationInProgress(false), - mDisconnectReplyID(0) { + mSwitchInProgress(false), + mDisconnectReplyID(0), + mSeekReplyID(0) { if (mUIDValid) { mHTTPDataSource->setUID(mUID); } - mPacketSources.add( - STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */)); - - mPacketSources.add( - STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */)); + mStreams[kAudioIndex] = StreamItem("audio"); + mStreams[kVideoIndex] = StreamItem("video"); + mStreams[kSubtitleIndex] = StreamItem("subtitles"); - mPacketSources.add( - STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */)); + for (size_t i = 0; i < kMaxStreams; ++i) { + mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); + mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); + } } LiveSession::~LiveSession() { } +sp<ABuffer> LiveSession::createFormatChangeBuffer(bool swap) { + ABuffer *discontinuity = new ABuffer(0); + discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE); + discontinuity->meta()->setInt32("swapPacketSource", swap); + discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration); + discontinuity->meta()->setInt64("timeUs", -1); + return discontinuity; +} + +void LiveSession::swapPacketSource(StreamType stream) { + sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream); + sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream); + sp<AnotherPacketSource> tmp = aps; + aps = aps2; + aps2 = tmp; + aps2->clear(); +} + status_t LiveSession::dequeueAccessUnit( StreamType stream, sp<ABuffer> *accessUnit) { if (!(mStreamMask & stream)) { - return UNKNOWN_ERROR; + // return -EWOULDBLOCK to avoid halting the decoder + // when switching between audio/video and audio only. + return -EWOULDBLOCK; } sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream); @@ -123,6 +150,25 @@ status_t LiveSession::dequeueAccessUnit( streamStr, type, extra == NULL ? "NULL" : extra->debugString().c_str()); + + int32_t swap; + if (type == ATSParser::DISCONTINUITY_FORMATCHANGE + && (*accessUnit)->meta()->findInt32("swapPacketSource", &swap) + && swap) { + + int32_t switchGeneration; + CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration)); + { + Mutex::Autolock lock(mSwapMutex); + if (switchGeneration == mSwitchGeneration) { + swapPacketSource(stream); + sp<AMessage> msg = new AMessage(kWhatSwapped, id()); + msg->setInt32("stream", stream); + msg->setInt32("switchGeneration", switchGeneration); + msg->post(); + } + } + } } else if (err == OK) { if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) { int64_t timeUs; @@ -144,6 +190,7 @@ status_t LiveSession::dequeueAccessUnit( } status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) { + // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit. if (!(mStreamMask & stream)) { return UNKNOWN_ERROR; } @@ -189,6 +236,10 @@ status_t LiveSession::seekTo(int64_t timeUs) { sp<AMessage> response; status_t err = msg->postAndAwaitResponse(&response); + uint32_t replyID; + CHECK(response == mSeekReply && 0 != mSeekReplyID); + mSeekReply.clear(); + mSeekReplyID = 0; return err; } @@ -214,15 +265,12 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { case kWhatSeek: { - uint32_t replyID; - CHECK(msg->senderAwaitsResponse(&replyID)); + CHECK(msg->senderAwaitsResponse(&mSeekReplyID)); status_t err = onSeek(msg); - sp<AMessage> response = new AMessage; - response->setInt32("err", err); - - response->postReply(replyID); + mSeekReply = new AMessage; + mSeekReply->setInt32("err", err); break; } @@ -240,13 +288,23 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { if (what == PlaylistFetcher::kWhatStopped) { AString uri; CHECK(msg->findString("uri", &uri)); - mFetcherInfos.removeItem(uri); + if (mFetcherInfos.removeItem(uri) < 0) { + // ignore duplicated kWhatStopped messages. + break; + } + + tryToFinishBandwidthSwitch(); } if (mContinuation != NULL) { CHECK_GT(mContinuationCounter, 0); if (--mContinuationCounter == 0) { mContinuation->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } break; @@ -276,6 +334,8 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { postPrepared(err); } + cancelBandwidthSwitch(); + mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err); mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err); @@ -314,6 +374,27 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { break; } + case PlaylistFetcher::kWhatStartedAt: + { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + + if (switchGeneration != mSwitchGeneration) { + break; + } + + // Resume fetcher for the original variant; the resumed fetcher should + // continue until the timestamps found in msg, which is stored by the + // new fetcher to indicate where the new variant has started buffering. + for (size_t i = 0; i < mFetcherInfos.size(); i++) { + const FetcherInfo info = mFetcherInfos.valueAt(i); + if (info.mToBeRemoved) { + info.mFetcher->resumeUntilAsync(msg); + } + } + break; + } + default: TRESPASS(); } @@ -358,6 +439,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) { break; } + case kWhatSwapped: + { + onSwapped(msg); + break; + } default: TRESPASS(); break; @@ -375,6 +461,12 @@ int LiveSession::SortByBandwidth(const BandwidthItem *a, const BandwidthItem *b) return 1; } +// static +LiveSession::StreamType LiveSession::indexToType(int idx) { + CHECK(idx >= 0 && idx < kMaxStreams); + return (StreamType)(1 << idx); +} + void LiveSession::onConnect(const sp<AMessage> &msg) { AString url; CHECK(msg->findString("url", &url)); @@ -462,6 +554,10 @@ void LiveSession::finishDisconnect() { // during disconnection either. cancelCheckBandwidthEvent(); + // Protect mPacketSources from a swapPacketSource race condition through disconnect. + // (finishDisconnect, onFinishDisconnect2) + cancelBandwidthSwitch(); + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { mFetcherInfos.valueAt(i).mFetcher->stopAsync(); } @@ -501,11 +597,13 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) { sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id()); notify->setString("uri", uri); + notify->setInt32("switchGeneration", mSwitchGeneration); FetcherInfo info; info.mFetcher = new PlaylistFetcher(notify, this, uri); info.mDurationUs = -1ll; info.mIsPrepared = false; + info.mToBeRemoved = false; looper()->registerHandler(info.mFetcher); mFetcherInfos.add(uri, info); @@ -513,54 +611,82 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) { return info.mFetcher; } -status_t LiveSession::fetchFile( +/* + * Illustration of parameters: + * + * 0 `range_offset` + * +------------+-------------------------------------------------------+--+--+ + * | | | next block to fetch | | | + * | | `source` handle => `out` buffer | | | | + * | `url` file |<--------- buffer size --------->|<--- `block_size` -->| | | + * | |<----------- `range_length` / buffer capacity ----------->| | + * |<------------------------------ file_size ------------------------------->| + * + * Special parameter values: + * - range_length == -1 means entire file + * - block_size == 0 means entire range + * + */ +ssize_t LiveSession::fetchFile( const char *url, sp<ABuffer> *out, int64_t range_offset, int64_t range_length, + uint32_t block_size, /* download block size */ + sp<DataSource> *source, /* to return and reuse source */ String8 *actualUrl) { - *out = NULL; + off64_t size; + sp<DataSource> temp_source; + if (source == NULL) { + source = &temp_source; + } - sp<DataSource> source; + if (*source == NULL) { + if (!strncasecmp(url, "file://", 7)) { + *source = new FileSource(url + 7); + } else if (strncasecmp(url, "http://", 7) + && strncasecmp(url, "https://", 8)) { + return ERROR_UNSUPPORTED; + } else { + KeyedVector<String8, String8> headers = mExtraHeaders; + if (range_offset > 0 || range_length >= 0) { + headers.add( + String8("Range"), + String8( + StringPrintf( + "bytes=%lld-%s", + range_offset, + range_length < 0 + ? "" : StringPrintf("%lld", + range_offset + range_length - 1).c_str()).c_str())); + } + status_t err = mHTTPDataSource->connect(url, &headers); - if (!strncasecmp(url, "file://", 7)) { - source = new FileSource(url + 7); - } else if (strncasecmp(url, "http://", 7) - && strncasecmp(url, "https://", 8)) { - return ERROR_UNSUPPORTED; - } else { - KeyedVector<String8, String8> headers = mExtraHeaders; - if (range_offset > 0 || range_length >= 0) { - headers.add( - String8("Range"), - String8( - StringPrintf( - "bytes=%lld-%s", - range_offset, - range_length < 0 - ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str())); - } - status_t err = mHTTPDataSource->connect(url, &headers); + if (err != OK) { + return err; + } - if (err != OK) { - return err; + *source = mHTTPDataSource; } - - source = mHTTPDataSource; } - off64_t size; - status_t err = source->getSize(&size); - - if (err != OK) { + status_t getSizeErr = (*source)->getSize(&size); + if (getSizeErr != OK) { size = 65536; } - sp<ABuffer> buffer = new ABuffer(size); - buffer->setRange(0, 0); + sp<ABuffer> buffer = *out != NULL ? *out : new ABuffer(size); + if (*out == NULL) { + buffer->setRange(0, 0); + } + ssize_t bytesRead = 0; + // adjust range_length if only reading partial block + if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) { + range_length = buffer->size() + block_size; + } for (;;) { + // Only resize when we don't know the size. size_t bufferRemaining = buffer->capacity() - buffer->size(); - - if (bufferRemaining == 0) { + if (bufferRemaining == 0 && getSizeErr != OK) { bufferRemaining = 32768; ALOGV("increasing download buffer to %zu bytes", @@ -585,7 +711,9 @@ status_t LiveSession::fetchFile( } } - ssize_t n = source->readAt( + // The DataSource is responsible for informing us of error (n < 0) or eof (n == 0) + // to help us break out of the loop. + ssize_t n = (*source)->readAt( buffer->size(), buffer->data() + buffer->size(), maxBytesToRead); @@ -598,17 +726,18 @@ status_t LiveSession::fetchFile( } buffer->setRange(0, buffer->size() + (size_t)n); + bytesRead += n; } *out = buffer; if (actualUrl != NULL) { - *actualUrl = source->getUri(); + *actualUrl = (*source)->getUri(); if (actualUrl->isEmpty()) { *actualUrl = url; } } - return OK; + return bytesRead; } sp<M3UParser> LiveSession::fetchPlaylist( @@ -619,9 +748,9 @@ sp<M3UParser> LiveSession::fetchPlaylist( sp<ABuffer> buffer; String8 actualUrl; - status_t err = fetchFile(url, &buffer, 0, -1, &actualUrl); + ssize_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl); - if (err != OK) { + if (err <= 0) { return NULL; } @@ -816,8 +945,25 @@ status_t LiveSession::selectTrack(size_t index, bool select) { return err; } +bool LiveSession::canSwitchUp() { + // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds. + status_t err = OK; + for (size_t i = 0; i < mPacketSources.size(); ++i) { + sp<AnotherPacketSource> source = mPacketSources.valueAt(i); + int64_t dur = source->getBufferedDurationUs(&err); + if (err == OK && dur > 10000000) { + return true; + } + } + return false; +} + void LiveSession::changeConfiguration( int64_t timeUs, size_t bandwidthIndex, bool pickTrack) { + // Protect mPacketSources from a swapPacketSource race condition through reconfiguration. + // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3). + cancelBandwidthSwitch(); + CHECK(!mReconfigurationInProgress); mReconfigurationInProgress = true; @@ -833,21 +979,14 @@ void LiveSession::changeConfiguration( CHECK_LT(bandwidthIndex, mBandwidthItems.size()); const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex); - uint32_t streamMask = 0; - - AString audioURI; - if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) { - streamMask |= STREAMTYPE_AUDIO; - } - - AString videoURI; - if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) { - streamMask |= STREAMTYPE_VIDEO; - } + uint32_t streamMask = 0; // streams that should be fetched by the new fetcher + uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher - AString subtitleURI; - if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) { - streamMask |= STREAMTYPE_SUBTITLES; + AString URIs[kMaxStreams]; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (mPlaylist->getTypeURI(item.mPlaylistIndex, mStreams[i].mType, &URIs[i])) { + streamMask |= indexToType(i); + } } // Step 1, stop and discard fetchers that are no longer needed. @@ -859,10 +998,15 @@ void LiveSession::changeConfiguration( // If we're seeking all current fetchers are discarded. if (timeUs < 0ll) { - if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) - || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) - || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) { - discardFetcher = false; + // delay fetcher removal + discardFetcher = false; + + for (size_t j = 0; j < kMaxStreams; ++j) { + StreamType type = indexToType(j); + if ((streamMask & type) && uri == URIs[j]) { + resumeMask |= type; + streamMask &= ~type; + } } } @@ -873,17 +1017,20 @@ void LiveSession::changeConfiguration( } } - sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id()); + sp<AMessage> msg; + if (timeUs < 0ll) { + // skip onChangeConfiguration2 (decoder destruction) if switching. + msg = new AMessage(kWhatChangeConfiguration3, id()); + } else { + msg = new AMessage(kWhatChangeConfiguration2, id()); + } msg->setInt32("streamMask", streamMask); + msg->setInt32("resumeMask", resumeMask); msg->setInt64("timeUs", timeUs); - if (streamMask & STREAMTYPE_AUDIO) { - msg->setString("audioURI", audioURI.c_str()); - } - if (streamMask & STREAMTYPE_VIDEO) { - msg->setString("videoURI", videoURI.c_str()); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - msg->setString("subtitleURI", subtitleURI.c_str()); + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + msg->setString(mStreams[i].uriKey().c_str(), URIs[i].c_str()); + } } // Every time a fetcher acknowledges the stopAsync or pauseAsync request @@ -895,6 +1042,11 @@ void LiveSession::changeConfiguration( if (mContinuationCounter == 0) { msg->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } @@ -914,18 +1066,13 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { uint32_t streamMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); - AString audioURI, videoURI, subtitleURI; - if (streamMask & STREAMTYPE_AUDIO) { - CHECK(msg->findString("audioURI", &audioURI)); - ALOGV("audioURI = '%s'", audioURI.c_str()); - } - if (streamMask & STREAMTYPE_VIDEO) { - CHECK(msg->findString("videoURI", &videoURI)); - ALOGV("videoURI = '%s'", videoURI.c_str()); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - CHECK(msg->findString("subtitleURI", &subtitleURI)); - ALOGV("subtitleURI = '%s'", subtitleURI.c_str()); + AString URIs[kMaxStreams]; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + const AString &uriKey = mStreams[i].uriKey(); + CHECK(msg->findString(uriKey.c_str(), &URIs[i])); + ALOGV("%s = '%s'", uriKey.c_str(), URIs[i].c_str()); + } } // Determine which decoders to shutdown on the player side, @@ -935,15 +1082,12 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { // 2) its streamtype was already active and still is but the URI // has changed. uint32_t changedMask = 0; - if (((mStreamMask & streamMask & STREAMTYPE_AUDIO) - && !(audioURI == mAudioURI)) - || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) { - changedMask |= STREAMTYPE_AUDIO; - } - if (((mStreamMask & streamMask & STREAMTYPE_VIDEO) - && !(videoURI == mVideoURI)) - || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) { - changedMask |= STREAMTYPE_VIDEO; + for (size_t i = 0; i < kMaxStreams && i != kSubtitleIndex; ++i) { + if (((mStreamMask & streamMask & indexToType(i)) + && !(URIs[i] == mStreams[i].mUri)) + || (mStreamMask & ~streamMask & indexToType(i))) { + changedMask |= indexToType(i); + } } if (changedMask == 0) { @@ -969,68 +1113,54 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { } void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { + mContinuation.clear(); // All remaining fetchers are still suspended, the player has shutdown // any decoders that needed it. - uint32_t streamMask; + uint32_t streamMask, resumeMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); + CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask)); - AString audioURI, videoURI, subtitleURI; - if (streamMask & STREAMTYPE_AUDIO) { - CHECK(msg->findString("audioURI", &audioURI)); - } - if (streamMask & STREAMTYPE_VIDEO) { - CHECK(msg->findString("videoURI", &videoURI)); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - CHECK(msg->findString("subtitleURI", &subtitleURI)); + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + CHECK(msg->findString(mStreams[i].uriKey().c_str(), &mStreams[i].mUri)); + } } int64_t timeUs; + bool switching = false; CHECK(msg->findInt64("timeUs", &timeUs)); if (timeUs < 0ll) { timeUs = mLastDequeuedTimeUs; + switching = true; } mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; - mStreamMask = streamMask; - mAudioURI = audioURI; - mVideoURI = videoURI; - mSubtitleURI = subtitleURI; + mNewStreamMask = streamMask; - // Resume all existing fetchers and assign them packet sources. + // Of all existing fetchers: + // * Resume fetchers that are still needed and assign them original packet sources. + // * Mark otherwise unneeded fetchers for removal. + ALOGV("resuming fetchers for mask 0x%08x", resumeMask); for (size_t i = 0; i < mFetcherInfos.size(); ++i) { const AString &uri = mFetcherInfos.keyAt(i); - uint32_t resumeMask = 0; - - sp<AnotherPacketSource> audioSource; - if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { - audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); - resumeMask |= STREAMTYPE_AUDIO; - } - - sp<AnotherPacketSource> videoSource; - if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { - videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); - resumeMask |= STREAMTYPE_VIDEO; + sp<AnotherPacketSource> sources[kMaxStreams]; + for (size_t j = 0; j < kMaxStreams; ++j) { + if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) { + sources[j] = mPacketSources.valueFor(indexToType(j)); + } } - sp<AnotherPacketSource> subtitleSource; - if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { - subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); - resumeMask |= STREAMTYPE_SUBTITLES; + FetcherInfo &info = mFetcherInfos.editValueAt(i); + if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL + || sources[kSubtitleIndex] != NULL) { + info.mFetcher->startAsync( + sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); + } else { + info.mToBeRemoved = true; } - - CHECK_NE(resumeMask, 0u); - - ALOGV("resuming fetchers for mask 0x%08x", resumeMask); - - streamMask &= ~resumeMask; - - mFetcherInfos.valueAt(i).mFetcher->startAsync( - audioSource, videoSource, subtitleSource); } // streamMask now only contains the types that need a new fetcher created. @@ -1039,52 +1169,65 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { ALOGV("creating new fetchers for mask 0x%08x", streamMask); } - while (streamMask != 0) { - StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1)); + // Find out when the original fetchers have buffered up to and start the new fetchers + // at a later timestamp. + for (size_t i = 0; i < kMaxStreams; i++) { + if (!(indexToType(i) & streamMask)) { + continue; + } AString uri; - switch (streamType) { - case STREAMTYPE_AUDIO: - uri = audioURI; - break; - case STREAMTYPE_VIDEO: - uri = videoURI; - break; - case STREAMTYPE_SUBTITLES: - uri = subtitleURI; - break; - default: - TRESPASS(); - } + uri = mStreams[i].mUri; sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str()); CHECK(fetcher != NULL); - sp<AnotherPacketSource> audioSource; - if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { - audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); - audioSource->clear(); - - streamMask &= ~STREAMTYPE_AUDIO; - } - - sp<AnotherPacketSource> videoSource; - if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { - videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); - videoSource->clear(); - - streamMask &= ~STREAMTYPE_VIDEO; - } + int32_t latestSeq = -1; + int64_t latestTimeUs = 0ll; + sp<AnotherPacketSource> sources[kMaxStreams]; + + // TRICKY: looping from i as earlier streams are already removed from streamMask + for (size_t j = i; j < kMaxStreams; ++j) { + if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { + sources[j] = mPacketSources.valueFor(indexToType(j)); + + if (!switching) { + sources[j]->clear(); + } else { + int32_t type, seq; + int64_t srcTimeUs; + sp<AMessage> meta = sources[j]->getLatestMeta(); + + if (meta != NULL && !meta->findInt32("discontinuity", &type)) { + CHECK(meta->findInt32("seq", &seq)); + if (seq > latestSeq) { + latestSeq = seq; + } + CHECK(meta->findInt64("timeUs", &srcTimeUs)); + if (srcTimeUs > latestTimeUs) { + latestTimeUs = srcTimeUs; + } + } - sp<AnotherPacketSource> subtitleSource; - if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { - subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); - subtitleSource->clear(); + sources[j] = mPacketSources2.valueFor(indexToType(j)); + sources[j]->clear(); + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + if (extraStreams & indexToType(j)) { + sources[j]->queueAccessUnit(createFormatChangeBuffer(/* swap = */ false)); + } + } - streamMask &= ~STREAMTYPE_SUBTITLES; + streamMask &= ~indexToType(j); + } } - fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs); + fetcher->startAsync( + sources[kAudioIndex], + sources[kVideoIndex], + sources[kSubtitleIndex], + timeUs, + latestTimeUs /* min start time(us) */, + latestSeq >= 0 ? latestSeq + 1 : -1 /* starting sequence number hint */ ); } // All fetchers have now been started, the configuration change @@ -1093,14 +1236,61 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { scheduleCheckBandwidthEvent(); ALOGV("XXX configuration change completed."); - mReconfigurationInProgress = false; + if (switching) { + mSwitchInProgress = true; + } else { + mStreamMask = mNewStreamMask; + } if (mDisconnectReplyID != 0) { finishDisconnect(); } } +void LiveSession::onSwapped(const sp<AMessage> &msg) { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + if (switchGeneration != mSwitchGeneration) { + return; + } + + int32_t stream; + CHECK(msg->findInt32("stream", &stream)); + mSwapMask |= stream; + if (mSwapMask != mStreamMask) { + return; + } + + // Check if new variant contains extra streams. + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + while (extraStreams) { + StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1)); + swapPacketSource(extraStream); + extraStreams &= ~extraStream; + } + + tryToFinishBandwidthSwitch(); +} + +// Mark switch done when: +// 1. all old buffers are swapped out, AND +// 2. all old fetchers are removed. +void LiveSession::tryToFinishBandwidthSwitch() { + bool needToRemoveFetchers = false; + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + if (mFetcherInfos.valueAt(i).mToBeRemoved) { + needToRemoveFetchers = true; + break; + } + } + if (!needToRemoveFetchers && mSwapMask == mStreamMask) { + mStreamMask = mNewStreamMask; + mSwitchInProgress = false; + mSwapMask = 0; + } +} + void LiveSession::scheduleCheckBandwidthEvent() { sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id()); msg->setInt32("generation", mCheckBandwidthGeneration); @@ -1111,16 +1301,37 @@ void LiveSession::cancelCheckBandwidthEvent() { ++mCheckBandwidthGeneration; } -void LiveSession::onCheckBandwidth() { - if (mReconfigurationInProgress) { - scheduleCheckBandwidthEvent(); - return; +void LiveSession::cancelBandwidthSwitch() { + Mutex::Autolock lock(mSwapMutex); + mSwitchGeneration++; + mSwitchInProgress = false; + mSwapMask = 0; +} + +bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) { + if (mReconfigurationInProgress || mSwitchInProgress) { + return false; } + if (mPrevBandwidthIndex < 0) { + return true; + } + + if (bandwidthIndex == (size_t)mPrevBandwidthIndex) { + return false; + } else if (bandwidthIndex > (size_t)mPrevBandwidthIndex) { + return canSwitchUp(); + } else { + return true; + } +} + +void LiveSession::onCheckBandwidth() { size_t bandwidthIndex = getBandwidthIndex(); - if (mPrevBandwidthIndex < 0 - || bandwidthIndex != (size_t)mPrevBandwidthIndex) { + if (canSwitchBandwidthTo(bandwidthIndex)) { changeConfiguration(-1ll /* timeUs */, bandwidthIndex); + } else { + scheduleCheckBandwidthEvent(); } // Handling the kWhatCheckBandwidth even here does _not_ automatically diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index 8f6a4ea..3f8fee5 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -42,10 +42,17 @@ struct LiveSession : public AHandler { const sp<AMessage> ¬ify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0); + enum StreamIndex { + kAudioIndex = 0, + kVideoIndex = 1, + kSubtitleIndex = 2, + kMaxStreams = 3, + }; + enum StreamType { - STREAMTYPE_AUDIO = 1, - STREAMTYPE_VIDEO = 2, - STREAMTYPE_SUBTITLES = 4, + STREAMTYPE_AUDIO = 1 << kAudioIndex, + STREAMTYPE_VIDEO = 1 << kVideoIndex, + STREAMTYPE_SUBTITLES = 1 << kSubtitleIndex, }; status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit); @@ -74,6 +81,11 @@ struct LiveSession : public AHandler { kWhatPreparationFailed, }; + // create a format-change discontinuity + // + // swap: + // whether is format-change discontinuity should trigger a buffer swap + sp<ABuffer> createFormatChangeBuffer(bool swap = true); protected: virtual ~LiveSession(); @@ -92,6 +104,7 @@ private: kWhatChangeConfiguration2 = 'chC2', kWhatChangeConfiguration3 = 'chC3', kWhatFinishDisconnect2 = 'fin2', + kWhatSwapped = 'swap', }; struct BandwidthItem { @@ -103,8 +116,22 @@ private: sp<PlaylistFetcher> mFetcher; int64_t mDurationUs; bool mIsPrepared; + bool mToBeRemoved; }; + struct StreamItem { + const char *mType; + AString mUri; + StreamItem() : mType("") {} + StreamItem(const char *type) : mType(type) {} + AString uriKey() { + AString key(mType); + key.append("URI"); + return key; + } + }; + StreamItem mStreams[kMaxStreams]; + sp<AMessage> mNotify; uint32_t mFlags; bool mUIDValid; @@ -123,21 +150,40 @@ private: sp<M3UParser> mPlaylist; KeyedVector<AString, FetcherInfo> mFetcherInfos; - AString mAudioURI, mVideoURI, mSubtitleURI; uint32_t mStreamMask; + // Masks used during reconfiguration: + // mNewStreamMask: streams in the variant playlist we're switching to; + // we don't want to immediately overwrite the original value. + uint32_t mNewStreamMask; + + // mSwapMask: streams that have started to playback content in the new variant playlist; + // we use this to track reconfiguration progress. + uint32_t mSwapMask; + KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources; + // A second set of packet sources that buffer content for the variant we're switching to. + KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources2; + + // A mutex used to serialize two sets of events: + // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND + // * a forced bandwidth switch termination in cancelSwitch on the live looper. + Mutex mSwapMutex; int32_t mCheckBandwidthGeneration; + int32_t mSwitchGeneration; size_t mContinuationCounter; sp<AMessage> mContinuation; + sp<AMessage> mSeekReply; int64_t mLastDequeuedTimeUs; int64_t mRealTimeBaseUs; bool mReconfigurationInProgress; + bool mSwitchInProgress; uint32_t mDisconnectReplyID; + uint32_t mSeekReplyID; sp<PlaylistFetcher> addFetcher(const char *uri); @@ -145,9 +191,25 @@ private: status_t onSeek(const sp<AMessage> &msg); void onFinishDisconnect2(); - status_t fetchFile( + // If given a non-zero block_size (default 0), it is used to cap the number of + // bytes read in from the DataSource. If given a non-NULL buffer, new content + // is read into the end. + // + // The DataSource we read from is responsible for signaling error or EOF to help us + // break out of the read loop. The DataSource can be returned to the caller, so + // that the caller can reuse it for subsequent fetches (within the initially + // requested range). + // + // For reused HTTP sources, the caller must download a file sequentially without + // any overlaps or gaps to prevent reconnection. + ssize_t fetchFile( const char *url, sp<ABuffer> *out, + /* request/open a file starting at range_offset for range_length bytes */ int64_t range_offset = 0, int64_t range_length = -1, + /* download block size */ + uint32_t block_size = 0, + /* reuse DataSource if doing partial fetch */ + sp<DataSource> *source = NULL, String8 *actualUrl = NULL); sp<M3UParser> fetchPlaylist( @@ -156,22 +218,34 @@ private: size_t getBandwidthIndex(); static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *); + static StreamType indexToType(int idx); void changeConfiguration( int64_t timeUs, size_t bandwidthIndex, bool pickTrack = false); void onChangeConfiguration(const sp<AMessage> &msg); void onChangeConfiguration2(const sp<AMessage> &msg); void onChangeConfiguration3(const sp<AMessage> &msg); + void onSwapped(const sp<AMessage> &msg); + void tryToFinishBandwidthSwitch(); void scheduleCheckBandwidthEvent(); void cancelCheckBandwidthEvent(); + // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources + // from being swapped out on stale discontinuities while manipulating + // mPacketSources/mPacketSources2. + void cancelBandwidthSwitch(); + + bool canSwitchBandwidthTo(size_t bandwidthIndex); void onCheckBandwidth(); void finishDisconnect(); void postPrepared(status_t err); + void swapPacketSource(StreamType stream); + bool canSwitchUp(); + DISALLOW_EVIL_CONSTRUCTORS(LiveSession); }; diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index b2a7010..20c3a76 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -24,6 +24,7 @@ #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> #include <media/stagefright/MediaErrors.h> +#include <media/stagefright/Utils.h> #include <media/mediaplayer.h> namespace android { @@ -352,9 +353,28 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { if (!meta->findString(key, &groupID)) { *uri = mItems.itemAt(index).mURI; - // Assume media without any more specific attribute contains - // audio and video, but no subtitles. - return !strcmp("audio", key) || !strcmp("video", key); + AString codecs; + if (!meta->findString("codecs", &codecs)) { + // Assume media without any more specific attribute contains + // audio and video, but no subtitles. + return !strcmp("audio", key) || !strcmp("video", key); + } else { + // Split the comma separated list of codecs. + size_t offset = 0; + ssize_t commaPos = -1; + codecs.append(','); + while ((commaPos = codecs.find(",", offset)) >= 0) { + AString codec(codecs, offset, commaPos - offset); + codec.trim(); + // return true only if a codec of type `key` ("audio"/"video") + // is found. + if (codecIsType(codec, key)) { + return true; + } + offset = commaPos + 1; + } + return false; + } } sp<MediaGroup> group = mMediaGroups.valueFor(groupID); @@ -369,18 +389,6 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { return true; } -bool M3UParser::getAudioURI(size_t index, AString *uri) const { - return getTypeURI(index, "audio", uri); -} - -bool M3UParser::getVideoURI(size_t index, AString *uri) const { - return getTypeURI(index, "video", uri); -} - -bool M3UParser::getSubtitleURI(size_t index, AString *uri) const { - return getTypeURI(index, "subtitles", uri); -} - static bool MakeURL(const char *baseURL, const char *url, AString *out) { out->clear(); @@ -694,12 +702,22 @@ status_t M3UParser::parseStreamInf( *meta = new AMessage; } (*meta)->setInt32("bandwidth", x); + } else if (!strcasecmp("codecs", key.c_str())) { + if (!isQuotedString(val)) { + ALOGE("Expected quoted string for %s attribute, " + "got '%s' instead.", + key.c_str(), val.c_str());; + + return ERROR_MALFORMED; + } + + key.tolower(); + const AString &codecs = unquoteString(val); + (*meta)->setString(key.c_str(), codecs.c_str()); } else if (!strcasecmp("audio", key.c_str()) || !strcasecmp("video", key.c_str()) || !strcasecmp("subtitles", key.c_str())) { - if (val.size() < 2 - || val.c_str()[0] != '"' - || val.c_str()[val.size() - 1] != '"') { + if (!isQuotedString(val)) { ALOGE("Expected quoted string for %s attribute, " "got '%s' instead.", key.c_str(), val.c_str()); @@ -707,7 +725,7 @@ status_t M3UParser::parseStreamInf( return ERROR_MALFORMED; } - AString groupID(val, 1, val.size() - 2); + const AString &groupID = unquoteString(val); ssize_t groupIndex = mMediaGroups.indexOfKey(groupID); if (groupIndex < 0) { @@ -1095,4 +1113,121 @@ status_t M3UParser::ParseDouble(const char *s, double *x) { return OK; } +// static +bool M3UParser::isQuotedString(const AString &str) { + if (str.size() < 2 + || str.c_str()[0] != '"' + || str.c_str()[str.size() - 1] != '"') { + return false; + } + return true; +} + +// static +AString M3UParser::unquoteString(const AString &str) { + if (!isQuotedString(str)) { + return str; + } + return AString(str, 1, str.size() - 2); +} + +// static +bool M3UParser::codecIsType(const AString &codec, const char *type) { + if (codec.size() < 4) { + return false; + } + const char *c = codec.c_str(); + switch (FOURCC(c[0], c[1], c[2], c[3])) { + // List extracted from http://www.mp4ra.org/codecs.html + case 'ac-3': + case 'alac': + case 'dra1': + case 'dtsc': + case 'dtse': + case 'dtsh': + case 'dtsl': + case 'ec-3': + case 'enca': + case 'g719': + case 'g726': + case 'm4ae': + case 'mlpa': + case 'mp4a': + case 'raw ': + case 'samr': + case 'sawb': + case 'sawp': + case 'sevc': + case 'sqcp': + case 'ssmv': + case 'twos': + case 'agsm': + case 'alaw': + case 'dvi ': + case 'fl32': + case 'fl64': + case 'ima4': + case 'in24': + case 'in32': + case 'lpcm': + case 'Qclp': + case 'QDM2': + case 'QDMC': + case 'ulaw': + case 'vdva': + return !strcmp("audio", type); + + case 'avc1': + case 'avc2': + case 'avcp': + case 'drac': + case 'encv': + case 'mjp2': + case 'mp4v': + case 'mvc1': + case 'mvc2': + case 'resv': + case 's263': + case 'svc1': + case 'vc-1': + case 'CFHD': + case 'civd': + case 'DV10': + case 'dvh5': + case 'dvh6': + case 'dvhp': + case 'DVOO': + case 'DVOR': + case 'DVTV': + case 'DVVT': + case 'flic': + case 'gif ': + case 'h261': + case 'h263': + case 'HD10': + case 'jpeg': + case 'M105': + case 'mjpa': + case 'mjpb': + case 'png ': + case 'PNTG': + case 'rle ': + case 'rpza': + case 'Shr0': + case 'Shr1': + case 'Shr2': + case 'Shr3': + case 'Shr4': + case 'SVQ1': + case 'SVQ3': + case 'tga ': + case 'tiff': + case 'WRLE': + return !strcmp("video", type); + + default: + return false; + } +} + } // namespace android diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h index 5248004..ccd6556 100644 --- a/media/libstagefright/httplive/M3UParser.h +++ b/media/libstagefright/httplive/M3UParser.h @@ -45,9 +45,7 @@ struct M3UParser : public RefBase { status_t getTrackInfo(Parcel* reply) const; ssize_t getSelectedIndex() const; - bool getAudioURI(size_t index, AString *uri) const; - bool getVideoURI(size_t index, AString *uri) const; - bool getSubtitleURI(size_t index, AString *uri) const; + bool getTypeURI(size_t index, const char *key, AString *uri) const; protected: virtual ~M3UParser(); @@ -95,11 +93,13 @@ private: status_t parseMedia(const AString &line); - bool getTypeURI(size_t index, const char *key, AString *uri) const; - static status_t ParseInt32(const char *s, int32_t *x); static status_t ParseDouble(const char *s, double *x); + static bool isQuotedString(const AString &str); + static AString unquoteString(const AString &str); + static bool codecIsType(const AString &codec, const char *type); + DISALLOW_EVIL_CONSTRUCTORS(M3UParser); }; diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 57bf7db..513f114 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -48,26 +48,35 @@ namespace android { // static const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; +const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; +const int32_t PlaylistFetcher::kDownloadBlockSize = 192; +const int32_t PlaylistFetcher::kNumSkipFrames = 10; PlaylistFetcher::PlaylistFetcher( const sp<AMessage> ¬ify, const sp<LiveSession> &session, const char *uri) : mNotify(notify), + mStartTimeUsNotify(notify->dup()), mSession(session), mURI(uri), mStreamTypeMask(0), mStartTimeUs(-1ll), + mMinStartTimeUs(0ll), + mStopParams(NULL), mLastPlaylistFetchTimeUs(-1ll), mSeqNumber(-1), mNumRetries(0), mStartup(true), + mPrepared(false), mNextPTSTimeUs(-1ll), mMonitorQueueGeneration(0), mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), mFirstPTSValid(false), mAbsoluteTimeAnchorUs(0ll) { memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); + mStartTimeUsNotify->setInt32("what", kWhatStartedAt); + mStartTimeUsNotify->setInt32("streamMask", 0); } PlaylistFetcher::~PlaylistFetcher() { @@ -104,10 +113,16 @@ int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { return segmentStartUs; } -bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { - if (mPlaylist == NULL) { +int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const { + int64_t nowUs = ALooper::GetNowUs(); + + if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) { CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); - return true; + return 0ll; + } + + if (mPlaylist->isComplete()) { + return (~0llu >> 1); } int32_t targetDurationSecs; @@ -158,11 +173,13 @@ bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { break; } - return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs; + int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs; + return delayUs > 0ll ? delayUs : 0ll; } status_t PlaylistFetcher::decryptBuffer( - size_t playlistIndex, const sp<ABuffer> &buffer) { + size_t playlistIndex, const sp<ABuffer> &buffer, + bool first) { sp<AMessage> itemMeta; bool found = false; AString method; @@ -180,6 +197,7 @@ status_t PlaylistFetcher::decryptBuffer( if (!found) { method = "NONE"; } + buffer->meta()->setString("cipher-method", method.c_str()); if (method == "NONE") { return OK; @@ -200,9 +218,9 @@ status_t PlaylistFetcher::decryptBuffer( if (index >= 0) { key = mAESKeyForURI.valueAt(index); } else { - status_t err = mSession->fetchFile(keyURI.c_str(), &key); + ssize_t err = mSession->fetchFile(keyURI.c_str(), &key); - if (err != OK) { + if (err < 0) { ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); return ERROR_IO; } else if (key->size() != 16) { @@ -219,63 +237,89 @@ status_t PlaylistFetcher::decryptBuffer( return UNKNOWN_ERROR; } - unsigned char aes_ivec[16]; + size_t n = buffer->size(); + if (!n) { + return OK; + } + CHECK(n % 16 == 0); - AString iv; - if (itemMeta->findString("cipher-iv", &iv)) { - if ((!iv.startsWith("0x") && !iv.startsWith("0X")) - || iv.size() != 16 * 2 + 2) { - ALOGE("malformed cipher IV '%s'.", iv.c_str()); - return ERROR_MALFORMED; - } + if (first) { + // If decrypting the first block in a file, read the iv from the manifest + // or derive the iv from the file's sequence number. - memset(aes_ivec, 0, sizeof(aes_ivec)); - for (size_t i = 0; i < 16; ++i) { - char c1 = tolower(iv.c_str()[2 + 2 * i]); - char c2 = tolower(iv.c_str()[3 + 2 * i]); - if (!isxdigit(c1) || !isxdigit(c2)) { + AString iv; + if (itemMeta->findString("cipher-iv", &iv)) { + if ((!iv.startsWith("0x") && !iv.startsWith("0X")) + || iv.size() != 16 * 2 + 2) { ALOGE("malformed cipher IV '%s'.", iv.c_str()); return ERROR_MALFORMED; } - uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; - uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; - aes_ivec[i] = nibble1 << 4 | nibble2; + memset(mAESInitVec, 0, sizeof(mAESInitVec)); + for (size_t i = 0; i < 16; ++i) { + char c1 = tolower(iv.c_str()[2 + 2 * i]); + char c2 = tolower(iv.c_str()[3 + 2 * i]); + if (!isxdigit(c1) || !isxdigit(c2)) { + ALOGE("malformed cipher IV '%s'.", iv.c_str()); + return ERROR_MALFORMED; + } + uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; + uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; + + mAESInitVec[i] = nibble1 << 4 | nibble2; + } + } else { + memset(mAESInitVec, 0, sizeof(mAESInitVec)); + mAESInitVec[15] = mSeqNumber & 0xff; + mAESInitVec[14] = (mSeqNumber >> 8) & 0xff; + mAESInitVec[13] = (mSeqNumber >> 16) & 0xff; + mAESInitVec[12] = (mSeqNumber >> 24) & 0xff; } - } else { - memset(aes_ivec, 0, sizeof(aes_ivec)); - aes_ivec[15] = mSeqNumber & 0xff; - aes_ivec[14] = (mSeqNumber >> 8) & 0xff; - aes_ivec[13] = (mSeqNumber >> 16) & 0xff; - aes_ivec[12] = (mSeqNumber >> 24) & 0xff; } AES_cbc_encrypt( buffer->data(), buffer->data(), buffer->size(), - &aes_key, aes_ivec, AES_DECRYPT); - - // hexdump(buffer->data(), buffer->size()); + &aes_key, mAESInitVec, AES_DECRYPT); - size_t n = buffer->size(); - CHECK_GT(n, 0u); + return OK; +} - size_t pad = buffer->data()[n - 1]; +status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) { + status_t err; + AString method; + CHECK(buffer->meta()->findString("cipher-method", &method)); + if (method == "NONE") { + return OK; + } - CHECK_GT(pad, 0u); - CHECK_LE(pad, 16u); - CHECK_GE((size_t)n, pad); - for (size_t i = 0; i < pad; ++i) { - CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad); + uint8_t padding = 0; + if (buffer->size() > 0) { + padding = buffer->data()[buffer->size() - 1]; } - n -= pad; + if (padding > 16) { + return ERROR_MALFORMED; + } - buffer->setRange(buffer->offset(), n); + for (size_t i = buffer->size() - padding; i < padding; i++) { + if (buffer->data()[i] != padding) { + return ERROR_MALFORMED; + } + } + buffer->setRange(buffer->offset(), buffer->size() - padding); return OK; } -void PlaylistFetcher::postMonitorQueue(int64_t delayUs) { +void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) { + int64_t maxDelayUs = delayUsToRefreshPlaylist(); + if (maxDelayUs < minDelayUs) { + maxDelayUs = minDelayUs; + } + if (delayUs > maxDelayUs) { + ALOGV("Need to refresh playlist in %lld", maxDelayUs); + delayUs = maxDelayUs; + } sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); msg->setInt32("generation", mMonitorQueueGeneration); msg->post(delayUs); @@ -289,7 +333,9 @@ void PlaylistFetcher::startAsync( const sp<AnotherPacketSource> &audioSource, const sp<AnotherPacketSource> &videoSource, const sp<AnotherPacketSource> &subtitleSource, - int64_t startTimeUs) { + int64_t startTimeUs, + int64_t minStartTimeUs, + int32_t startSeqNumberHint) { sp<AMessage> msg = new AMessage(kWhatStart, id()); uint32_t streamTypeMask = 0ul; @@ -311,6 +357,8 @@ void PlaylistFetcher::startAsync( msg->setInt32("streamTypeMask", streamTypeMask); msg->setInt64("startTimeUs", startTimeUs); + msg->setInt64("minStartTimeUs", minStartTimeUs); + msg->setInt32("startSeqNumberHint", startSeqNumberHint); msg->post(); } @@ -318,8 +366,16 @@ void PlaylistFetcher::pauseAsync() { (new AMessage(kWhatPause, id()))->post(); } -void PlaylistFetcher::stopAsync() { - (new AMessage(kWhatStop, id()))->post(); +void PlaylistFetcher::stopAsync(bool selfTriggered) { + sp<AMessage> msg = new AMessage(kWhatStop, id()); + msg->setInt32("selfTriggered", selfTriggered); + msg->post(); +} + +void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) { + AMessage* msg = new AMessage(kWhatResumeUntil, id()); + msg->setMessage("params", params); + msg->post(); } void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { @@ -347,7 +403,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { case kWhatStop: { - onStop(); + onStop(msg); sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatStopped); @@ -356,6 +412,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { } case kWhatMonitorQueue: + case kWhatDownloadNext: { int32_t generation; CHECK(msg->findInt32("generation", &generation)); @@ -365,7 +422,17 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { break; } - onMonitorQueue(); + if (msg->what() == kWhatMonitorQueue) { + onMonitorQueue(); + } else { + onDownloadNext(); + } + break; + } + + case kWhatResumeUntil: + { + onResumeUntil(msg); break; } @@ -381,7 +448,10 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask)); int64_t startTimeUs; + int32_t startSeqNumberHint; CHECK(msg->findInt64("startTimeUs", &startTimeUs)); + CHECK(msg->findInt64("minStartTimeUs", (int64_t *) &mMinStartTimeUs)); + CHECK(msg->findInt32("startSeqNumberHint", &startSeqNumberHint)); if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) { void *ptr; @@ -416,6 +486,11 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { if (mStartTimeUs >= 0ll) { mSeqNumber = -1; mStartup = true; + mPrepared = false; + } + + if (startSeqNumberHint >= 0) { + mSeqNumber = startSeqNumberHint; } postMonitorQueue(); @@ -425,22 +500,83 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { void PlaylistFetcher::onPause() { cancelMonitorQueue(); - - mPacketSources.clear(); - mStreamTypeMask = 0; } -void PlaylistFetcher::onStop() { +void PlaylistFetcher::onStop(const sp<AMessage> &msg) { cancelMonitorQueue(); - for (size_t i = 0; i < mPacketSources.size(); ++i) { - mPacketSources.valueAt(i)->clear(); + int32_t selfTriggered; + CHECK(msg->findInt32("selfTriggered", &selfTriggered)); + if (!selfTriggered) { + // Self triggered stops only happen during switching, in which case we do not want + // to clear the discontinuities queued at the end of packet sources. + for (size_t i = 0; i < mPacketSources.size(); i++) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + packetSource->clear(); + } } mPacketSources.clear(); mStreamTypeMask = 0; } +// Resume until we have reached the boundary timestamps listed in `msg`; when +// the remaining time is too short (within a resume threshold) stop immediately +// instead. +status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) { + sp<AMessage> params; + CHECK(msg->findMessage("params", ¶ms)); + + bool stop = false; + for (size_t i = 0; i < mPacketSources.size(); i++) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + + const char *stopKey; + int streamType = mPacketSources.keyAt(i); + switch (streamType) { + case LiveSession::STREAMTYPE_VIDEO: + stopKey = "timeUsVideo"; + break; + + case LiveSession::STREAMTYPE_AUDIO: + stopKey = "timeUsAudio"; + break; + + case LiveSession::STREAMTYPE_SUBTITLES: + stopKey = "timeUsSubtitle"; + break; + + default: + TRESPASS(); + } + + // Don't resume if we would stop within a resume threshold. + int64_t latestTimeUs = 0, stopTimeUs = 0; + sp<AMessage> latestMeta = packetSource->getLatestMeta(); + if (latestMeta != NULL + && (latestMeta->findInt64("timeUs", &latestTimeUs) + && params->findInt64(stopKey, &stopTimeUs))) { + int64_t diffUs = stopTimeUs - latestTimeUs; + if (diffUs < resumeThreshold(latestMeta)) { + stop = true; + } + } + } + + if (stop) { + for (size_t i = 0; i < mPacketSources.size(); i++) { + mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); + } + stopAsync(/* selfTriggered = */ true); + return OK; + } + + mStopParams = params; + postMonitorQueue(); + + return OK; +} + void PlaylistFetcher::notifyError(status_t err) { sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatError); @@ -457,41 +593,70 @@ void PlaylistFetcher::queueDiscontinuity( void PlaylistFetcher::onMonitorQueue() { bool downloadMore = false; + refreshPlaylist(); + + int32_t targetDurationSecs; + int64_t targetDurationUs = kMinBufferedDurationUs; + if (mPlaylist != NULL) { + CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); + targetDurationUs = targetDurationSecs * 1000000ll; + } - status_t finalResult; + // buffer at least 3 times the target duration, or up to 10 seconds + int64_t durationToBufferUs = targetDurationUs * 3; + if (durationToBufferUs > kMinBufferedDurationUs) { + durationToBufferUs = kMinBufferedDurationUs; + } + + int64_t bufferedDurationUs = 0ll; + status_t finalResult = NOT_ENOUGH_DATA; if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) { sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); - int64_t bufferedDurationUs = + bufferedDurationUs = packetSource->getBufferedDurationUs(&finalResult); - - downloadMore = (bufferedDurationUs < kMinBufferedDurationUs); finalResult = OK; } else { - bool first = true; - int64_t minBufferedDurationUs = 0ll; - + // Use max stream duration to prevent us from waiting on a non-existent stream; + // when we cannot make out from the manifest what streams are included in a playlist + // we might assume extra streams. for (size_t i = 0; i < mPacketSources.size(); ++i) { if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { continue; } - int64_t bufferedDurationUs = + int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - - if (first || bufferedDurationUs < minBufferedDurationUs) { - minBufferedDurationUs = bufferedDurationUs; - first = false; + ALOGV("buffered %lld for stream %d", + bufferedStreamDurationUs, mPacketSources.keyAt(i)); + if (bufferedStreamDurationUs > bufferedDurationUs) { + bufferedDurationUs = bufferedStreamDurationUs; } } + } + downloadMore = (bufferedDurationUs < durationToBufferUs); + + // signal start if buffered up at least the target size + if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) { + mPrepared = true; - downloadMore = - !first && (minBufferedDurationUs < kMinBufferedDurationUs); + ALOGV("prepared, buffered=%lld > %lld", + bufferedDurationUs, targetDurationUs); + sp<AMessage> msg = mNotify->dup(); + msg->setInt32("what", kWhatTemporarilyDoneFetching); + msg->post(); } if (finalResult == OK && downloadMore) { - onDownloadNext(); + ALOGV("monitoring, buffered=%lld < %lld", + bufferedDurationUs, durationToBufferUs); + // delay the next download slightly; hopefully this gives other concurrent fetchers + // a better chance to run. + // onDownloadNext(); + sp<AMessage> msg = new AMessage(kWhatDownloadNext, id()); + msg->setInt32("generation", mMonitorQueueGeneration); + msg->post(1000l); } else { // Nothing to do yet, try again in a second. @@ -499,15 +664,17 @@ void PlaylistFetcher::onMonitorQueue() { msg->setInt32("what", kWhatTemporarilyDoneFetching); msg->post(); - postMonitorQueue(1000000ll); + int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2; + ALOGV("pausing for %lld, buffered=%lld > %lld", + delayUs, bufferedDurationUs, durationToBufferUs); + // :TRICKY: need to enforce minimum delay because the delay to + // refresh the playlist will become 0 + postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0); } } -void PlaylistFetcher::onDownloadNext() { - int64_t nowUs = ALooper::GetNowUs(); - - if (mLastPlaylistFetchTimeUs < 0ll - || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) { +status_t PlaylistFetcher::refreshPlaylist() { + if (delayUsToRefreshPlaylist() <= 0) { bool unchanged; sp<M3UParser> playlist = mSession->fetchPlaylist( mURI.c_str(), mPlaylistHash, &unchanged); @@ -523,7 +690,7 @@ void PlaylistFetcher::onDownloadNext() { } else { ALOGE("failed to load playlist at url '%s'", mURI.c_str()); notifyError(ERROR_IO); - return; + return ERROR_IO; } } else { mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; @@ -536,6 +703,18 @@ void PlaylistFetcher::onDownloadNext() { mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); } + return OK; +} + +// static +bool PlaylistFetcher::bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer) { + return buffer->size() > 0 && buffer->data()[0] == 0x47; +} + +void PlaylistFetcher::onDownloadNext() { + if (refreshPlaylist() != OK) { + return; + } int32_t firstSeqNumberInPlaylist; if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( @@ -549,17 +728,29 @@ void PlaylistFetcher::onDownloadNext() { const int32_t lastSeqNumberInPlaylist = firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; + if (mStartup && mSeqNumber >= 0 + && (mSeqNumber < firstSeqNumberInPlaylist || mSeqNumber > lastSeqNumberInPlaylist)) { + // in case we guessed wrong during reconfiguration, try fetching the latest content. + mSeqNumber = lastSeqNumberInPlaylist; + } + if (mSeqNumber < 0) { CHECK_GE(mStartTimeUs, 0ll); if (mPlaylist->isComplete() || mPlaylist->isEvent()) { mSeqNumber = getSeqNumberForTime(mStartTimeUs); + ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)", + mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); } else { // If this is a live session, start 3 segments from the end. mSeqNumber = lastSeqNumberInPlaylist - 3; if (mSeqNumber < firstSeqNumberInPlaylist) { mSeqNumber = firstSeqNumberInPlaylist; } + ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)", + mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); } mStartTimeUs = -1ll; @@ -571,16 +762,34 @@ void PlaylistFetcher::onDownloadNext() { ++mNumRetries; if (mSeqNumber > lastSeqNumberInPlaylist) { - mLastPlaylistFetchTimeUs = -1; - postMonitorQueue(3000000ll); + // refresh in increasing fraction (1/2, 1/3, ...) of the + // playlist's target duration or 3 seconds, whichever is less + int32_t targetDurationSecs; + CHECK(mPlaylist->meta()->findInt32( + "target-duration", &targetDurationSecs)); + int64_t delayUs = mPlaylist->size() * targetDurationSecs * + 1000000ll / (1 + mNumRetries); + if (delayUs > kMaxMonitorDelayUs) { + delayUs = kMaxMonitorDelayUs; + } + ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)", + mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist, delayUs, mNumRetries); + postMonitorQueue(delayUs); return; } // we've missed the boat, let's start from the lowest sequence // number available and signal a discontinuity. - ALOGI("We've missed the boat, restarting playback."); - mSeqNumber = lastSeqNumberInPlaylist; + ALOGI("We've missed the boat, restarting playback." + " mStartup=%d, was looking for %d in %d-%d", + mStartup, mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); + mSeqNumber = lastSeqNumberInPlaylist - 3; + if (mSeqNumber < firstSeqNumberInPlaylist) { + mSeqNumber = firstSeqNumberInPlaylist; + } explicitDiscontinuity = true; // fall through @@ -621,50 +830,160 @@ void PlaylistFetcher::onDownloadNext() { ALOGV("fetching '%s'", uri.c_str()); - sp<ABuffer> buffer; - status_t err = mSession->fetchFile( - uri.c_str(), &buffer, range_offset, range_length); - - if (err != OK) { - ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); - notifyError(err); - return; + sp<DataSource> source; + sp<ABuffer> buffer, tsBuffer; + // decrypt a junk buffer to prefetch key; since a session uses only one http connection, + // this avoids interleaved connections to the key and segment file. + { + sp<ABuffer> junk = new ABuffer(16); + junk->setRange(0, 16); + status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, junk, + true /* first */); + if (err != OK) { + notifyError(err); + return; + } } - CHECK(buffer != NULL); + // block-wise download + ssize_t bytesRead; + do { + bytesRead = mSession->fetchFile( + uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source); + + if (bytesRead < 0) { + status_t err = bytesRead; + ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); + notifyError(err); + return; + } - err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); + CHECK(buffer != NULL); - if (err != OK) { - ALOGE("decryptBuffer failed w/ error %d", err); + size_t size = buffer->size(); + // Set decryption range. + buffer->setRange(size - bytesRead, bytesRead); + status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer, + buffer->offset() == 0 /* first */); + // Unset decryption range. + buffer->setRange(0, size); - notifyError(err); - return; - } + if (err != OK) { + ALOGE("decryptBuffer failed w/ error %d", err); - if (mStartup || seekDiscontinuity || explicitDiscontinuity) { - // Signal discontinuity. + notifyError(err); + return; + } - if (mPlaylist->isComplete() || mPlaylist->isEvent()) { - // If this was a live event this made no sense since - // we don't have access to all the segment before the current - // one. - mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber); + if (mStartup || seekDiscontinuity || explicitDiscontinuity) { + // Signal discontinuity. + + if (mPlaylist->isComplete() || mPlaylist->isEvent()) { + // If this was a live event this made no sense since + // we don't have access to all the segment before the current + // one. + mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber); + } + + if (seekDiscontinuity || explicitDiscontinuity) { + ALOGI("queueing discontinuity (seek=%d, explicit=%d)", + seekDiscontinuity, explicitDiscontinuity); + + queueDiscontinuity( + explicitDiscontinuity + ? ATSParser::DISCONTINUITY_FORMATCHANGE + : ATSParser::DISCONTINUITY_SEEK, + NULL /* extra */); + } + } + + err = OK; + if (bufferStartsWithTsSyncByte(buffer)) { + // Incremental extraction is only supported for MPEG2 transport streams. + if (tsBuffer == NULL) { + tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); + tsBuffer->setRange(0, 0); + } else if (tsBuffer->capacity() != buffer->capacity()) { + size_t tsOff = tsBuffer->offset(), tsSize = tsBuffer->size(); + tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); + tsBuffer->setRange(tsOff, tsSize); + } + tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead); + + err = extractAndQueueAccessUnitsFromTs(tsBuffer); + } + + if (err == -EAGAIN) { + // bad starting sequence number hint + postMonitorQueue(); + return; } - if (seekDiscontinuity || explicitDiscontinuity) { - ALOGI("queueing discontinuity (seek=%d, explicit=%d)", - seekDiscontinuity, explicitDiscontinuity); + if (err == ERROR_OUT_OF_RANGE) { + // reached stopping point + stopAsync(/* selfTriggered = */ true); + return; + } - queueDiscontinuity( - explicitDiscontinuity - ? ATSParser::DISCONTINUITY_FORMATCHANGE - : ATSParser::DISCONTINUITY_SEEK, - NULL /* extra */); + if (err != OK) { + notifyError(err); + return; } + + mStartup = false; + } while (bytesRead != 0); + + if (bufferStartsWithTsSyncByte(buffer)) { + // If we still don't see a stream after fetching a full ts segment mark it as + // nonexistent. + const size_t kNumTypes = ATSParser::NUM_SOURCE_TYPES; + ATSParser::SourceType srcTypes[kNumTypes] = + { ATSParser::VIDEO, ATSParser::AUDIO }; + LiveSession::StreamType streamTypes[kNumTypes] = + { LiveSession::STREAMTYPE_VIDEO, LiveSession::STREAMTYPE_AUDIO }; + + for (size_t i = 0; i < kNumTypes; i++) { + ATSParser::SourceType srcType = srcTypes[i]; + LiveSession::StreamType streamType = streamTypes[i]; + + sp<AnotherPacketSource> source = + static_cast<AnotherPacketSource *>( + mTSParser->getSource(srcType).get()); + + if (source == NULL) { + ALOGW("MPEG2 Transport stream does not contain %s data.", + srcType == ATSParser::VIDEO ? "video" : "audio"); + + mStreamTypeMask &= ~streamType; + mPacketSources.removeItem(streamType); + } + } + + } + + if (checkDecryptPadding(buffer) != OK) { + ALOGE("Incorrect padding bytes after decryption."); + notifyError(ERROR_MALFORMED); + return; } - err = extractAndQueueAccessUnits(buffer, itemMeta); + status_t err = OK; + if (tsBuffer != NULL) { + AString method; + CHECK(buffer->meta()->findString("cipher-method", &method)); + if ((tsBuffer->size() > 0 && method == "NONE") + || tsBuffer->size() > 16) { + ALOGE("MPEG2 transport stream is not an even multiple of 188 " + "bytes in length."); + notifyError(ERROR_MALFORMED); + return; + } + } + + // bulk extract non-ts files + if (tsBuffer == NULL) { + err = extractAndQueueAccessUnits(buffer, itemMeta); + } if (err != OK) { notifyError(err); @@ -674,8 +993,6 @@ void PlaylistFetcher::onDownloadNext() { ++mSeqNumber; postMonitorQueue(); - - mStartup = false; } int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { @@ -710,95 +1027,163 @@ int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { return firstSeqNumberInPlaylist + index; } -status_t PlaylistFetcher::extractAndQueueAccessUnits( - const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) { - if (buffer->size() > 0 && buffer->data()[0] == 0x47) { - // Let's assume this is an MPEG2 transport stream. +status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer) { + if (mTSParser == NULL) { + // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. + mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); + } - if ((buffer->size() % 188) != 0) { - ALOGE("MPEG2 transport stream is not an even multiple of 188 " - "bytes in length."); - return ERROR_MALFORMED; - } + if (mNextPTSTimeUs >= 0ll) { + sp<AMessage> extra = new AMessage; + // Since we are using absolute timestamps, signal an offset of 0 to prevent + // ATSParser from skewing the timestamps of access units. + extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0); - if (mTSParser == NULL) { - mTSParser = new ATSParser; - } + mTSParser->signalDiscontinuity( + ATSParser::DISCONTINUITY_SEEK, extra); - if (mNextPTSTimeUs >= 0ll) { - sp<AMessage> extra = new AMessage; - extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs); + mNextPTSTimeUs = -1ll; + } - mTSParser->signalDiscontinuity( - ATSParser::DISCONTINUITY_SEEK, extra); + size_t offset = 0; + while (offset + 188 <= buffer->size()) { + status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); - mNextPTSTimeUs = -1ll; + if (err != OK) { + return err; } - size_t offset = 0; - while (offset < buffer->size()) { - status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); + offset += 188; + } + // setRange to indicate consumed bytes. + buffer->setRange(buffer->offset() + offset, buffer->size() - offset); + + status_t err = OK; + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + + const char *key; + ATSParser::SourceType type; + const LiveSession::StreamType stream = mPacketSources.keyAt(i); + switch (stream) { + case LiveSession::STREAMTYPE_VIDEO: + type = ATSParser::VIDEO; + key = "timeUsVideo"; + break; + + case LiveSession::STREAMTYPE_AUDIO: + type = ATSParser::AUDIO; + key = "timeUsAudio"; + break; - if (err != OK) { - return err; + case LiveSession::STREAMTYPE_SUBTITLES: + { + ALOGE("MPEG2 Transport streams do not contain subtitles."); + return ERROR_MALFORMED; + break; } - offset += 188; + default: + TRESPASS(); } - for (size_t i = mPacketSources.size(); i-- > 0;) { - sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + sp<AnotherPacketSource> source = + static_cast<AnotherPacketSource *>( + mTSParser->getSource(type).get()); - ATSParser::SourceType type; - switch (mPacketSources.keyAt(i)) { - case LiveSession::STREAMTYPE_VIDEO: - type = ATSParser::VIDEO; - break; + if (source == NULL) { + continue; + } - case LiveSession::STREAMTYPE_AUDIO: - type = ATSParser::AUDIO; + int64_t timeUs; + sp<ABuffer> accessUnit; + status_t finalResult; + while (source->hasBufferAvailable(&finalResult) + && source->dequeueAccessUnit(&accessUnit) == OK) { + + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + if (mMinStartTimeUs > 0) { + if (timeUs < mMinStartTimeUs) { + // TODO untested path + // try a later ts + int32_t targetDuration; + mPlaylist->meta()->findInt32("target-duration", &targetDuration); + int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration; + if (incr == 0) { + // increment mSeqNumber by at least one + incr = 1; + } + mSeqNumber += incr; + err = -EAGAIN; break; + } else { + int64_t startTimeUs; + if (mStartTimeUsNotify != NULL + && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) { + mStartTimeUsNotify->setInt64(key, timeUs); + + uint32_t streamMask = 0; + mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask); + streamMask |= mPacketSources.keyAt(i); + mStartTimeUsNotify->setInt32("streamMask", streamMask); + + if (streamMask == mStreamTypeMask) { + mStartTimeUsNotify->post(); + mStartTimeUsNotify.clear(); + } + } + } + } - case LiveSession::STREAMTYPE_SUBTITLES: - { - ALOGE("MPEG2 Transport streams do not contain subtitles."); - return ERROR_MALFORMED; + if (mStopParams != NULL) { + // Queue discontinuity in original stream. + int64_t stopTimeUs; + if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) { + packetSource->queueAccessUnit(mSession->createFormatChangeBuffer()); + mStreamTypeMask &= ~stream; + mPacketSources.removeItemsAt(i); break; } - - default: - TRESPASS(); } - sp<AnotherPacketSource> source = - static_cast<AnotherPacketSource *>( - mTSParser->getSource(type).get()); - - if (source == NULL) { - ALOGW("MPEG2 Transport stream does not contain %s data.", - type == ATSParser::VIDEO ? "video" : "audio"); + // Note that we do NOT dequeue any discontinuities except for format change. - mStreamTypeMask &= ~mPacketSources.keyAt(i); - mPacketSources.removeItemsAt(i); - continue; + // for simplicity, store a reference to the format in each unit + sp<MetaData> format = source->getFormat(); + if (format != NULL) { + accessUnit->meta()->setObject("format", format); } - sp<ABuffer> accessUnit; - status_t finalResult; - while (source->hasBufferAvailable(&finalResult) - && source->dequeueAccessUnit(&accessUnit) == OK) { - // Note that we do NOT dequeue any discontinuities. + // Stash the sequence number so we can hint future playlist where to start at. + accessUnit->meta()->setInt32("seq", mSeqNumber); + packetSource->queueAccessUnit(accessUnit); + } - packetSource->queueAccessUnit(accessUnit); - } + if (err != OK) { + break; + } + } - if (packetSource->getFormat() == NULL) { - packetSource->setFormat(source->getFormat()); - } + if (err != OK) { + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + packetSource->clear(); } + return err; + } - return OK; - } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { + if (!mStreamTypeMask) { + // Signal gap is filled between original and new stream. + ALOGV("ERROR OUT OF RANGE"); + return ERROR_OUT_OF_RANGE; + } + + return OK; +} + +status_t PlaylistFetcher::extractAndQueueAccessUnits( + const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) { + if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { ALOGE("This stream only contains subtitles."); return ERROR_MALFORMED; @@ -811,6 +1196,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( CHECK(itemMeta->findInt64("durationUs", &durationUs)); buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber)); buffer->meta()->setInt64("durationUs", durationUs); + buffer->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(buffer); return OK; @@ -936,6 +1322,18 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( | (adtsHeader[4] << 3) | (adtsHeader[5] >> 5); + if (aac_frame_length == 0) { + const uint8_t *id3Header = adtsHeader; + if (!memcmp(id3Header, "ID3", 3)) { + ID3 id3(id3Header, buffer->size() - offset, true); + if (id3.isValid()) { + offset += id3.rawSize(); + continue; + }; + } + return ERROR_MALFORMED; + } + CHECK_LE(offset + aac_frame_length, buffer->size()); sp<ABuffer> unit = new ABuffer(aac_frame_length); @@ -947,6 +1345,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( // Each AAC frame encodes 1024 samples. numSamples += 1024; + unit->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(unit); offset += aac_frame_length; @@ -974,4 +1373,33 @@ void PlaylistFetcher::updateDuration() { msg->post(); } +int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) { + int64_t durationUs, threshold; + if (msg->findInt64("durationUs", &durationUs)) { + return kNumSkipFrames * durationUs; + } + + sp<RefBase> obj; + msg->findObject("format", &obj); + MetaData *format = static_cast<MetaData *>(obj.get()); + + const char *mime; + CHECK(format->findCString(kKeyMIMEType, &mime)); + bool audio = !strncasecmp(mime, "audio/", 6); + if (audio) { + // Assumes 1000 samples per frame. + int32_t sampleRate; + CHECK(format->findInt32(kKeySampleRate, &sampleRate)); + return kNumSkipFrames /* frames */ * 1000 /* samples */ + * (1000000 / sampleRate) /* sample duration (us) */; + } else { + int32_t frameRate; + if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) { + return kNumSkipFrames * (1000000 / frameRate); + } + } + + return 500000ll; +} + } // namespace android diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 1648e02..7e21523 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -43,6 +43,7 @@ struct PlaylistFetcher : public AHandler { kWhatTemporarilyDoneFetching, kWhatPrepared, kWhatPreparationFailed, + kWhatStartedAt, }; PlaylistFetcher( @@ -56,11 +57,15 @@ struct PlaylistFetcher : public AHandler { const sp<AnotherPacketSource> &audioSource, const sp<AnotherPacketSource> &videoSource, const sp<AnotherPacketSource> &subtitleSource, - int64_t startTimeUs = -1ll); + int64_t startTimeUs = -1ll, + int64_t minStartTimeUs = 0ll /* start after this timestamp */, + int32_t startSeqNumberHint = -1 /* try starting at this sequence number */); void pauseAsync(); - void stopAsync(); + void stopAsync(bool selfTriggered = false); + + void resumeUntilAsync(const sp<AMessage> ¶ms); protected: virtual ~PlaylistFetcher(); @@ -76,16 +81,28 @@ private: kWhatPause = 'paus', kWhatStop = 'stop', kWhatMonitorQueue = 'moni', + kWhatResumeUntil = 'rsme', + kWhatDownloadNext = 'dlnx', }; static const int64_t kMinBufferedDurationUs; + static const int64_t kMaxMonitorDelayUs; + static const int32_t kDownloadBlockSize; + static const int32_t kNumSkipFrames; + + static bool bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer); + // notifications to mSession sp<AMessage> mNotify; + sp<AMessage> mStartTimeUsNotify; + sp<LiveSession> mSession; AString mURI; uint32_t mStreamTypeMask; int64_t mStartTimeUs; + int64_t mMinStartTimeUs; // start fetching no earlier than this value + sp<AMessage> mStopParams; // message containing the latest timestamps we should fetch. KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> > mPacketSources; @@ -97,6 +114,7 @@ private: int32_t mSeqNumber; int32_t mNumRetries; bool mStartup; + bool mPrepared; int64_t mNextPTSTimeUs; int32_t mMonitorQueueGeneration; @@ -117,13 +135,29 @@ private: uint64_t mFirstPTS; int64_t mAbsoluteTimeAnchorUs; + // Stores the initialization vector to decrypt the next block of cipher text, which can + // either be derived from the sequence number, read from the manifest, or copied from + // the last block of cipher text (cipher-block chaining). + unsigned char mAESInitVec[16]; + + // Set first to true if decrypting the first segment of a playlist segment. When + // first is true, reset the initialization vector based on the available + // information in the manifest; otherwise, use the initialization vector as + // updated by the last call to AES_cbc_encrypt. + // + // For the input to decrypt correctly, decryptBuffer must be called on + // consecutive byte ranges on block boundaries, e.g. 0..15, 16..47, 48..63, + // and so on. status_t decryptBuffer( - size_t playlistIndex, const sp<ABuffer> &buffer); + size_t playlistIndex, const sp<ABuffer> &buffer, + bool first = true); + status_t checkDecryptPadding(const sp<ABuffer> &buffer); - void postMonitorQueue(int64_t delayUs = 0); + void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0); void cancelMonitorQueue(); - bool timeToRefreshPlaylist(int64_t nowUs) const; + int64_t delayUsToRefreshPlaylist() const; + status_t refreshPlaylist(); // Returns the media time in us of the segment specified by seqNumber. // This is computed by summing the durations of all segments before it. @@ -131,10 +165,15 @@ private: status_t onStart(const sp<AMessage> &msg); void onPause(); - void onStop(); + void onStop(const sp<AMessage> &msg); void onMonitorQueue(); void onDownloadNext(); + // Resume a fetcher to continue until the stopping point stored in msg. + status_t onResumeUntil(const sp<AMessage> &msg); + + status_t extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer); + status_t extractAndQueueAccessUnits( const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta); @@ -147,6 +186,10 @@ private: void updateDuration(); + // Before resuming a fetcher in onResume, check the remaining duration is longer than that + // returned by resumeThreshold. + int64_t resumeThreshold(const sp<AMessage> &msg); + DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher); }; diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp index 3647583..0e4dd2b 100644 --- a/media/libstagefright/matroska/MatroskaExtractor.cpp +++ b/media/libstagefright/matroska/MatroskaExtractor.cpp @@ -718,41 +718,61 @@ bool MatroskaExtractor::isLiveStreaming() const { return mIsLiveStreaming; } +static int bytesForSize(size_t size) { + // use at most 28 bits (4 times 7) + CHECK(size <= 0xfffffff); + + if (size > 0x1fffff) { + return 4; + } else if (size > 0x3fff) { + return 3; + } else if (size > 0x7f) { + return 2; + } + return 1; +} + +static void storeSize(uint8_t *data, size_t &idx, size_t size) { + int numBytes = bytesForSize(size); + idx += numBytes; + + data += idx; + size_t next = 0; + while (numBytes--) { + *--data = (size & 0x7f) | next; + size >>= 7; + next = 0x80; + } +} + static void addESDSFromCodecPrivate( const sp<MetaData> &meta, bool isAudio, const void *priv, size_t privSize) { - static const uint8_t kStaticESDS[] = { - 0x03, 22, - 0x00, 0x00, // ES_ID - 0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag - - 0x04, 17, - 0x40, // ObjectTypeIndication - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - - 0x05, - // CodecSpecificInfo (with size prefix) follows - }; - // Make sure all sizes can be coded in a single byte. - CHECK(privSize + 22 - 2 < 128); - size_t esdsSize = sizeof(kStaticESDS) + privSize + 1; + int privSizeBytesRequired = bytesForSize(privSize); + int esdsSize2 = 14 + privSizeBytesRequired + privSize; + int esdsSize2BytesRequired = bytesForSize(esdsSize2); + int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2; + int esdsSize1BytesRequired = bytesForSize(esdsSize1); + size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1; uint8_t *esds = new uint8_t[esdsSize]; - memcpy(esds, kStaticESDS, sizeof(kStaticESDS)); - uint8_t *ptr = esds + sizeof(kStaticESDS); - *ptr++ = privSize; - memcpy(ptr, priv, privSize); - - // Increment by codecPrivateSize less 2 bytes that are accounted for - // already in lengths of 22/17 - esds[1] += privSize - 2; - esds[6] += privSize - 2; - - // Set ObjectTypeIndication. - esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3 - : 0x20; // Visual ISO/IEC 14496-2 + + size_t idx = 0; + esds[idx++] = 0x03; + storeSize(esds, idx, esdsSize1); + esds[idx++] = 0x00; // ES_ID + esds[idx++] = 0x00; // ES_ID + esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag + esds[idx++] = 0x04; + storeSize(esds, idx, esdsSize2); + esds[idx++] = isAudio ? 0x40 // Audio ISO/IEC 14496-3 + : 0x20; // Visual ISO/IEC 14496-2 + for (int i = 0; i < 12; i++) { + esds[idx++] = 0x00; + } + esds[idx++] = 0x05; + storeSize(esds, idx, privSize); + memcpy(esds + idx, priv, privSize); meta->setData(kKeyESDS, 0, esds, esdsSize); diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h index a10edc9..8a80069 100644 --- a/media/libstagefright/mpeg2ts/ATSParser.h +++ b/media/libstagefright/mpeg2ts/ATSParser.h @@ -71,8 +71,9 @@ struct ATSParser : public RefBase { void signalEOS(status_t finalResult); enum SourceType { - VIDEO, - AUDIO + VIDEO = 0, + AUDIO = 1, + NUM_SOURCE_TYPES = 2 }; sp<MediaSource> getSource(SourceType type); diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp index 13f073a..6dfaa94 100644 --- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp +++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp @@ -34,7 +34,8 @@ AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta) : mIsAudio(false), mFormat(NULL), mLastQueuedTimeUs(0), - mEOSResult(OK) { + mEOSResult(OK), + mLatestEnqueuedMeta(NULL) { setFormat(meta); } @@ -70,7 +71,27 @@ status_t AnotherPacketSource::stop() { } sp<MetaData> AnotherPacketSource::getFormat() { - return mFormat; + Mutex::Autolock autoLock(mLock); + if (mFormat != NULL) { + return mFormat; + } + + List<sp<ABuffer> >::iterator it = mBuffers.begin(); + while (it != mBuffers.end()) { + sp<ABuffer> buffer = *it; + int32_t discontinuity; + if (buffer->meta()->findInt32("discontinuity", &discontinuity)) { + break; + } + + sp<RefBase> object; + if (buffer->meta()->findObject("format", &object)) { + return static_cast<MetaData*>(object.get()); + } + + ++it; + } + return NULL; } status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) { @@ -94,6 +115,11 @@ status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) { return INFO_DISCONTINUITY; } + sp<RefBase> object; + if ((*buffer)->meta()->findObject("format", &object)) { + mFormat = static_cast<MetaData*>(object.get()); + } + return OK; } @@ -120,17 +146,22 @@ status_t AnotherPacketSource::read( } return INFO_DISCONTINUITY; - } else { - int64_t timeUs; - CHECK(buffer->meta()->findInt64("timeUs", &timeUs)); + } + + sp<RefBase> object; + if (buffer->meta()->findObject("format", &object)) { + mFormat = static_cast<MetaData*>(object.get()); + } - MediaBuffer *mediaBuffer = new MediaBuffer(buffer); + int64_t timeUs; + CHECK(buffer->meta()->findInt64("timeUs", &timeUs)); - mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs); + MediaBuffer *mediaBuffer = new MediaBuffer(buffer); - *out = mediaBuffer; - return OK; - } + mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs); + + *out = mediaBuffer; + return OK; } return mEOSResult; @@ -152,12 +183,24 @@ void AnotherPacketSource::queueAccessUnit(const sp<ABuffer> &buffer) { return; } - CHECK(buffer->meta()->findInt64("timeUs", &mLastQueuedTimeUs)); + int64_t lastQueuedTimeUs; + CHECK(buffer->meta()->findInt64("timeUs", &lastQueuedTimeUs)); + mLastQueuedTimeUs = lastQueuedTimeUs; ALOGV("queueAccessUnit timeUs=%lld us (%.2f secs)", mLastQueuedTimeUs, mLastQueuedTimeUs / 1E6); Mutex::Autolock autoLock(mLock); mBuffers.push_back(buffer); mCondition.signal(); + + if (!mLatestEnqueuedMeta.get()) { + mLatestEnqueuedMeta = buffer->meta(); + } else { + int64_t latestTimeUs = 0; + CHECK(mLatestEnqueuedMeta->findInt64("timeUs", &latestTimeUs)); + if (lastQueuedTimeUs > latestTimeUs) { + mLatestEnqueuedMeta = buffer->meta(); + } + } } void AnotherPacketSource::clear() { @@ -167,6 +210,7 @@ void AnotherPacketSource::clear() { mEOSResult = OK; mFormat = NULL; + mLatestEnqueuedMeta = NULL; } void AnotherPacketSource::queueDiscontinuity( @@ -191,6 +235,7 @@ void AnotherPacketSource::queueDiscontinuity( mEOSResult = OK; mLastQueuedTimeUs = 0; + mLatestEnqueuedMeta = NULL; sp<ABuffer> buffer = new ABuffer(0); buffer->meta()->setInt32("discontinuity", static_cast<int32_t>(type)); @@ -278,4 +323,9 @@ bool AnotherPacketSource::isFinished(int64_t duration) const { return (mEOSResult != OK); } +sp<AMessage> AnotherPacketSource::getLatestMeta() { + Mutex::Autolock autoLock(mLock); + return mLatestEnqueuedMeta; +} + } // namespace android diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h index e16cf78..9b193a2 100644 --- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h +++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h @@ -62,6 +62,8 @@ struct AnotherPacketSource : public MediaSource { bool isFinished(int64_t duration) const; + sp<AMessage> getLatestMeta(); + protected: virtual ~AnotherPacketSource(); @@ -74,6 +76,7 @@ private: int64_t mLastQueuedTimeUs; List<sp<ABuffer> > mBuffers; status_t mEOSResult; + sp<AMessage> mLatestEnqueuedMeta; bool wasFormatChange(int32_t discontinuityType) const; diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp index 4d3930b..3fe9c23 100644 --- a/media/libstagefright/omx/GraphicBufferSource.cpp +++ b/media/libstagefright/omx/GraphicBufferSource.cpp @@ -44,7 +44,11 @@ GraphicBufferSource::GraphicBufferSource(OMXNodeInstance* nodeInstance, mEndOfStream(false), mEndOfStreamSent(false), mRepeatAfterUs(-1ll), + mMaxTimestampGapUs(-1ll), + mPrevOriginalTimeUs(-1ll), + mPrevModifiedTimeUs(-1ll), mRepeatLastFrameGeneration(0), + mRepeatLastFrameTimestamp(-1ll), mLatestSubmittedBufferId(-1), mLatestSubmittedBufferFrameNum(0), mLatestSubmittedBufferUseCount(0), @@ -301,6 +305,32 @@ void GraphicBufferSource::codecBufferEmptied(OMX_BUFFERHEADERTYPE* header) { return; } +void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) { + Mutex::Autolock autoLock(mMutex); + + if (mMaxTimestampGapUs > 0ll + && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) { + ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp); + if (index >= 0) { + ALOGV("OUT timestamp: %lld -> %lld", + header->nTimeStamp, mOriginalTimeUs[index]); + header->nTimeStamp = mOriginalTimeUs[index]; + mOriginalTimeUs.removeItemsAt(index); + } else { + // giving up the effort as encoder doesn't appear to preserve pts + ALOGW("giving up limiting timestamp gap (pts = %lld)", + header->nTimeStamp); + mMaxTimestampGapUs = -1ll; + } + if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) { + // something terribly wrong must have happened, giving up... + ALOGE("mOriginalTimeUs has too many entries (%d)", + mOriginalTimeUs.size()); + mMaxTimestampGapUs = -1ll; + } + } +} + void GraphicBufferSource::suspend(bool suspend) { Mutex::Autolock autoLock(mMutex); @@ -433,6 +463,7 @@ bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() { BufferQueue::BufferItem item; item.mBuf = mLatestSubmittedBufferId; item.mFrameNumber = mLatestSubmittedBufferFrameNum; + item.mTimestamp = mRepeatLastFrameTimestamp; status_t err = submitBuffer_l(item, cbi); @@ -442,6 +473,20 @@ bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() { ++mLatestSubmittedBufferUseCount; + /* repeat last frame up to kRepeatLastFrameCount times. + * in case of static scene, a single repeat might not get rid of encoder + * ghosting completely, refresh a couple more times to get better quality + */ + if (--mRepeatLastFrameCount > 0) { + mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000; + + if (mReflector != NULL) { + sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id()); + msg->setInt32("generation", ++mRepeatLastFrameGeneration); + msg->post(mRepeatAfterUs); + } + } + return true; } @@ -462,8 +507,11 @@ void GraphicBufferSource::setLatestSubmittedBuffer_l( mLatestSubmittedBufferId = item.mBuf; mLatestSubmittedBufferFrameNum = item.mFrameNumber; + mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000; + mLatestSubmittedBufferUseCount = 1; mRepeatBufferDeferred = false; + mRepeatLastFrameCount = kRepeatLastFrameCount; if (mReflector != NULL) { sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id()); @@ -499,9 +547,48 @@ status_t GraphicBufferSource::signalEndOfInputStream() { return OK; } +int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) { + int64_t timeUs = item.mTimestamp / 1000; + + if (mMaxTimestampGapUs > 0ll) { + /* Cap timestamp gap between adjacent frames to specified max + * + * In the scenario of cast mirroring, encoding could be suspended for + * prolonged periods. Limiting the pts gap to workaround the problem + * where encoder's rate control logic produces huge frames after a + * long period of suspension. + */ + + int64_t originalTimeUs = timeUs; + if (mPrevOriginalTimeUs >= 0ll) { + if (originalTimeUs < mPrevOriginalTimeUs) { + // Drop the frame if it's going backward in time. Bad timestamp + // could disrupt encoder's rate control completely. + ALOGW("Dropping frame that's going backward in time"); + return -1; + } + int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs; + timeUs = (timestampGapUs < mMaxTimestampGapUs ? + timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs; + } + mPrevOriginalTimeUs = originalTimeUs; + mPrevModifiedTimeUs = timeUs; + mOriginalTimeUs.add(timeUs, originalTimeUs); + ALOGV("IN timestamp: %lld -> %lld", originalTimeUs, timeUs); + } + + return timeUs; +} + status_t GraphicBufferSource::submitBuffer_l( const BufferQueue::BufferItem &item, int cbi) { ALOGV("submitBuffer_l cbi=%d", cbi); + + int64_t timeUs = getTimestamp(item); + if (timeUs < 0ll) { + return UNKNOWN_ERROR; + } + CodecBuffer& codecBuffer(mCodecBuffers.editItemAt(cbi)); codecBuffer.mGraphicBuffer = mBufferSlot[item.mBuf]; codecBuffer.mBuf = item.mBuf; @@ -517,7 +604,7 @@ status_t GraphicBufferSource::submitBuffer_l( status_t err = mNodeInstance->emptyDirectBuffer(header, 0, 4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME, - item.mTimestamp / 1000); + timeUs); if (err != OK) { ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err); codecBuffer.mGraphicBuffer = NULL; @@ -611,6 +698,12 @@ void GraphicBufferSource::onFrameAvailable() { BufferQueue::BufferItem item; status_t err = mBufferQueue->acquireBuffer(&item, 0); if (err == OK) { + // If this is the first time we're seeing this buffer, add it to our + // slot table. + if (item.mGraphicBuffer != NULL) { + ALOGV("fillCodecBuffer_l: setting mBufferSlot %d", item.mBuf); + mBufferSlot[item.mBuf] = item.mGraphicBuffer; + } mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence); } @@ -660,6 +753,17 @@ status_t GraphicBufferSource::setRepeatPreviousFrameDelayUs( return OK; } +status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) { + Mutex::Autolock autoLock(mMutex); + + if (mExecuting || maxGapUs <= 0ll) { + return INVALID_OPERATION; + } + + mMaxTimestampGapUs = maxGapUs; + + return OK; +} void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) { switch (msg->what()) { case kWhatRepeatLastFrame: diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h index 9e5eee6..3b0e454 100644 --- a/media/libstagefright/omx/GraphicBufferSource.h +++ b/media/libstagefright/omx/GraphicBufferSource.h @@ -87,6 +87,10 @@ public: // fill it with a new frame of data; otherwise, just mark it as available. void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header); + // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the + // buffer source will fix timestamp in the header if needed.) + void codecBufferFilled(OMX_BUFFERHEADERTYPE* header); + // This is called after the last input frame has been submitted. We // need to submit an empty buffer with the EOS flag set. If we don't // have a codec buffer ready, we just set the mEndOfStream flag. @@ -105,6 +109,15 @@ public: // state and once this behaviour is specified it cannot be reset. status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs); + // When set, the timestamp fed to the encoder will be modified such that + // the gap between two adjacent frames is capped at maxGapUs. Timestamp + // will be restored to the original when the encoded frame is returned to + // the client. + // This is to solve a problem in certain real-time streaming case, where + // encoder's rate control logic produces huge frames after a long period + // of suspension on input. + status_t setMaxTimestampGapUs(int64_t maxGapUs); + protected: // BufferQueue::ConsumerListener interface, called when a new frame of // data is available. If we're executing and a codec buffer is @@ -165,6 +178,7 @@ private: void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item); bool repeatLatestSubmittedBuffer_l(); + int64_t getTimestamp(const BufferQueue::BufferItem &item); // Lock, covers all member variables. mutable Mutex mMutex; @@ -206,13 +220,22 @@ private: enum { kWhatRepeatLastFrame, }; - + enum { + kRepeatLastFrameCount = 10, + }; int64_t mRepeatAfterUs; + int64_t mMaxTimestampGapUs; + + KeyedVector<int64_t, int64_t> mOriginalTimeUs; + int64_t mPrevOriginalTimeUs; + int64_t mPrevModifiedTimeUs; sp<ALooper> mLooper; sp<AHandlerReflector<GraphicBufferSource> > mReflector; int32_t mRepeatLastFrameGeneration; + int64_t mRepeatLastFrameTimestamp; + int32_t mRepeatLastFrameCount; int mLatestSubmittedBufferId; uint64_t mLatestSubmittedBufferFrameNum; diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp index 1422210..8391290 100644 --- a/media/libstagefright/omx/OMXNodeInstance.cpp +++ b/media/libstagefright/omx/OMXNodeInstance.cpp @@ -849,6 +849,7 @@ status_t OMXNodeInstance::setInternalOption( switch (type) { case IOMX::INTERNAL_OPTION_SUSPEND: case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY: + case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP: { const sp<GraphicBufferSource> &bufferSource = getGraphicBufferSource(); @@ -864,7 +865,8 @@ status_t OMXNodeInstance::setInternalOption( bool suspend = *(bool *)data; bufferSource->suspend(suspend); - } else { + } else if (type == + IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){ if (size != sizeof(int64_t)) { return INVALID_OPERATION; } @@ -872,6 +874,14 @@ status_t OMXNodeInstance::setInternalOption( int64_t delayUs = *(int64_t *)data; return bufferSource->setRepeatPreviousFrameDelayUs(delayUs); + } else { + if (size != sizeof(int64_t)) { + return INVALID_OPERATION; + } + + int64_t maxGapUs = *(int64_t *)data; + + return bufferSource->setMaxTimestampGapUs(maxGapUs); } return OK; @@ -883,6 +893,8 @@ status_t OMXNodeInstance::setInternalOption( } void OMXNodeInstance::onMessage(const omx_message &msg) { + const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource()); + if (msg.type == omx_message::FILL_BUFFER_DONE) { OMX_BUFFERHEADERTYPE *buffer = static_cast<OMX_BUFFERHEADERTYPE *>( @@ -892,10 +904,18 @@ void OMXNodeInstance::onMessage(const omx_message &msg) { static_cast<BufferMeta *>(buffer->pAppPrivate); buffer_meta->CopyFromOMX(buffer); - } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) { - const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource()); if (bufferSource != NULL) { + // fix up the buffer info (especially timestamp) if needed + bufferSource->codecBufferFilled(buffer); + + omx_message newMsg = msg; + newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp; + mObserver->onMessage(newMsg); + return; + } + } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) { + if (bufferSource != NULL) { // This is one of the buffers used exclusively by // GraphicBufferSource. // Don't dispatch a message back to ACodec, since it doesn't diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp index edcc087..50d317a 100644 --- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp +++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp @@ -216,7 +216,7 @@ sp<ABuffer> TSPacketizer::Track::prependADTSHeader( uint8_t *ptr = dup->data(); *ptr++ = 0xff; - *ptr++ = 0xf1; // b11110001, ID=0, layer=0, protection_absent=1 + *ptr++ = 0xf9; // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1 *ptr++ = profile << 6 diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 1da5147..cac785a 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -1324,6 +1324,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId, uid); } + if (track == 0 || track->getCblk() == NULL || track->name() < 0) { lStatus = NO_MEMORY; // track must be cleared from the caller as the caller has the AF lock @@ -1916,7 +1917,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() // otherwise use the HAL / AudioStreamOut directly } else { // Direct output and offload threads - size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); + size_t offset = (mCurrentWriteLength - mBytesRemaining); if (mUseAsyncWrite) { ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); mWriteAckSequence += 2; @@ -1927,7 +1928,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() // FIXME We should have an implementation of timestamps for direct output threads. // They are used e.g for multichannel PCM playback over HDMI. bytesWritten = mOutput->stream->write(mOutput->stream, - mMixBuffer + offset, mBytesRemaining); + (char *)mMixBuffer + offset, mBytesRemaining); if (mUseAsyncWrite && ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { // do not wait for async callback in case of error of full write diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index b59333f..d07113c 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -551,7 +551,7 @@ size_t AudioFlinger::PlaybackThread::Track::framesReleased() const // Don't call for fast tracks; the framesReady() could result in priority inversion bool AudioFlinger::PlaybackThread::Track::isReady() const { - if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { + if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) { return true; } diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp index 141b1b3..9ce7daf 100644 --- a/services/camera/libcameraservice/CameraService.cpp +++ b/services/camera/libcameraservice/CameraService.cpp @@ -655,6 +655,11 @@ status_t CameraService::addListener( const sp<ICameraServiceListener>& listener) { ALOGV("%s: Add listener %p", __FUNCTION__, listener.get()); + if (listener == 0) { + ALOGE("%s: Listener must not be null", __FUNCTION__); + return BAD_VALUE; + } + Mutex::Autolock lock(mServiceLock); Vector<sp<ICameraServiceListener> >::iterator it, end; @@ -683,6 +688,11 @@ status_t CameraService::removeListener( const sp<ICameraServiceListener>& listener) { ALOGV("%s: Remove listener %p", __FUNCTION__, listener.get()); + if (listener == 0) { + ALOGE("%s: Listener must not be null", __FUNCTION__); + return BAD_VALUE; + } + Mutex::Autolock lock(mServiceLock); Vector<sp<ICameraServiceListener> >::iterator it; diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp index ba1e772..af23557 100644 --- a/services/camera/libcameraservice/api1/Camera2Client.cpp +++ b/services/camera/libcameraservice/api1/Camera2Client.cpp @@ -407,12 +407,6 @@ void Camera2Client::disconnect() { l.mParameters.state = Parameters::DISCONNECTED; } - mStreamingProcessor->deletePreviewStream(); - mStreamingProcessor->deleteRecordingStream(); - mJpegProcessor->deleteStream(); - mCallbackProcessor->deleteStream(); - mZslProcessor->deleteStream(); - mStreamingProcessor->requestExit(); mFrameProcessor->requestExit(); mCaptureSequencer->requestExit(); @@ -429,6 +423,14 @@ void Camera2Client::disconnect() { mZslProcessorThread->join(); mCallbackProcessor->join(); + ALOGV("Camera %d: Deleting streams", mCameraId); + + mStreamingProcessor->deletePreviewStream(); + mStreamingProcessor->deleteRecordingStream(); + mJpegProcessor->deleteStream(); + mCallbackProcessor->deleteStream(); + mZslProcessor->deleteStream(); + ALOGV("Camera %d: Disconnecting device", mCameraId); mDevice->disconnect(); @@ -732,6 +734,7 @@ status_t Camera2Client::startPreviewL(Parameters ¶ms, bool restart) { return OK; } params.state = Parameters::STOPPED; + int lastPreviewStreamId = mStreamingProcessor->getPreviewStreamId(); res = mStreamingProcessor->updatePreviewStream(params); if (res != OK) { @@ -740,6 +743,8 @@ status_t Camera2Client::startPreviewL(Parameters ¶ms, bool restart) { return res; } + bool previewStreamChanged = mStreamingProcessor->getPreviewStreamId() != lastPreviewStreamId; + // We could wait to create the JPEG output stream until first actual use // (first takePicture call). However, this would substantially increase the // first capture latency on HAL3 devices, and potentially on some HAL2 @@ -789,6 +794,19 @@ status_t Camera2Client::startPreviewL(Parameters ¶ms, bool restart) { return res; } outputStreams.push(getCallbackStreamId()); + } else if (previewStreamChanged && mCallbackProcessor->getStreamId() != NO_STREAM) { + /** + * Delete the unused callback stream when preview stream is changed and + * preview is not enabled. Don't need stop preview stream as preview is in + * STOPPED state now. + */ + ALOGV("%s: Camera %d: Delete unused preview callback stream.", __FUNCTION__, mCameraId); + res = mCallbackProcessor->deleteStream(); + if (res != OK) { + ALOGE("%s: Camera %d: Unable to delete callback stream %s (%d)", + __FUNCTION__, mCameraId, strerror(-res), res); + return res; + } } if (params.zslMode && !params.recordingHint) { res = updateProcessorStream(mZslProcessor, params); diff --git a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp index 1f4d39b..2de7a2b 100644 --- a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp +++ b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp @@ -200,50 +200,59 @@ status_t JpegProcessor::processNewCapture() { ATRACE_CALL(); status_t res; sp<Camera2Heap> captureHeap; + sp<MemoryBase> captureBuffer; CpuConsumer::LockedBuffer imgBuffer; - res = mCaptureConsumer->lockNextBuffer(&imgBuffer); - if (res != OK) { - if (res != BAD_VALUE) { - ALOGE("%s: Camera %d: Error receiving still image buffer: " - "%s (%d)", __FUNCTION__, - mId, strerror(-res), res); + { + Mutex::Autolock l(mInputMutex); + if (mCaptureStreamId == NO_STREAM) { + ALOGW("%s: Camera %d: No stream is available", __FUNCTION__, mId); + return INVALID_OPERATION; } - return res; - } - ALOGV("%s: Camera %d: Still capture available", __FUNCTION__, - mId); + res = mCaptureConsumer->lockNextBuffer(&imgBuffer); + if (res != OK) { + if (res != BAD_VALUE) { + ALOGE("%s: Camera %d: Error receiving still image buffer: " + "%s (%d)", __FUNCTION__, + mId, strerror(-res), res); + } + return res; + } - if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) { - ALOGE("%s: Camera %d: Unexpected format for still image: " - "%x, expected %x", __FUNCTION__, mId, - imgBuffer.format, - HAL_PIXEL_FORMAT_BLOB); - mCaptureConsumer->unlockBuffer(imgBuffer); - return OK; - } + ALOGV("%s: Camera %d: Still capture available", __FUNCTION__, + mId); - // Find size of JPEG image - size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width); - if (jpegSize == 0) { // failed to find size, default to whole buffer - jpegSize = imgBuffer.width; - } - size_t heapSize = mCaptureHeap->getSize(); - if (jpegSize > heapSize) { - ALOGW("%s: JPEG image is larger than expected, truncating " - "(got %zu, expected at most %zu bytes)", - __FUNCTION__, jpegSize, heapSize); - jpegSize = heapSize; - } + if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) { + ALOGE("%s: Camera %d: Unexpected format for still image: " + "%x, expected %x", __FUNCTION__, mId, + imgBuffer.format, + HAL_PIXEL_FORMAT_BLOB); + mCaptureConsumer->unlockBuffer(imgBuffer); + return OK; + } - // TODO: Optimize this to avoid memcopy - sp<MemoryBase> captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize); - void* captureMemory = mCaptureHeap->getBase(); - memcpy(captureMemory, imgBuffer.data, jpegSize); + // Find size of JPEG image + size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width); + if (jpegSize == 0) { // failed to find size, default to whole buffer + jpegSize = imgBuffer.width; + } + size_t heapSize = mCaptureHeap->getSize(); + if (jpegSize > heapSize) { + ALOGW("%s: JPEG image is larger than expected, truncating " + "(got %zu, expected at most %zu bytes)", + __FUNCTION__, jpegSize, heapSize); + jpegSize = heapSize; + } + + // TODO: Optimize this to avoid memcopy + captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize); + void* captureMemory = mCaptureHeap->getBase(); + memcpy(captureMemory, imgBuffer.data, jpegSize); - mCaptureConsumer->unlockBuffer(imgBuffer); + mCaptureConsumer->unlockBuffer(imgBuffer); + } sp<CaptureSequencer> sequencer = mSequencer.promote(); if (sequencer != 0) { diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp index 0cceb76..07654c0 100644 --- a/services/camera/libcameraservice/api1/client2/Parameters.cpp +++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp @@ -16,7 +16,7 @@ #define LOG_TAG "Camera2-Parameters" #define ATRACE_TAG ATRACE_TAG_CAMERA -//#define LOG_NDEBUG 0 +// #define LOG_NDEBUG 0 #include <utils/Log.h> #include <utils/Trace.h> @@ -92,26 +92,6 @@ status_t Parameters::initialize(const CameraMetadata *info) { staticInfo(ANDROID_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES, 2); if (!availableFpsRanges.count) return NO_INIT; - previewFpsRange[0] = availableFpsRanges.data.i32[0]; - previewFpsRange[1] = availableFpsRanges.data.i32[1]; - - params.set(CameraParameters::KEY_PREVIEW_FPS_RANGE, - String8::format("%d,%d", - previewFpsRange[0] * kFpsToApiScale, - previewFpsRange[1] * kFpsToApiScale)); - - { - String8 supportedPreviewFpsRange; - for (size_t i=0; i < availableFpsRanges.count; i += 2) { - if (i != 0) supportedPreviewFpsRange += ","; - supportedPreviewFpsRange += String8::format("(%d,%d)", - availableFpsRanges.data.i32[i] * kFpsToApiScale, - availableFpsRanges.data.i32[i+1] * kFpsToApiScale); - } - params.set(CameraParameters::KEY_SUPPORTED_PREVIEW_FPS_RANGE, - supportedPreviewFpsRange); - } - previewFormat = HAL_PIXEL_FORMAT_YCrCb_420_SP; params.set(CameraParameters::KEY_PREVIEW_FORMAT, formatEnumToString(previewFormat)); // NV21 @@ -179,6 +159,9 @@ status_t Parameters::initialize(const CameraMetadata *info) { supportedPreviewFormats); } + previewFpsRange[0] = availableFpsRanges.data.i32[0]; + previewFpsRange[1] = availableFpsRanges.data.i32[1]; + // PREVIEW_FRAME_RATE / SUPPORTED_PREVIEW_FRAME_RATES are deprecated, but // still have to do something sane for them @@ -187,6 +170,27 @@ status_t Parameters::initialize(const CameraMetadata *info) { params.set(CameraParameters::KEY_PREVIEW_FRAME_RATE, previewFps); + // PREVIEW_FPS_RANGE + // -- Order matters. Set range after single value to so that a roundtrip + // of setParameters(getParameters()) would keep the FPS range in higher + // order. + params.set(CameraParameters::KEY_PREVIEW_FPS_RANGE, + String8::format("%d,%d", + previewFpsRange[0] * kFpsToApiScale, + previewFpsRange[1] * kFpsToApiScale)); + + { + String8 supportedPreviewFpsRange; + for (size_t i=0; i < availableFpsRanges.count; i += 2) { + if (i != 0) supportedPreviewFpsRange += ","; + supportedPreviewFpsRange += String8::format("(%d,%d)", + availableFpsRanges.data.i32[i] * kFpsToApiScale, + availableFpsRanges.data.i32[i+1] * kFpsToApiScale); + } + params.set(CameraParameters::KEY_SUPPORTED_PREVIEW_FPS_RANGE, + supportedPreviewFpsRange); + } + { SortedVector<int32_t> sortedPreviewFrameRates; @@ -1084,7 +1088,7 @@ camera_metadata_ro_entry_t Parameters::staticInfo(uint32_t tag, status_t Parameters::set(const String8& paramString) { status_t res; - CameraParameters newParams(paramString); + CameraParameters2 newParams(paramString); // TODO: Currently ignoring any changes to supposedly read-only parameters // such as supported preview sizes, etc. Should probably produce an error if @@ -1127,29 +1131,73 @@ status_t Parameters::set(const String8& paramString) { // RECORDING_HINT (always supported) validatedParams.recordingHint = boolFromString( newParams.get(CameraParameters::KEY_RECORDING_HINT) ); - bool recordingHintChanged = validatedParams.recordingHint != recordingHint; - ALOGV_IF(recordingHintChanged, "%s: Recording hint changed to %d", - __FUNCTION__, recordingHintChanged); + IF_ALOGV() { // Avoid unused variable warning + bool recordingHintChanged = + validatedParams.recordingHint != recordingHint; + if (recordingHintChanged) { + ALOGV("%s: Recording hint changed to %d", + __FUNCTION__, validatedParams.recordingHint); + } + } // PREVIEW_FPS_RANGE - bool fpsRangeChanged = false; - int32_t lastSetFpsRange[2]; - params.getPreviewFpsRange(&lastSetFpsRange[0], &lastSetFpsRange[1]); - lastSetFpsRange[0] /= kFpsToApiScale; - lastSetFpsRange[1] /= kFpsToApiScale; + /** + * Use the single FPS value if it was set later than the range. + * Otherwise, use the range value. + */ + bool fpsUseSingleValue; + { + const char *fpsRange, *fpsSingle; + + fpsRange = newParams.get(CameraParameters::KEY_PREVIEW_FRAME_RATE); + fpsSingle = newParams.get(CameraParameters::KEY_PREVIEW_FPS_RANGE); + + /** + * Pick either the range or the single key if only one was set. + * + * If both are set, pick the one that has greater set order. + */ + if (fpsRange == NULL && fpsSingle == NULL) { + ALOGE("%s: FPS was not set. One of %s or %s must be set.", + __FUNCTION__, CameraParameters::KEY_PREVIEW_FRAME_RATE, + CameraParameters::KEY_PREVIEW_FPS_RANGE); + return BAD_VALUE; + } else if (fpsRange == NULL) { + fpsUseSingleValue = true; + ALOGV("%s: FPS range not set, using FPS single value", + __FUNCTION__); + } else if (fpsSingle == NULL) { + fpsUseSingleValue = false; + ALOGV("%s: FPS single not set, using FPS range value", + __FUNCTION__); + } else { + int fpsKeyOrder; + res = newParams.compareSetOrder( + CameraParameters::KEY_PREVIEW_FRAME_RATE, + CameraParameters::KEY_PREVIEW_FPS_RANGE, + &fpsKeyOrder); + LOG_ALWAYS_FATAL_IF(res != OK, "Impossibly bad FPS keys"); + + fpsUseSingleValue = (fpsKeyOrder > 0); + } + + ALOGV("%s: Preview FPS value is used from '%s'", + __FUNCTION__, fpsUseSingleValue ? "single" : "range"); + } newParams.getPreviewFpsRange(&validatedParams.previewFpsRange[0], &validatedParams.previewFpsRange[1]); + validatedParams.previewFpsRange[0] /= kFpsToApiScale; validatedParams.previewFpsRange[1] /= kFpsToApiScale; - // Compare the FPS range value from the last set() to the current set() - // to determine if the client has changed it - if (validatedParams.previewFpsRange[0] != lastSetFpsRange[0] || - validatedParams.previewFpsRange[1] != lastSetFpsRange[1]) { + // Ignore the FPS range if the FPS single has higher precedence + if (!fpsUseSingleValue) { + ALOGV("%s: Preview FPS range (%d, %d)", __FUNCTION__, + validatedParams.previewFpsRange[0], + validatedParams.previewFpsRange[1]); - fpsRangeChanged = true; camera_metadata_ro_entry_t availablePreviewFpsRanges = staticInfo(ANDROID_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES, 2); for (i = 0; i < availablePreviewFpsRanges.count; i += 2) { @@ -1200,14 +1248,13 @@ status_t Parameters::set(const String8& paramString) { } } - // PREVIEW_FRAME_RATE Deprecated, only use if the preview fps range is - // unchanged this time. The single-value FPS is the same as the minimum of - // the range. To detect whether the application has changed the value of - // previewFps, compare against their last-set preview FPS. - if (!fpsRangeChanged) { + // PREVIEW_FRAME_RATE Deprecated + // - Use only if the single FPS value was set later than the FPS range + if (fpsUseSingleValue) { int previewFps = newParams.getPreviewFrameRate(); - int lastSetPreviewFps = params.getPreviewFrameRate(); - if (previewFps != lastSetPreviewFps || recordingHintChanged) { + ALOGV("%s: Preview FPS single value requested: %d", + __FUNCTION__, previewFps); + { camera_metadata_ro_entry_t availableFrameRates = staticInfo(ANDROID_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES); /** @@ -1276,6 +1323,35 @@ status_t Parameters::set(const String8& paramString) { } } + /** + * Update Preview FPS and Preview FPS ranges based on + * what we actually set. + * + * This updates the API-visible (Camera.Parameters#getParameters) values of + * the FPS fields, not only the internal versions. + * + * Order matters: The value that was set last takes precedence. + * - If the client does a setParameters(getParameters()) we retain + * the same order for preview FPS. + */ + if (!fpsUseSingleValue) { + // Set fps single, then fps range (range wins) + newParams.setPreviewFrameRate( + fpsFromRange(/*min*/validatedParams.previewFpsRange[0], + /*max*/validatedParams.previewFpsRange[1])); + newParams.setPreviewFpsRange( + validatedParams.previewFpsRange[0] * kFpsToApiScale, + validatedParams.previewFpsRange[1] * kFpsToApiScale); + } else { + // Set fps range, then fps single (single wins) + newParams.setPreviewFpsRange( + validatedParams.previewFpsRange[0] * kFpsToApiScale, + validatedParams.previewFpsRange[1] * kFpsToApiScale); + // Set this to the same value, but with higher priority + newParams.setPreviewFrameRate( + newParams.getPreviewFrameRate()); + } + // PICTURE_SIZE newParams.getPictureSize(&validatedParams.pictureWidth, &validatedParams.pictureHeight); @@ -1858,23 +1934,23 @@ status_t Parameters::updateRequest(CameraMetadata *request) const { size_t reqFocusingAreasSize = focusingAreas.size() * 5; int32_t *reqFocusingAreas = new int32_t[reqFocusingAreasSize]; - for (size_t i = 0; i < reqFocusingAreasSize; i += 5) { - if (focusingAreas[i].weight != 0) { + for (size_t i = 0, j = 0; i < reqFocusingAreasSize; i += 5, j++) { + if (focusingAreas[j].weight != 0) { reqFocusingAreas[i + 0] = - normalizedXToArray(focusingAreas[i].left); + normalizedXToArray(focusingAreas[j].left); reqFocusingAreas[i + 1] = - normalizedYToArray(focusingAreas[i].top); + normalizedYToArray(focusingAreas[j].top); reqFocusingAreas[i + 2] = - normalizedXToArray(focusingAreas[i].right); + normalizedXToArray(focusingAreas[j].right); reqFocusingAreas[i + 3] = - normalizedYToArray(focusingAreas[i].bottom); + normalizedYToArray(focusingAreas[j].bottom); } else { reqFocusingAreas[i + 0] = 0; reqFocusingAreas[i + 1] = 0; reqFocusingAreas[i + 2] = 0; reqFocusingAreas[i + 3] = 0; } - reqFocusingAreas[i + 4] = focusingAreas[i].weight; + reqFocusingAreas[i + 4] = focusingAreas[j].weight; } res = request->update(ANDROID_CONTROL_AF_REGIONS, reqFocusingAreas, reqFocusingAreasSize); @@ -1887,23 +1963,23 @@ status_t Parameters::updateRequest(CameraMetadata *request) const { size_t reqMeteringAreasSize = meteringAreas.size() * 5; int32_t *reqMeteringAreas = new int32_t[reqMeteringAreasSize]; - for (size_t i = 0; i < reqMeteringAreasSize; i += 5) { - if (meteringAreas[i].weight != 0) { + for (size_t i = 0, j = 0; i < reqMeteringAreasSize; i += 5, j++) { + if (meteringAreas[j].weight != 0) { reqMeteringAreas[i + 0] = - normalizedXToArray(meteringAreas[i].left); + normalizedXToArray(meteringAreas[j].left); reqMeteringAreas[i + 1] = - normalizedYToArray(meteringAreas[i].top); + normalizedYToArray(meteringAreas[j].top); reqMeteringAreas[i + 2] = - normalizedXToArray(meteringAreas[i].right); + normalizedXToArray(meteringAreas[j].right); reqMeteringAreas[i + 3] = - normalizedYToArray(meteringAreas[i].bottom); + normalizedYToArray(meteringAreas[j].bottom); } else { reqMeteringAreas[i + 0] = 0; reqMeteringAreas[i + 1] = 0; reqMeteringAreas[i + 2] = 0; reqMeteringAreas[i + 3] = 0; } - reqMeteringAreas[i + 4] = meteringAreas[i].weight; + reqMeteringAreas[i + 4] = meteringAreas[j].weight; } res = request->update(ANDROID_CONTROL_AE_REGIONS, reqMeteringAreas, reqMeteringAreasSize); diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h index 32dbd42..da07ccf 100644 --- a/services/camera/libcameraservice/api1/client2/Parameters.h +++ b/services/camera/libcameraservice/api1/client2/Parameters.h @@ -25,6 +25,7 @@ #include <utils/Vector.h> #include <utils/KeyedVector.h> #include <camera/CameraParameters.h> +#include <camera/CameraParameters2.h> #include <camera/CameraMetadata.h> namespace android { @@ -32,7 +33,7 @@ namespace camera2 { /** * Current camera state; this is the full state of the Camera under the old - * camera API (contents of the CameraParameters object in a more-efficient + * camera API (contents of the CameraParameters2 object in a more-efficient * format, plus other state). The enum values are mostly based off the * corresponding camera2 enums, not the camera1 strings. A few are defined here * if they don't cleanly map to camera2 values. @@ -128,7 +129,7 @@ struct Parameters { LIGHTFX_HDR } lightFx; - CameraParameters params; + CameraParameters2 params; String8 paramsFlattened; // These parameters are also part of the camera API-visible state, but not |