Merge "NuPlayer2CCDecoder: Add bound check before memcpy"
diff --git a/camera/cameraserver/cameraserver.rc b/camera/cameraserver/cameraserver.rc
index fea5a1d..a9aae0b 100644
--- a/camera/cameraserver/cameraserver.rc
+++ b/camera/cameraserver/cameraserver.rc
@@ -4,3 +4,4 @@
group audio camera input drmrpc
ioprio rt 4
writepid /dev/cpuset/camera-daemon/tasks /dev/stune/top-app/tasks
+ rlimit rtprio 10 10
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index 0501006..1dfa4f7 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -2312,7 +2312,9 @@
* <p>If this device is the largest or only camera device with a given facing, then this
* position will be <code>(0, 0, 0)</code>; a camera device with a lens optical center located 3 cm
* from the main sensor along the +X axis (to the right from the user's perspective) will
- * report <code>(0.03, 0, 0)</code>.</p>
+ * report <code>(0.03, 0, 0)</code>. Note that this means that, for many computer vision
+ * applications, the position needs to be negated to convert it to a translation from the
+ * camera to the origin.</p>
* <p>To transform a pixel coordinates between two cameras facing the same direction, first
* the source camera ACAMERA_LENS_DISTORTION must be corrected for. Then the source
* camera ACAMERA_LENS_INTRINSIC_CALIBRATION needs to be applied, followed by the
@@ -2324,7 +2326,8 @@
* <p>To compare this against a real image from the destination camera, the destination camera
* image then needs to be corrected for radial distortion before comparison or sampling.</p>
* <p>When ACAMERA_LENS_POSE_REFERENCE is GYROSCOPE, then this position is relative to
- * the center of the primary gyroscope on the device.</p>
+ * the center of the primary gyroscope on the device. The axis definitions are the same as
+ * with PRIMARY_CAMERA.</p>
*
* @see ACAMERA_LENS_DISTORTION
* @see ACAMERA_LENS_INTRINSIC_CALIBRATION
@@ -2418,13 +2421,15 @@
* </code></pre>
* <p>which can then be combined with the camera pose rotation
* <code>R</code> and translation <code>t</code> (ACAMERA_LENS_POSE_ROTATION and
- * ACAMERA_LENS_POSE_TRANSLATION, respective) to calculate the
+ * ACAMERA_LENS_POSE_TRANSLATION, respectively) to calculate the
* complete transform from world coordinates to pixel
* coordinates:</p>
- * <pre><code>P = [ K 0 * [ R t
- * 0 1 ] 0 1 ]
+ * <pre><code>P = [ K 0 * [ R -Rt
+ * 0 1 ] 0 1 ]
* </code></pre>
- * <p>and with <code>p_w</code> being a point in the world coordinate system
+ * <p>(Note the negation of poseTranslation when mapping from camera
+ * to world coordinates, and multiplication by the rotation).</p>
+ * <p>With <code>p_w</code> being a point in the world coordinate system
* and <code>p_s</code> being a point in the camera active pixel array
* coordinate system, and with the mapping including the
* homogeneous division by z:</p>
@@ -2446,6 +2451,13 @@
* activeArraySize rectangle), to determine the final pixel
* coordinate of the world point for processed (non-RAW)
* output buffers.</p>
+ * <p>For camera devices, the center of pixel <code>(x,y)</code> is located at
+ * coordinate <code>(x + 0.5, y + 0.5)</code>. So on a device with a
+ * precorrection active array of size <code>(10,10)</code>, the valid pixel
+ * indices go from <code>(0,0)-(9,9)</code>, and an perfectly-built camera would
+ * have an optical center at the exact center of the pixel grid, at
+ * coordinates <code>(5.0, 5.0)</code>, which is the top-left corner of pixel
+ * <code>(5,5)</code>.</p>
*
* @see ACAMERA_LENS_DISTORTION
* @see ACAMERA_LENS_POSE_ROTATION
@@ -3059,7 +3071,7 @@
* outputs will crop horizontally (pillarbox), and 16:9
* streams will match exactly. These additional crops will
* be centered within the crop region.</p>
- * <p>If the coordinate system is android.sensor.info.activeArraysSize, the width and height
+ * <p>If the coordinate system is ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, the width and height
* of the crop region cannot be set to be smaller than
* <code>floor( activeArraySize.width / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code> and
* <code>floor( activeArraySize.height / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code>, respectively.</p>
@@ -4675,8 +4687,8 @@
ACAMERA_STATISTICS_LENS_SHADING_MAP_MODE = // byte (acamera_metadata_enum_android_statistics_lens_shading_map_mode_t)
ACAMERA_STATISTICS_START + 16,
/**
- * <p>A control for selecting whether OIS position information is included in output
- * result metadata.</p>
+ * <p>A control for selecting whether optical stabilization (OIS) position
+ * information is included in output result metadata.</p>
*
* <p>Type: byte (acamera_metadata_enum_android_statistics_ois_data_mode_t)</p>
*
@@ -4686,6 +4698,12 @@
* <li>ACaptureRequest</li>
* </ul></p>
*
+ * <p>Since optical image stabilization generally involves motion much faster than the duration
+ * of individualq image exposure, multiple OIS samples can be included for a single capture
+ * result. For example, if the OIS reporting operates at 200 Hz, a typical camera operating
+ * at 30fps may have 6-7 OIS samples per capture result. This information can be combined
+ * with the rolling shutter skew to account for lens motion during image exposure in
+ * post-processing algorithms.</p>
*/
ACAMERA_STATISTICS_OIS_DATA_MODE = // byte (acamera_metadata_enum_android_statistics_ois_data_mode_t)
ACAMERA_STATISTICS_START + 17,
@@ -4717,11 +4735,15 @@
* </ul></p>
*
* <p>The array contains the amount of shifts in x direction, in pixels, based on OIS samples.
- * A positive value is a shift from left to right in active array coordinate system. For
- * example, if the optical center is (1000, 500) in active array coordinates, a shift of
- * (3, 0) puts the new optical center at (1003, 500).</p>
+ * A positive value is a shift from left to right in the pre-correction active array
+ * coordinate system. For example, if the optical center is (1000, 500) in pre-correction
+ * active array coordinates, a shift of (3, 0) puts the new optical center at (1003, 500).</p>
* <p>The number of shifts must match the number of timestamps in
* ACAMERA_STATISTICS_OIS_TIMESTAMPS.</p>
+ * <p>The OIS samples are not affected by whether lens distortion correction is enabled (on
+ * supporting devices). They are always reported in pre-correction active array coordinates,
+ * since the scaling of OIS shifts would depend on the specific spot on the sensor the shift
+ * is needed.</p>
*
* @see ACAMERA_STATISTICS_OIS_TIMESTAMPS
*/
@@ -4738,11 +4760,15 @@
* </ul></p>
*
* <p>The array contains the amount of shifts in y direction, in pixels, based on OIS samples.
- * A positive value is a shift from top to bottom in active array coordinate system. For
- * example, if the optical center is (1000, 500) in active array coordinates, a shift of
- * (0, 5) puts the new optical center at (1000, 505).</p>
+ * A positive value is a shift from top to bottom in pre-correction active array coordinate
+ * system. For example, if the optical center is (1000, 500) in active array coordinates, a
+ * shift of (0, 5) puts the new optical center at (1000, 505).</p>
* <p>The number of shifts must match the number of timestamps in
* ACAMERA_STATISTICS_OIS_TIMESTAMPS.</p>
+ * <p>The OIS samples are not affected by whether lens distortion correction is enabled (on
+ * supporting devices). They are always reported in pre-correction active array coordinates,
+ * since the scaling of OIS shifts would depend on the specific spot on the sensor the shift
+ * is needed.</p>
*
* @see ACAMERA_STATISTICS_OIS_TIMESTAMPS
*/
@@ -5432,16 +5458,34 @@
* any correction at all would slow down capture rate. Every output stream will have a
* similar amount of enhancement applied.</p>
* <p>The correction only applies to processed outputs such as YUV, JPEG, or DEPTH16; it is not
- * applied to any RAW output. Metadata coordinates such as face rectangles or metering
- * regions are also not affected by correction.</p>
+ * applied to any RAW output.</p>
* <p>This control will be on by default on devices that support this control. Applications
* disabling distortion correction need to pay extra attention with the coordinate system of
* metering regions, crop region, and face rectangles. When distortion correction is OFF,
* metadata coordinates follow the coordinate system of
* ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE. When distortion is not OFF, metadata
- * coordinates follow the coordinate system of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.</p>
+ * coordinates follow the coordinate system of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE. The
+ * camera device will map these metadata fields to match the corrected image produced by the
+ * camera device, for both capture requests and results. However, this mapping is not very
+ * precise, since rectangles do not generally map to rectangles when corrected. Only linear
+ * scaling between the active array and precorrection active array coordinates is
+ * performed. Applications that require precise correction of metadata need to undo that
+ * linear scaling, and apply a more complete correction that takes into the account the app's
+ * own requirements.</p>
+ * <p>The full list of metadata that is affected in this way by distortion correction is:</p>
+ * <ul>
+ * <li>ACAMERA_CONTROL_AF_REGIONS</li>
+ * <li>ACAMERA_CONTROL_AE_REGIONS</li>
+ * <li>ACAMERA_CONTROL_AWB_REGIONS</li>
+ * <li>ACAMERA_SCALER_CROP_REGION</li>
+ * <li>android.statistics.faces</li>
+ * </ul>
*
+ * @see ACAMERA_CONTROL_AE_REGIONS
+ * @see ACAMERA_CONTROL_AF_REGIONS
+ * @see ACAMERA_CONTROL_AWB_REGIONS
* @see ACAMERA_LENS_DISTORTION
+ * @see ACAMERA_SCALER_CROP_REGION
* @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
* @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
*/
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 70c281a..8a97299 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -21,6 +21,9 @@
libsoundtriggerservice \
libutils
+LOCAL_STATIC_LIBRARIES := \
+ libjsoncpp
+
# TODO oboeservice is the old folder name for aaudioservice. It will be changed.
LOCAL_C_INCLUDES := \
frameworks/av/services/audioflinger \
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 91ebf73..84f9c22 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -338,7 +338,7 @@
aaudio_sharing_mode_t requestedInputSharingMode = AAUDIO_SHARING_MODE_SHARED;
int requestedInputChannelCount = NUM_INPUT_CHANNELS;
aaudio_format_t requestedInputFormat = AAUDIO_FORMAT_UNSPECIFIED;
- int32_t requestedInputCapacity = -1;
+ int32_t requestedInputCapacity = AAUDIO_UNSPECIFIED;
aaudio_performance_mode_t inputPerformanceLevel = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
int32_t outputFramesPerBurst = 0;
@@ -459,15 +459,8 @@
argParser.setPerformanceMode(inputPerformanceLevel);
argParser.setChannelCount(requestedInputChannelCount);
argParser.setSharingMode(requestedInputSharingMode);
-
- // Make sure the input buffer has plenty of capacity.
- // Extra capacity on input should not increase latency if we keep it drained.
- int32_t inputBufferCapacity = requestedInputCapacity;
- if (inputBufferCapacity < 0) {
- int32_t outputBufferCapacity = AAudioStream_getBufferCapacityInFrames(outputStream);
- inputBufferCapacity = 2 * outputBufferCapacity;
- }
- argParser.setBufferCapacity(inputBufferCapacity);
+ // Warning! If you change input capacity then you may not get a FAST track on Legacy path.
+ argParser.setBufferCapacity(requestedInputCapacity);
result = recorder.open(argParser);
if (result != AAUDIO_OK) {
diff --git a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
index 9d29cf1..d61efd3 100644
--- a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
+++ b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
@@ -30,6 +30,26 @@
#include <audio_effects/effect_loudnessenhancer.h>
#include "dsp/core/dynamic_range_compression.h"
+// BUILD_FLOAT targets building a float effect instead of the legacy int16_t effect.
+#define BUILD_FLOAT
+
+#ifdef BUILD_FLOAT
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_FLOAT;
+
+#else
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_16_BIT;
+
+static inline int16_t clamp16(int32_t sample)
+{
+ if ((sample>>15) ^ (sample>>31))
+ sample = 0x7FFF ^ (sample>>31);
+ return sample;
+}
+
+#endif // BUILD_FLOAT
+
extern "C" {
// effect_handle_t interface implementation for LE effect
@@ -80,13 +100,6 @@
}
}
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
-
//----------------------------------------------------------------------------
// LE_setConfig()
//----------------------------------------------------------------------------
@@ -111,7 +124,7 @@
if (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO) return -EINVAL;
if (pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_WRITE &&
pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_ACCUMULATE) return -EINVAL;
- if (pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) return -EINVAL;
+ if (pConfig->inputCfg.format != kProcessFormat) return -EINVAL;
pContext->mConfig = *pConfig;
@@ -159,7 +172,7 @@
pContext->mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
pContext->mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
- pContext->mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pContext->mConfig.inputCfg.format = kProcessFormat;
pContext->mConfig.inputCfg.samplingRate = 44100;
pContext->mConfig.inputCfg.bufferProvider.getBuffer = NULL;
pContext->mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
@@ -167,7 +180,7 @@
pContext->mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
pContext->mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
pContext->mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
- pContext->mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pContext->mConfig.outputCfg.format = kProcessFormat;
pContext->mConfig.outputCfg.samplingRate = 44100;
pContext->mConfig.outputCfg.bufferProvider.getBuffer = NULL;
pContext->mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
@@ -284,18 +297,41 @@
//ALOGV("LE about to process %d samples", inBuffer->frameCount);
uint16_t inIdx;
+#ifdef BUILD_FLOAT
+ constexpr float scale = 1 << 15; // power of 2 is lossless conversion to int16_t range
+ constexpr float inverseScale = 1.f / scale;
+ const float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f) * scale;
+#else
float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f);
+#endif
float leftSample, rightSample;
for (inIdx = 0 ; inIdx < inBuffer->frameCount ; inIdx++) {
// makeup gain is applied on the input of the compressor
+#ifdef BUILD_FLOAT
+ leftSample = inputAmp * inBuffer->f32[2*inIdx];
+ rightSample = inputAmp * inBuffer->f32[2*inIdx +1];
+ pContext->mCompressor->Compress(&leftSample, &rightSample);
+ inBuffer->f32[2*inIdx] = leftSample * inverseScale;
+ inBuffer->f32[2*inIdx +1] = rightSample * inverseScale;
+#else
leftSample = inputAmp * (float)inBuffer->s16[2*inIdx];
rightSample = inputAmp * (float)inBuffer->s16[2*inIdx +1];
pContext->mCompressor->Compress(&leftSample, &rightSample);
inBuffer->s16[2*inIdx] = (int16_t) leftSample;
inBuffer->s16[2*inIdx +1] = (int16_t) rightSample;
+#endif // BUILD_FLOAT
}
if (inBuffer->raw != outBuffer->raw) {
+#ifdef BUILD_FLOAT
+ if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+ for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
+ outBuffer->f32[i] += inBuffer->f32[i];
+ }
+ } else {
+ memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(float));
+ }
+#else
if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
outBuffer->s16[i] = clamp16(outBuffer->s16[i] + inBuffer->s16[i]);
@@ -303,6 +339,7 @@
} else {
memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(int16_t));
}
+#endif // BUILD_FLOAT
}
if (pContext->mState != LOUDNESS_ENHANCER_STATE_ACTIVE) {
return -ENODATA;
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index 70409de..3534149 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -19,7 +19,8 @@
LOCAL_MODULE:= libvisualizer
LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-effects)
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils)
LOCAL_HEADER_LIBRARIES += libhardware_headers
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 807f24d..e2ccfb7 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -24,11 +24,25 @@
#include <string.h>
#include <time.h>
+#include <algorithm> // max
#include <new>
#include <log/log.h>
#include <audio_effects/effect_visualizer.h>
+#include <audio_utils/primitives.h>
+
+#define BUILD_FLOAT
+
+#ifdef BUILD_FLOAT
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_FLOAT;
+
+#else
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_16_BIT;
+
+#endif // BUILD_FLOAT
extern "C" {
@@ -146,7 +160,7 @@
if (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO) return -EINVAL;
if (pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_WRITE &&
pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_ACCUMULATE) return -EINVAL;
- if (pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) return -EINVAL;
+ if (pConfig->inputCfg.format != kProcessFormat) return -EINVAL;
pContext->mConfig = *pConfig;
@@ -192,7 +206,7 @@
{
pContext->mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
pContext->mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
- pContext->mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pContext->mConfig.inputCfg.format = kProcessFormat;
pContext->mConfig.inputCfg.samplingRate = 44100;
pContext->mConfig.inputCfg.bufferProvider.getBuffer = NULL;
pContext->mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
@@ -200,7 +214,7 @@
pContext->mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
pContext->mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
pContext->mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
- pContext->mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pContext->mConfig.outputCfg.format = kProcessFormat;
pContext->mConfig.outputCfg.samplingRate = 44100;
pContext->mConfig.outputCfg.bufferProvider.getBuffer = NULL;
pContext->mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
@@ -301,15 +315,8 @@
//--- Effect Control Interface Implementation
//
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
-
int Visualizer_process(
- effect_handle_t self,audio_buffer_t *inBuffer, audio_buffer_t *outBuffer)
+ effect_handle_t self, audio_buffer_t *inBuffer, audio_buffer_t *outBuffer)
{
VisualizerContext * pContext = (VisualizerContext *)self;
@@ -324,20 +331,28 @@
return -EINVAL;
}
+ const size_t sampleLen = inBuffer->frameCount * pContext->mChannelCount;
+
// perform measurements if needed
if (pContext->mMeasurementMode & MEASUREMENT_MODE_PEAK_RMS) {
// find the peak and RMS squared for the new buffer
- uint32_t inIdx;
- int16_t maxSample = 0;
float rmsSqAcc = 0;
- for (inIdx = 0 ; inIdx < inBuffer->frameCount * pContext->mChannelCount ; inIdx++) {
- if (inBuffer->s16[inIdx] > maxSample) {
- maxSample = inBuffer->s16[inIdx];
- } else if (-inBuffer->s16[inIdx] > maxSample) {
- maxSample = -inBuffer->s16[inIdx];
- }
- rmsSqAcc += (inBuffer->s16[inIdx] * inBuffer->s16[inIdx]);
+
+#ifdef BUILD_FLOAT
+ float maxSample = 0.f;
+ for (size_t inIdx = 0; inIdx < sampleLen; ++inIdx) {
+ maxSample = fmax(maxSample, fabs(inBuffer->f32[inIdx]));
+ rmsSqAcc += inBuffer->f32[inIdx] * inBuffer->f32[inIdx];
}
+ maxSample *= 1 << 15; // scale to int16_t, with exactly 1 << 15 representing positive num.
+ rmsSqAcc *= 1 << 30; // scale to int16_t * 2
+#else
+ int maxSample = 0;
+ for (size_t inIdx = 0; inIdx < sampleLen; ++inIdx) {
+ maxSample = std::max(maxSample, std::abs(int32_t(inBuffer->s16[inIdx])));
+ rmsSqAcc += inBuffer->s16[inIdx] * inBuffer->s16[inIdx];
+ }
+#endif
// store the measurement
pContext->mPastMeasurements[pContext->mMeasurementBufferIdx].mPeakU16 = (uint16_t)maxSample;
pContext->mPastMeasurements[pContext->mMeasurementBufferIdx].mRmsSquared =
@@ -348,32 +363,59 @@
}
}
- // all code below assumes stereo 16 bit PCM output and input
+#ifdef BUILD_FLOAT
+ float fscale; // multiplicative scale
+#else
int32_t shift;
+#endif // BUILD_FLOAT
if (pContext->mScalingMode == VISUALIZER_SCALING_MODE_NORMALIZED) {
// derive capture scaling factor from peak value in current buffer
// this gives more interesting captures for display.
- shift = 32;
- int len = inBuffer->frameCount * 2;
- for (int i = 0; i < len; i++) {
+
+#ifdef BUILD_FLOAT
+ float maxSample = 0.f;
+ for (size_t inIdx = 0; inIdx < sampleLen; ++inIdx) {
+ maxSample = fmax(maxSample, fabs(inBuffer->f32[inIdx]));
+ }
+ if (maxSample > 0.f) {
+ constexpr float halfish = 127.f / 256.f;
+ fscale = halfish / maxSample;
+ int exp; // unused
+ const float significand = frexp(fscale, &exp);
+ if (significand == 0.5f) {
+ fscale *= 255.f / 256.f; // avoid returning unaltered PCM signal
+ }
+ } else {
+ // scale doesn't matter, the values are all 0.
+ fscale = 1.f;
+ }
+#else
+ int32_t orAccum = 0;
+ for (size_t i = 0; i < sampleLen; ++i) {
int32_t smp = inBuffer->s16[i];
if (smp < 0) smp = -smp - 1; // take care to keep the max negative in range
- int32_t clz = __builtin_clz(smp);
- if (shift > clz) shift = clz;
+ orAccum |= smp;
}
+
// A maximum amplitude signal will have 17 leading zeros, which we want to
// translate to a shift of 8 (for converting 16 bit to 8 bit)
- shift = 25 - shift;
+ shift = 25 - __builtin_clz(orAccum);
+
// Never scale by less than 8 to avoid returning unaltered PCM signal.
if (shift < 3) {
shift = 3;
}
// add one to combine the division by 2 needed after summing left and right channels below
shift++;
+#endif // BUILD_FLOAT
} else {
assert(pContext->mScalingMode == VISUALIZER_SCALING_MODE_AS_PLAYED);
+#ifdef BUILD_FLOAT
+ fscale = 0.5f; // default divide by 2 to account for sum of L + R.
+#else
shift = 9;
+#endif // BUILD_FLOAT
}
uint32_t captIdx;
@@ -386,9 +428,13 @@
// wrap around
captIdx = 0;
}
- int32_t smp = inBuffer->s16[2 * inIdx] + inBuffer->s16[2 * inIdx + 1];
- smp = smp >> shift;
+#ifdef BUILD_FLOAT
+ const float smp = (inBuffer->f32[2 * inIdx] + inBuffer->f32[2 * inIdx + 1]) * fscale;
+ buf[captIdx] = clamp8_from_float(smp);
+#else
+ const int32_t smp = (inBuffer->s16[2 * inIdx] + inBuffer->s16[2 * inIdx + 1]) >> shift;
buf[captIdx] = ((uint8_t)smp)^0x80;
+#endif // BUILD_FLOAT
}
// XXX the following two should really be atomic, though it probably doesn't
@@ -400,6 +446,15 @@
}
if (inBuffer->raw != outBuffer->raw) {
+#ifdef BUILD_FLOAT
+ if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+ for (size_t i = 0; i < sampleLen; ++i) {
+ outBuffer->f32[i] += inBuffer->f32[i];
+ }
+ } else {
+ memcpy(outBuffer->raw, inBuffer->raw, sampleLen * sizeof(float));
+ }
+#else
if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
outBuffer->s16[i] = clamp16(outBuffer->s16[i] + inBuffer->s16[i]);
@@ -407,6 +462,7 @@
} else {
memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(int16_t));
}
+#endif // BUILD_FLOAT
}
if (pContext->mState != VISUALIZER_STATE_ACTIVE) {
return -ENODATA;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index a5f5fc6..0784939 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1120,6 +1120,7 @@
} else if (what == DecoderBase::kWhatShutdownCompleted) {
ALOGV("%s shutdown completed", audio ? "audio" : "video");
if (audio) {
+ Mutex::Autolock autoLock(mDecoderLock);
mAudioDecoder.clear();
mAudioDecoderError = false;
++mAudioDecoderGeneration;
@@ -1127,6 +1128,7 @@
CHECK_EQ((int)mFlushingAudio, (int)SHUTTING_DOWN_DECODER);
mFlushingAudio = SHUT_DOWN;
} else {
+ Mutex::Autolock autoLock(mDecoderLock);
mVideoDecoder.clear();
mVideoDecoderError = false;
++mVideoDecoderGeneration;
@@ -1447,29 +1449,6 @@
break;
}
- case kWhatGetStats:
- {
- ALOGV("kWhatGetStats");
-
- Vector<sp<AMessage>> *trackStats;
- CHECK(msg->findPointer("trackstats", (void**)&trackStats));
-
- trackStats->clear();
- if (mVideoDecoder != NULL) {
- trackStats->push_back(mVideoDecoder->getStats());
- }
- if (mAudioDecoder != NULL) {
- trackStats->push_back(mAudioDecoder->getStats());
- }
-
- // respond for synchronization
- sp<AMessage> response = new AMessage;
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
- break;
- }
-
default:
TRESPASS();
break;
@@ -1817,6 +1796,7 @@
(long long)currentPositionUs, forceNonOffload, needsToCreateAudioDecoder);
if (mAudioDecoder != NULL) {
mAudioDecoder->pause();
+ Mutex::Autolock autoLock(mDecoderLock);
mAudioDecoder.clear();
mAudioDecoderError = false;
++mAudioDecoderGeneration;
@@ -1935,6 +1915,8 @@
}
}
+ Mutex::Autolock autoLock(mDecoderLock);
+
if (audio) {
sp<AMessage> notify = new AMessage(kWhatAudioNotify, this);
++mAudioDecoderGeneration;
@@ -2236,13 +2218,15 @@
void NuPlayer::getStats(Vector<sp<AMessage> > *trackStats) {
CHECK(trackStats != NULL);
- ALOGV("NuPlayer::getStats()");
- sp<AMessage> msg = new AMessage(kWhatGetStats, this);
- msg->setPointer("trackstats", trackStats);
+ trackStats->clear();
- sp<AMessage> response;
- (void) msg->postAndAwaitResponse(&response);
- // response is for synchronization, ignore contents
+ Mutex::Autolock autoLock(mDecoderLock);
+ if (mVideoDecoder != NULL) {
+ trackStats->push_back(mVideoDecoder->getStats());
+ }
+ if (mAudioDecoder != NULL) {
+ trackStats->push_back(mAudioDecoder->getStats());
+ }
}
sp<MetaData> NuPlayer::getFileMeta() {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index e400d16..9f5be06 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -159,7 +159,6 @@
kWhatPrepareDrm = 'pDrm',
kWhatReleaseDrm = 'rDrm',
kWhatMediaClockNotify = 'mckN',
- kWhatGetStats = 'gSts',
};
wp<NuPlayerDriver> mDriver;
@@ -175,6 +174,7 @@
sp<DecoderBase> mVideoDecoder;
bool mOffloadAudio;
sp<DecoderBase> mAudioDecoder;
+ Mutex mDecoderLock; // guard |mAudioDecoder| and |mVideoDecoder|.
sp<CCDecoder> mCCDecoder;
sp<Renderer> mRenderer;
sp<ALooper> mRendererLooper;
diff --git a/media/libstagefright/MetaDataUtils.cpp b/media/libstagefright/MetaDataUtils.cpp
index 04f6ade..2475e7b 100644
--- a/media/libstagefright/MetaDataUtils.cpp
+++ b/media/libstagefright/MetaDataUtils.cpp
@@ -16,8 +16,10 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MetaDataUtils"
+#include <utils/Log.h>
#include <media/stagefright/foundation/avc_utils.h>
+#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaDataUtils.h>
@@ -25,6 +27,10 @@
namespace android {
bool MakeAVCCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size) {
+ if (data == nullptr || size == 0) {
+ return false;
+ }
+
int32_t width;
int32_t height;
int32_t sarWidth;
@@ -46,6 +52,44 @@
return true;
}
+bool MakeAACCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size) {
+ if (data == nullptr || size < 7) {
+ return false;
+ }
+
+ ABitReader bits(data, size);
+
+ // adts_fixed_header
+
+ if (bits.getBits(12) != 0xfffu) {
+ ALOGE("Wrong atds_fixed_header");
+ return false;
+ }
+
+ bits.skipBits(4); // ID, layer, protection_absent
+
+ unsigned profile = bits.getBits(2);
+ if (profile == 3u) {
+ ALOGE("profile should not be 3");
+ return false;
+ }
+ unsigned sampling_freq_index = bits.getBits(4);
+ bits.getBits(1); // private_bit
+ unsigned channel_configuration = bits.getBits(3);
+ if (channel_configuration == 0u) {
+ ALOGE("channel_config should not be 0");
+ return false;
+ }
+
+ if (!MakeAACCodecSpecificData(
+ meta, profile, sampling_freq_index, channel_configuration)) {
+ return false;
+ }
+
+ meta.setInt32(kKeyIsADTS, true);
+ return true;
+}
+
bool MakeAACCodecSpecificData(
MetaDataBase &meta,
unsigned profile, unsigned sampling_freq_index,
diff --git a/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp b/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
index f173e0f..06b15b3 100644
--- a/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
+++ b/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
@@ -689,7 +689,6 @@
notify(OMX_EventError, OMX_ErrorUndefined, err_code, NULL);
return;
}
- mIsCodecConfigFlushRequired = true;
}
if (!mSampFreq || !mNumChannels) {
@@ -713,10 +712,14 @@
signed int bytesConsumed = 0;
int errorCode = 0;
if (mIsCodecInitialized) {
+ mIsCodecConfigFlushRequired = true;
errorCode =
decodeXAACStream(inBuffer, inBufferLength, &bytesConsumed, &numOutBytes);
- } else {
+ } else if (!mIsCodecConfigFlushRequired) {
ALOGW("Assumption that first frame after header initializes decoder failed!");
+ mSignalledError = true;
+ notify(OMX_EventError, OMX_ErrorUndefined, -1, NULL);
+ return;
}
inHeader->nFilledLen -= bytesConsumed;
inHeader->nOffset += bytesConsumed;
diff --git a/media/libstagefright/include/media/stagefright/MetaDataUtils.h b/media/libstagefright/include/media/stagefright/MetaDataUtils.h
index d5a8080..4a7107d 100644
--- a/media/libstagefright/include/media/stagefright/MetaDataUtils.h
+++ b/media/libstagefright/include/media/stagefright/MetaDataUtils.h
@@ -24,6 +24,7 @@
struct ABuffer;
bool MakeAVCCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size);
+bool MakeAACCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size);
bool MakeAACCodecSpecificData(MetaDataBase &meta, unsigned profile, unsigned sampling_freq_index,
unsigned channel_configuration);
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 32635d1..fb498d4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -775,10 +775,12 @@
ALOGV("new stream PID 0x%02x, type 0x%02x, scrambled %d, SampleEncrypted: %d",
elementaryPID, streamType, mScrambled, mSampleEncrypted);
- uint32_t flags =
- (isVideo() && mScrambled) ? ElementaryStreamQueue::kFlag_ScrambledData :
- (mSampleEncrypted) ? ElementaryStreamQueue::kFlag_SampleEncryptedData :
- 0;
+ uint32_t flags = 0;
+ if (((isVideo() || isAudio()) && mScrambled)) {
+ flags = ElementaryStreamQueue::kFlag_ScrambledData;
+ } else if (mSampleEncrypted) {
+ flags = ElementaryStreamQueue::kFlag_SampleEncryptedData;
+ }
ElementaryStreamQueue::Mode mode = ElementaryStreamQueue::INVALID;
@@ -1504,7 +1506,13 @@
descrambleBytes = bytesWritten;
}
- sp<ABuffer> buffer;
+ // |buffer| points to the buffer from which we'd parse the PES header.
+ // When the output stream is scrambled, it points to mDescrambledBuffer
+ // (unless all packets in this PES are actually clear, in which case,
+ // it points to mBuffer since we never copied into mDescrambledBuffer).
+ // When the output stream is clear, it points to mBuffer, and we'll
+ // copy all descrambled data back to mBuffer.
+ sp<ABuffer> buffer = mBuffer;
if (mQueue->isScrambled()) {
// Queue subSample info for scrambled queue
sp<ABuffer> clearSizesBuffer = new ABuffer(mSubSamples.size() * 4);
@@ -1533,15 +1541,18 @@
}
// Pass the original TS subsample size now. The PES header adjust
// will be applied when the scrambled AU is dequeued.
+ // Note that if descrambleBytes is 0, it means this PES contains only
+ // all ts packets, leadingClearBytes is entire buffer size.
mQueue->appendScrambledData(
- mBuffer->data(), mBuffer->size(), sctrl,
- isSync, clearSizesBuffer, encSizesBuffer);
+ mBuffer->data(), mBuffer->size(),
+ (descrambleBytes > 0) ? descrambleBytes : mBuffer->size(),
+ sctrl, isSync, clearSizesBuffer, encSizesBuffer);
- buffer = mDescrambledBuffer;
+ if (descrambleBytes > 0) {
+ buffer = mDescrambledBuffer;
+ }
} else {
memcpy(mBuffer->data(), mDescrambledBuffer->data(), descrambleBytes);
-
- buffer = mBuffer;
}
ABitReader br(buffer->data(), buffer->size());
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 90005c3..fb8b9fd 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -439,7 +439,8 @@
ALOGE("appending data after EOS");
return ERROR_MALFORMED;
}
- if (mBuffer == NULL || mBuffer->size() == 0) {
+
+ if (!isScrambled() && (mBuffer == NULL || mBuffer->size() == 0)) {
switch (mMode) {
case H264:
case MPEG_VIDEO:
@@ -698,6 +699,7 @@
void ElementaryStreamQueue::appendScrambledData(
const void *data, size_t size,
+ size_t leadingClearBytes,
int32_t keyId, bool isSync,
sp<ABuffer> clearSizes, sp<ABuffer> encSizes) {
if (!isScrambled()) {
@@ -725,6 +727,7 @@
ScrambledRangeInfo scrambledInfo;
scrambledInfo.mLength = size;
+ scrambledInfo.mLeadingClearBytes = leadingClearBytes;
scrambledInfo.mKeyId = keyId;
scrambledInfo.mIsSync = isSync;
scrambledInfo.mClearSizes = clearSizes;
@@ -737,7 +740,6 @@
sp<ABuffer> ElementaryStreamQueue::dequeueScrambledAccessUnit() {
size_t nextScan = mBuffer->size();
- mBuffer->setRange(0, 0);
int32_t pesOffset = 0, pesScramblingControl = 0;
int64_t timeUs = fetchTimestamp(nextScan, &pesOffset, &pesScramblingControl);
if (timeUs < 0ll) {
@@ -748,6 +750,7 @@
// return scrambled unit
int32_t keyId = pesScramblingControl, isSync = 0, scrambledLength = 0;
sp<ABuffer> clearSizes, encSizes;
+ size_t leadingClearBytes;
while (mScrambledRangeInfos.size() > mRangeInfos.size()) {
auto it = mScrambledRangeInfos.begin();
ALOGV("[stream %d] fetching scrambled range: size=%zu", mMode, it->mLength);
@@ -765,6 +768,7 @@
clearSizes = it->mClearSizes;
encSizes = it->mEncSizes;
isSync = it->mIsSync;
+ leadingClearBytes = it->mLeadingClearBytes;
mScrambledRangeInfos.erase(it);
}
if (scrambledLength == 0) {
@@ -772,6 +776,70 @@
return NULL;
}
+ // Retrieve the leading clear bytes info, and use it to set the clear
+ // range on mBuffer. Note that the leading clear bytes includes the
+ // PES header portion, while mBuffer doesn't.
+ if ((int32_t)leadingClearBytes > pesOffset) {
+ mBuffer->setRange(0, leadingClearBytes - pesOffset);
+ } else {
+ mBuffer->setRange(0, 0);
+ }
+
+ // Try to parse formats, and if unavailable set up a dummy format.
+ // Only support the following modes for scrambled content for now.
+ // (will be expanded later).
+ if (mFormat == NULL) {
+ mFormat = new MetaData;
+ switch (mMode) {
+ case H264:
+ {
+ if (!MakeAVCCodecSpecificData(
+ *mFormat, mBuffer->data(), mBuffer->size())) {
+ ALOGI("Creating dummy AVC format for scrambled content");
+
+ mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+ mFormat->setInt32(kKeyWidth, 1280);
+ mFormat->setInt32(kKeyHeight, 720);
+ }
+ break;
+ }
+ case AAC:
+ {
+ if (!MakeAACCodecSpecificData(
+ *mFormat, mBuffer->data(), mBuffer->size())) {
+ ALOGI("Creating dummy AAC format for scrambled content");
+
+ MakeAACCodecSpecificData(*mFormat,
+ 1 /*profile*/, 7 /*sampling_freq_index*/, 1 /*channel_config*/);
+ mFormat->setInt32(kKeyIsADTS, true);
+ }
+
+ break;
+ }
+ case MPEG_VIDEO:
+ {
+ ALOGI("Creating dummy MPEG format for scrambled content");
+
+ mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG2);
+ mFormat->setInt32(kKeyWidth, 1280);
+ mFormat->setInt32(kKeyHeight, 720);
+ break;
+ }
+ default:
+ {
+ ALOGE("Unknown mode for scrambled content");
+ return NULL;
+ }
+ }
+
+ // for MediaExtractor.CasInfo
+ mFormat->setInt32(kKeyCASystemID, mCASystemId);
+ mFormat->setData(kKeyCASessionID,
+ 0, mCasSessionId.data(), mCasSessionId.size());
+ }
+
+ mBuffer->setRange(0, 0);
+
// copy into scrambled access unit
sp<ABuffer> scrambledAccessUnit = ABuffer::CreateAsCopy(
mScrambledBuffer->data(), scrambledLength);
@@ -803,7 +871,11 @@
}
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() {
- if ((mFlags & kFlag_AlignedData) && mMode == H264 && !isScrambled()) {
+ if (isScrambled()) {
+ return dequeueScrambledAccessUnit();
+ }
+
+ if ((mFlags & kFlag_AlignedData) && mMode == H264) {
if (mRangeInfos.empty()) {
return NULL;
}
@@ -1114,25 +1186,11 @@
bool protection_absent = bits.getBits(1) != 0;
if (mFormat == NULL) {
- unsigned profile = bits.getBits(2);
- if (profile == 3u) {
- ALOGE("profile should not be 3");
- return NULL;
- }
- unsigned sampling_freq_index = bits.getBits(4);
- bits.getBits(1); // private_bit
- unsigned channel_configuration = bits.getBits(3);
- if (channel_configuration == 0u) {
- ALOGE("channel_config should not be 0");
- return NULL;
- }
- bits.skipBits(2); // original_copy, home
-
mFormat = new MetaData;
- MakeAACCodecSpecificData(*mFormat,
- profile, sampling_freq_index, channel_configuration);
-
- mFormat->setInt32(kKeyIsADTS, true);
+ if (!MakeAACCodecSpecificData(
+ *mFormat, mBuffer->data() + offset, mBuffer->size() - offset)) {
+ return NULL;
+ }
int32_t sampleRate;
int32_t numChannels;
@@ -1147,12 +1205,12 @@
ALOGI("found AAC codec config (%d Hz, %d channels)",
sampleRate, numChannels);
- } else {
- // profile_ObjectType, sampling_frequency_index, private_bits,
- // channel_configuration, original_copy, home
- bits.skipBits(12);
}
+ // profile_ObjectType, sampling_frequency_index, private_bits,
+ // channel_configuration, original_copy, home
+ bits.skipBits(12);
+
// adts_variable_header
// copyright_identification_bit, copyright_identification_start
@@ -1267,27 +1325,6 @@
}
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitH264() {
- if (isScrambled()) {
- if (mBuffer == NULL || mBuffer->size() == 0) {
- return NULL;
- }
- if (mFormat == NULL) {
- mFormat = new MetaData;
- if (!MakeAVCCodecSpecificData(*mFormat, mBuffer->data(), mBuffer->size())) {
- ALOGW("Creating dummy AVC format for scrambled content");
- mFormat = new MetaData;
- mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
- mFormat->setInt32(kKeyWidth, 1280);
- mFormat->setInt32(kKeyHeight, 720);
- }
- // for MediaExtractor.CasInfo
- mFormat->setInt32(kKeyCASystemID, mCASystemId);
- mFormat->setData(kKeyCASessionID, 0,
- mCasSessionId.data(), mCasSessionId.size());
- }
- return dequeueScrambledAccessUnit();
- }
-
const uint8_t *data = mBuffer->data();
size_t size = mBuffer->size();
@@ -1587,25 +1624,6 @@
}
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitMPEGVideo() {
- if (isScrambled()) {
- if (mBuffer == NULL || mBuffer->size() == 0) {
- return NULL;
- }
- if (mFormat == NULL) {
- ALOGI("Creating dummy MPEG format for scrambled content");
- mFormat = new MetaData;
- mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG2);
- mFormat->setInt32(kKeyWidth, 1280);
- mFormat->setInt32(kKeyHeight, 720);
-
- // for MediaExtractor.CasInfo
- mFormat->setInt32(kKeyCASystemID, mCASystemId);
- mFormat->setData(kKeyCASessionID, 0,
- mCasSessionId.data(), mCasSessionId.size());
- }
- return dequeueScrambledAccessUnit();
- }
-
const uint8_t *data = mBuffer->data();
size_t size = mBuffer->size();
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 8c1d112..3227f47 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -61,6 +61,7 @@
void appendScrambledData(
const void *data, size_t size,
+ size_t leadingClearBytes,
int32_t keyId, bool isSync,
sp<ABuffer> clearSizes, sp<ABuffer> encSizes);
@@ -86,8 +87,8 @@
};
struct ScrambledRangeInfo {
- //int64_t mTimestampUs;
size_t mLength;
+ size_t mLeadingClearBytes;
int32_t mKeyId;
int32_t mIsSync;
sp<ABuffer> mClearSizes;
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index c0aa477..2c26ba4 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -42,6 +42,7 @@
LOCAL_STATIC_LIBRARIES := \
libcpustats \
+ libjsoncpp \
libsndfile \
LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 38483c3..43566b7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -58,6 +58,8 @@
#include <audio_utils/primitives.h>
+#include <json/json.h>
+
#include <powermanager/PowerManager.h>
#include <media/IMediaLogService.h>
@@ -440,8 +442,11 @@
const bool formatJson = std::any_of(args.begin(), args.end(),
[](const String16 &arg) { return arg == String16("--json"); });
if (formatJson) {
+ Json::Value root = getJsonDump();
+ Json::FastWriter writer;
+ std::string rootStr = writer.write(root);
// XXX consider buffering if the string happens to be too long.
- dprintf(fd, "%s", getJsonString().c_str());
+ dprintf(fd, "%s", rootStr.c_str());
return NO_ERROR;
}
@@ -556,34 +561,30 @@
return NO_ERROR;
}
-std::string AudioFlinger::getJsonString()
+Json::Value AudioFlinger::getJsonDump()
{
- std::string jsonStr = "{\n";
+ Json::Value root(Json::objectValue);
const bool locked = dumpTryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
- jsonStr += " \"deadlock_message\": ";
- jsonStr += kDeadlockedString;
- jsonStr += ",\n";
+ root["deadlock_message"] = kDeadlockedString;
}
// FIXME risky to access data structures without a lock held?
- jsonStr += " \"Playback_Threads\": [\n";
+ Json::Value playbackThreads = Json::arrayValue;
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- if (i != 0) {
- jsonStr += ",\n";
- }
- jsonStr += mPlaybackThreads.valueAt(i)->getJsonString();
+ playbackThreads.append(mPlaybackThreads.valueAt(i)->getJsonDump());
}
- jsonStr += "\n ]\n}\n";
if (locked) {
mLock.unlock();
}
- return jsonStr;
+ root["playback_threads"] = playbackThreads;
+
+ return root;
}
sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 9b9a15d..e9e6e94 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -79,6 +79,7 @@
#include <powermanager/IPowerManager.h>
+#include <json/json.h>
#include <media/nblog/NBLog.h>
#include <private/media/AudioEffectShared.h>
#include <private/media/AudioTrackShared.h>
@@ -114,7 +115,7 @@
static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
- std::string getJsonString();
+ Json::Value getJsonDump();
// IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index e26dca1..2abfbfb 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -24,6 +24,7 @@
#include <cpustats/ThreadCpuUsage.h>
#endif
#endif
+#include <json/json.h>
#include <string>
#include <utils/Debug.h>
#include <utils/Log.h>
@@ -205,14 +206,13 @@
}
}
-// TODO get rid of extraneous lines and use better key names.
-// TODO may go back to using a library to do the json formatting.
-std::string FastMixerDumpState::getJsonString() const
+Json::Value FastMixerDumpState::getJsonDump() const
{
+ Json::Value root(Json::objectValue);
if (mCommand == FastMixerState::INITIAL) {
- return " {\n \"status\": \"uninitialized\"\n }";
+ root["status"] = "uninitialized";
+ return root;
}
- std::string jsonStr = " {\n";
#ifdef FAST_THREAD_STATISTICS
// find the interval of valid samples
const uint32_t bounds = mBounds;
@@ -230,31 +230,25 @@
}
// statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
// and adjusted CPU load in MHz normalized for CPU clock frequency
- std::string jsonWallStr = " \"wall_clock_time\":[";
- std::string jsonLoadNsStr = " \"raw_cpu_load\":[";
+ Json::Value jsonWall(Json::arrayValue);
+ Json::Value jsonLoadNs(Json::arrayValue);
// loop over all the samples
for (uint32_t j = 0; j < n; ++j) {
size_t i = oldestClosed++ & (mSamplingN - 1);
uint32_t wallNs = mMonotonicNs[i];
- if (j != 0) {
- jsonWallStr += ',';
- jsonLoadNsStr += ',';
- }
- /* jsonObject["wall"].append(wallNs); */
- jsonWallStr += std::to_string(wallNs);
+ jsonWall.append(wallNs);
uint32_t sampleLoadNs = mLoadNs[i];
- jsonLoadNsStr += std::to_string(sampleLoadNs);
+ jsonLoadNs.append(sampleLoadNs);
}
- jsonWallStr += ']';
- jsonLoadNsStr += ']';
if (n) {
- jsonStr += jsonWallStr + ",\n" + jsonLoadNsStr + "\n";
+ root["wall_clock_time_ns"] = jsonWall;
+ root["raw_cpu_load_ns"] = jsonLoadNs;
+ root["status"] = "ok";
} else {
- //dprintf(fd, " No FastMixer statistics available currently\n");
+ root["status"] = "unavailable";
}
#endif
- jsonStr += " }";
- return jsonStr;
+ return root;
}
} // android
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 81c4175..69c2e4e 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -20,6 +20,7 @@
#include <stdint.h>
#include <string>
#include <audio_utils/TimestampVerifier.h>
+#include <json/json.h>
#include "Configuration.h"
#include "FastThreadDumpState.h"
#include "FastMixerState.h"
@@ -67,7 +68,7 @@
/*virtual*/ ~FastMixerDumpState();
void dump(int fd) const; // should only be called on a stable copy, not the original
- std::string getJsonString() const; // should only be called on a stable copy, not the original
+ Json::Value getJsonDump() const; // should only be called on a stable copy, not the original
double mLatencyMs = 0.; // measured latency, default of 0 if no valid timestamp read.
uint32_t mWriteSequence; // incremented before and after each write()
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 3ef8ce2..cd585f5 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -42,6 +42,7 @@
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
#include <audio_utils/minifloat.h>
+#include <json/json.h>
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_aec.h>
#include <system/audio.h>
@@ -1763,9 +1764,9 @@
mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
}
-std::string AudioFlinger::PlaybackThread::getJsonString() const
+Json::Value AudioFlinger::PlaybackThread::getJsonDump() const
{
- return "{}";
+ return Json::Value(Json::objectValue);
}
void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
@@ -5141,14 +5142,20 @@
}
}
-std::string AudioFlinger::MixerThread::getJsonString() const
+Json::Value AudioFlinger::MixerThread::getJsonDump() const
{
- // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
- // while we are dumping it. It may be inconsistent, but it won't mutate!
- // This is a large object so we place it on the heap.
- // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
- return std::unique_ptr<FastMixerDumpState>(new FastMixerDumpState(mFastMixerDumpState))
- ->getJsonString();
+ Json::Value root;
+ if (hasFastMixer()) {
+ // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+ // while we are dumping it. It may be inconsistent, but it won't mutate!
+ // This is a large object so we place it on the heap.
+ // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
+ const std::unique_ptr<FastMixerDumpState> copy(new FastMixerDumpState(mFastMixerDumpState));
+ root["fastmixer_stats"] = copy->getJsonDump();
+ } else {
+ root["fastmixer_stats"] = "no_fastmixer";
+ }
+ return root;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index f0d625c..2ab0bee 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -662,7 +662,7 @@
void dump(int fd, const Vector<String16>& args);
// returns a string of audio performance related data in JSON format.
- virtual std::string getJsonString() const;
+ virtual Json::Value getJsonDump() const;
// Thread virtuals
virtual bool threadLoop();
@@ -1108,7 +1108,7 @@
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
virtual void dumpInternals(int fd, const Vector<String16>& args);
- std::string getJsonString() const override;
+ Json::Value getJsonDump() const override;
virtual bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index d1515e2..96079cc 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -4761,6 +4761,7 @@
nextAudioPortGeneration();
+ audio_devices_t device = inputDesc->mDevice;
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
@@ -4771,6 +4772,12 @@
inputDesc->close();
mInputs.removeItem(input);
+
+ audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
+ if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
+ SoundTrigger::setCaptureState(false);
+ }
}
SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 491ed72..4c6718c 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2738,13 +2738,13 @@
status_t Camera3Device::registerInFlight(uint32_t frameNumber,
int32_t numBuffers, CaptureResultExtras resultExtras, bool hasInput,
bool hasAppCallback, nsecs_t maxExpectedDuration,
- std::set<String8>& physicalCameraIds) {
+ std::set<String8>& physicalCameraIds, bool isStillCapture) {
ATRACE_CALL();
Mutex::Autolock l(mInFlightLock);
ssize_t res;
res = mInFlightMap.add(frameNumber, InFlightRequest(numBuffers, resultExtras, hasInput,
- hasAppCallback, maxExpectedDuration, physicalCameraIds));
+ hasAppCallback, maxExpectedDuration, physicalCameraIds, isStillCapture));
if (res < 0) return res;
if (mInFlightMap.size() == 1) {
@@ -2810,6 +2810,10 @@
if (request.numBuffersLeft == 0 &&
(request.skipResultMetadata ||
(request.haveResultMetadata && shutterTimestamp != 0))) {
+ if (request.stillCapture) {
+ ATRACE_ASYNC_END("still capture", frameNumber);
+ }
+
ATRACE_ASYNC_END("frame capture", frameNumber);
// Sanity check - if sensor timestamp matches shutter timestamp in the
@@ -4939,12 +4943,21 @@
if (batchedRequest && i != mNextRequests.size()-1) {
hasCallback = false;
}
+ bool isStillCapture = false;
+ if (!mNextRequests[0].captureRequest->mSettingsList.begin()->metadata.isEmpty()) {
+ camera_metadata_ro_entry_t e = camera_metadata_ro_entry_t();
+ find_camera_metadata_ro_entry(halRequest->settings, ANDROID_CONTROL_CAPTURE_INTENT, &e);
+ if ((e.count > 0) && (e.data.u8[0] == ANDROID_CONTROL_CAPTURE_INTENT_STILL_CAPTURE)) {
+ isStillCapture = true;
+ ATRACE_ASYNC_BEGIN("still capture", mNextRequests[i].halRequest.frame_number);
+ }
+ }
res = parent->registerInFlight(halRequest->frame_number,
totalNumBuffers, captureRequest->mResultExtras,
/*hasInput*/halRequest->input_buffer != NULL,
hasCallback,
calculateMaxExpectedDuration(halRequest->settings),
- requestedPhysicalCameras);
+ requestedPhysicalCameras, isStillCapture);
ALOGVV("%s: registered in flight requestId = %" PRId32 ", frameNumber = %" PRId64
", burstId = %" PRId32 ".",
__FUNCTION__,
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 96212ab..51e1fb0 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -994,6 +994,9 @@
// Map of physicalCameraId <-> Metadata
std::vector<PhysicalCaptureResultInfo> physicalMetadatas;
+ // Indicates a still capture request.
+ bool stillCapture;
+
// Default constructor needed by KeyedVector
InFlightRequest() :
shutterTimestamp(0),
@@ -1004,12 +1007,13 @@
hasInputBuffer(false),
hasCallback(true),
maxExpectedDuration(kDefaultExpectedDuration),
- skipResultMetadata(false) {
+ skipResultMetadata(false),
+ stillCapture(false) {
}
InFlightRequest(int numBuffers, CaptureResultExtras extras, bool hasInput,
bool hasAppCallback, nsecs_t maxDuration,
- const std::set<String8>& physicalCameraIdSet) :
+ const std::set<String8>& physicalCameraIdSet, bool isStillCapture) :
shutterTimestamp(0),
sensorTimestamp(0),
requestStatus(OK),
@@ -1020,7 +1024,8 @@
hasCallback(hasAppCallback),
maxExpectedDuration(maxDuration),
skipResultMetadata(false),
- physicalCameraIds(physicalCameraIdSet) {
+ physicalCameraIds(physicalCameraIdSet),
+ stillCapture(isStillCapture) {
}
};
@@ -1034,10 +1039,10 @@
nsecs_t mExpectedInflightDuration = 0;
int mInFlightStatusId;
-
status_t registerInFlight(uint32_t frameNumber,
int32_t numBuffers, CaptureResultExtras resultExtras, bool hasInput,
- bool callback, nsecs_t maxExpectedDuration, std::set<String8>& physicalCameraIds);
+ bool callback, nsecs_t maxExpectedDuration, std::set<String8>& physicalCameraIds,
+ bool isStillCapture);
/**
* Returns the maximum expected time it'll take for all currently in-flight