am 08e42967: Merge "Add support for retrieving location information in mp4/3gpp files" into ics-mr1
* commit '08e42967b9e459b7beb22e859f1f1eba0546741b':
Add support for retrieving location information in mp4/3gpp files
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index d1a8105..922452e 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -416,7 +416,7 @@
*/
status_t dump(int fd, const Vector<String16>& args) const;
-private:
+protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
@@ -487,8 +487,27 @@
int mAuxEffectId;
Mutex mLock;
status_t mRestoreStatus;
+ bool mIsTimed;
};
+class TimedAudioTrack : public AudioTrack
+{
+public:
+ TimedAudioTrack();
+
+ /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
+ status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
+
+ /* queue a buffer obtained via allocateTimedBuffer for playback at the
+ given timestamp */
+ status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+
+ /* define a transform between media time and either common time or
+ local time */
+ enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target);
+};
}; // namespace android
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9e3cb7f..f72452c 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -54,6 +54,7 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
+ bool isTimed,
int *sessionId,
status_t *status) = 0;
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 47d530b..7343a48 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -24,7 +24,7 @@
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
-
+#include <utils/LinearTransform.h>
namespace android {
@@ -69,6 +69,23 @@
/* get this tracks control block */
virtual sp<IMemory> getCblk() const = 0;
+
+ /* Allocate a shared memory buffer suitable for holding timed audio
+ samples */
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) = 0;
+
+ /* Queue a buffer obtained via allocateTimedBuffer for playback at the given
+ timestamp */
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) = 0;
+
+ /* Define the linear transform that will be applied to the timestamps
+ given to queueTimedBuffer (which are expressed in media time).
+ Target specifies whether this transform converts media time to local time
+ or Tungsten time. The values for target are defined in AudioTrack.h */
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 80f43a3..354f91d 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -46,6 +46,9 @@
// The shared library with the test player is passed passed as an
// argument to the 'test:' url in the setDataSource call.
TEST_PLAYER = 5,
+
+ AAH_RX_PLAYER = 100,
+ AAH_TX_PLAYER = 101,
};
diff --git a/media/libaah_rtp/Android.mk b/media/libaah_rtp/Android.mk
new file mode 100644
index 0000000..1dceda9
--- /dev/null
+++ b/media/libaah_rtp/Android.mk
@@ -0,0 +1,40 @@
+LOCAL_PATH:= $(call my-dir)
+#
+# libaah_rtp
+#
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libaah_rtp
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := \
+ aah_decoder_pump.cpp \
+ aah_rx_player.cpp \
+ aah_rx_player_core.cpp \
+ aah_rx_player_ring_buffer.cpp \
+ aah_rx_player_substream.cpp \
+ aah_tx_packet.cpp \
+ aah_tx_player.cpp \
+ aah_tx_sender.cpp \
+ pipe_event.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media \
+ frameworks/base/media/libstagefright
+
+LOCAL_SHARED_LIBRARIES := \
+ libaah_timesrv_client \
+ libbinder \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libutils
+
+LOCAL_LDLIBS := \
+ -lpthread
+
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/media/libaah_rtp/aah_decoder_pump.cpp b/media/libaah_rtp/aah_decoder_pump.cpp
new file mode 100644
index 0000000..b2efbe6
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.cpp
@@ -0,0 +1,457 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <poll.h>
+#include <pthread.h>
+
+#include <aah_timesrv/cc_helper.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+
+namespace android {
+
+AAH_DecoderPump::AAH_DecoderPump(OMXClient& omx)
+ : omx_(omx)
+ , thread_status_(OK)
+ , renderer_(NULL)
+ , last_queued_pts_valid_(false)
+ , last_queued_pts_(0)
+ , last_ts_transform_valid_(false)
+ , last_volume_(0xFF) {
+ thread_ = new ThreadWrapper(this);
+}
+
+AAH_DecoderPump::~AAH_DecoderPump() {
+ shutdown();
+}
+
+status_t AAH_DecoderPump::initCheck() {
+ if (thread_ == NULL) {
+ LOGE("Failed to allocate thread");
+ return NO_MEMORY;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::queueForDecode(MediaBuffer* buf) {
+ if (NULL == buf) {
+ return BAD_VALUE;
+ }
+
+ if (OK != thread_status_) {
+ return thread_status_;
+ }
+
+ { // Explicit scope for AutoMutex pattern.
+ AutoMutex lock(&thread_lock_);
+ in_queue_.push_back(buf);
+ }
+
+ thread_cond_.signal();
+
+ return OK;
+}
+
+void AAH_DecoderPump::queueToRenderer(MediaBuffer* decoded_sample) {
+ Mutex::Autolock lock(&render_lock_);
+ sp<MetaData> meta;
+ int64_t ts;
+ status_t res;
+
+ // Fetch the metadata and make sure the sample has a timestamp. We
+ // cannot render samples which are missing PTSs.
+ meta = decoded_sample->meta_data();
+ if ((meta == NULL) || (!meta->findInt64(kKeyTime, &ts))) {
+ LOGV("Decoded sample missing timestamp, cannot render.");
+ } else {
+ // If we currently are not holding on to a renderer, go ahead and
+ // make one now.
+ if (NULL == renderer_) {
+ renderer_ = new TimedAudioTrack();
+ if (NULL != renderer_) {
+ int frameCount;
+ AudioTrack::getMinFrameCount(&frameCount,
+ AUDIO_STREAM_DEFAULT,
+ static_cast<int>(format_sample_rate_));
+ int ch_format = (format_channels_ == 1)
+ ? AUDIO_CHANNEL_OUT_MONO
+ : AUDIO_CHANNEL_OUT_STEREO;
+
+ res = renderer_->set(AUDIO_STREAM_DEFAULT,
+ format_sample_rate_,
+ AUDIO_FORMAT_PCM_16_BIT,
+ ch_format,
+ frameCount);
+ if (res != OK) {
+ LOGE("Failed to setup audio renderer. (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ CHECK(last_ts_transform_valid_);
+
+ res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ LOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ float volume = static_cast<float>(last_volume_)
+ / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ LOGW("%s: setVolume failed", __FUNCTION__);
+ }
+
+ renderer_->start();
+ }
+ }
+ } else {
+ LOGE("Failed to allocate AudioTrack to use as a renderer.");
+ }
+ }
+
+ if (NULL != renderer_) {
+ uint8_t* decoded_data =
+ reinterpret_cast<uint8_t*>(decoded_sample->data());
+ uint32_t decoded_amt = decoded_sample->range_length();
+ decoded_data += decoded_sample->range_offset();
+
+ sp<IMemory> pcm_payload;
+ res = renderer_->allocateTimedBuffer(decoded_amt, &pcm_payload);
+ if (res != OK) {
+ LOGE("Failed to allocate %d byte audio track buffer."
+ " (res = %d)", decoded_amt, res);
+ } else {
+ memcpy(pcm_payload->pointer(), decoded_data, decoded_amt);
+
+ res = renderer_->queueTimedBuffer(pcm_payload, ts);
+ if (res != OK) {
+ LOGE("Failed to queue %d byte audio track buffer with media"
+ " PTS %lld. (res = %d)", decoded_amt, ts, res);
+ } else {
+ last_queued_pts_valid_ = true;
+ last_queued_pts_ = ts;
+ }
+ }
+
+ } else {
+ LOGE("No renderer, dropping audio payload.");
+ }
+ }
+}
+
+void AAH_DecoderPump::stopAndCleanupRenderer() {
+ if (NULL == renderer_) {
+ return;
+ }
+
+ renderer_->stop();
+ delete renderer_;
+ renderer_ = NULL;
+}
+
+void AAH_DecoderPump::setRenderTSTransform(const LinearTransform& trans) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (last_ts_transform_valid_ && !memcmp(&trans,
+ &last_ts_transform_,
+ sizeof(trans))) {
+ return;
+ }
+
+ last_ts_transform_ = trans;
+ last_ts_transform_valid_ = true;
+
+ if (NULL != renderer_) {
+ status_t res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ LOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ }
+ }
+}
+
+void AAH_DecoderPump::setRenderVolume(uint8_t volume) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (volume == last_volume_) {
+ return;
+ }
+
+ last_volume_ = volume;
+ if (renderer_ != NULL) {
+ float volume = static_cast<float>(last_volume_) / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ LOGW("%s: setVolume failed", __FUNCTION__);
+ }
+ }
+}
+
+// isAboutToUnderflow is something of a hack used to figure out when it might be
+// time to give up on trying to fill in a gap in the RTP sequence and simply
+// move on with a discontinuity. If we had perfect knowledge of when we were
+// going to underflow, it would not be a hack, but unfortunately we do not.
+// Right now, we just take the PTS of the last sample queued, and check to see
+// if its presentation time is within kAboutToUnderflowThreshold from now. If
+// it is, then we say that we are about to underflow. This decision is based on
+// two (possibly invalid) assumptions.
+//
+// 1) The transmitter is leading the clock by more than
+// kAboutToUnderflowThreshold.
+// 2) The delta between the PTS of the last sample queued and the next sample
+// is less than the transmitter's clock lead amount.
+//
+// Right now, the default transmitter lead time is 1 second, which is a pretty
+// large number and greater than the 50mSec that kAboutToUnderflowThreshold is
+// currently set to. This should satisfy assumption #1 for now, but changes to
+// the transmitter clock lead time could effect this.
+//
+// For non-sparse streams with a homogeneous sample rate (the vast majority of
+// streams in the world), the delta between any two adjacent PTSs will always be
+// the homogeneous sample period. It is very uncommon to see a sample period
+// greater than the 1 second clock lead we are currently using, and you
+// certainly will not see it in an MP3 file which should satisfy assumption #2.
+// Sparse audio streams (where no audio is transmitted for long periods of
+// silence) and extremely low framerate video stream (like an MPEG-2 slideshow
+// or the video stream for a pay TV audio channel) are examples of streams which
+// might violate assumption #2.
+bool AAH_DecoderPump::isAboutToUnderflow(int64_t threshold) {
+ Mutex::Autolock lock(&render_lock_);
+
+ // If we have never queued anything to the decoder, we really don't know if
+ // we are going to underflow or not.
+ if (!last_queued_pts_valid_ || !last_ts_transform_valid_) {
+ return false;
+ }
+
+ // Don't have access to Common Time? If so, then things are Very Bad
+ // elsewhere in the system; it pretty much does not matter what we do here.
+ // Since we cannot really tell if we are about to underflow or not, its
+ // probably best to assume that we are not and proceed accordingly.
+ int64_t tt_now;
+ if (OK != CCHelper::getCommonTime(&tt_now)) {
+ return false;
+ }
+
+ // Transform from media time to common time.
+ int64_t last_queued_pts_tt;
+ if (!last_ts_transform_.doForwardTransform(last_queued_pts_,
+ &last_queued_pts_tt)) {
+ return false;
+ }
+
+ // Check to see if we are underflowing.
+ return ((tt_now + threshold - last_queued_pts_tt) > 0);
+}
+
+void* AAH_DecoderPump::workThread() {
+ // No need to lock when accessing decoder_ from the thread. The
+ // implementation of init and shutdown ensure that other threads never touch
+ // decoder_ while the work thread is running.
+ CHECK(decoder_ != NULL);
+ CHECK(format_ != NULL);
+
+ // Start the decoder and note its result code. If something goes horribly
+ // wrong, callers of queueForDecode and getOutput will be able to detect
+ // that the thread encountered a fatal error and shut down by examining
+ // thread_status_.
+ thread_status_ = decoder_->start(format_.get());
+ if (OK != thread_status_) {
+ LOGE("AAH_DecoderPump's work thread failed to start decoder (res = %d)",
+ thread_status_);
+ return NULL;
+ }
+
+ while (!thread_->exitPending()) {
+ status_t res;
+ MediaBuffer* bufOut = NULL;
+
+ res = decoder_->read(&bufOut);
+
+ if (res == INFO_FORMAT_CHANGED) {
+ // Format has changed. Destroy our current renderer so that a new
+ // one can be created during queueToRenderer with the proper format.
+ //
+ // TODO : In order to transition seamlessly, we should change this
+ // to put the old renderer in a queue to play out completely before
+ // we destroy it. We can still create a new renderer, the timed
+ // nature of the renderer should ensure a seamless splice.
+ stopAndCleanupRenderer();
+ res = OK;
+ }
+
+ // Any error aside from INFO_FORMAT_CHANGED is considered to be fatal
+ // and will result in shutdown of the decoder pump thread.
+ if (res != OK) {
+ LOGE("%s: Failed to decode data (res = %d)",
+ __PRETTY_FUNCTION__, res);
+ CHECK(NULL == bufOut);
+ thread_status_ = res;
+ break;
+ }
+
+ if (NULL == bufOut) {
+ LOGW("%s: Successful decode, but no buffer produced",
+ __PRETTY_FUNCTION__);
+ continue;
+ }
+
+ queueToRenderer(bufOut);
+ bufOut->release();
+ }
+
+ decoder_->stop();
+ stopAndCleanupRenderer();
+
+ return NULL;
+}
+
+status_t AAH_DecoderPump::init(sp<MetaData> params) {
+ Mutex::Autolock lock(&init_lock_);
+
+ if (decoder_ != NULL) {
+ // already inited
+ return OK;
+ }
+
+ if (params == NULL) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeyChannelCount, &format_channels_)) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeySampleRate, &format_sample_rate_)) {
+ return BAD_VALUE;
+ }
+
+ CHECK(OK == thread_status_);
+ CHECK(decoder_ == NULL);
+
+ status_t ret_val = UNKNOWN_ERROR;
+
+ // Cache the format and attempt to create the decoder.
+ format_ = params;
+ decoder_ = OMXCodec::Create(
+ omx_.interface(), // IOMX Handle
+ format_, // Metadata for substream (indicates codec)
+ false, // Make a decoder, not an encoder
+ sp<MediaSource>(this)); // We will be the source for this codec.
+
+ if (decoder_ == NULL) {
+ LOGE("Failed to allocate decoder in %s", __PRETTY_FUNCTION__);
+ goto bailout;
+ }
+
+ // Fire up the pump thread. It will take care of starting and stopping the
+ // decoder.
+ ret_val = thread_->run("aah_decode_pump", ANDROID_PRIORITY_AUDIO);
+ if (OK != ret_val) {
+ LOGE("Failed to start work thread in %s (res = %d)",
+ __PRETTY_FUNCTION__, ret_val);
+ goto bailout;
+ }
+
+bailout:
+ if (OK != ret_val) {
+ decoder_ = NULL;
+ format_ = NULL;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::shutdown() {
+ Mutex::Autolock lock(&init_lock_);
+ return shutdown_l();
+}
+
+status_t AAH_DecoderPump::shutdown_l() {
+ thread_->requestExit();
+ thread_cond_.signal();
+ thread_->requestExitAndWait();
+
+ MBQueue::iterator I;
+ for (I = in_queue_.begin(); I != in_queue_.end(); ++I) {
+ (*I)->release();
+ }
+ in_queue_.clear();
+
+ last_queued_pts_valid_ = false;
+ last_ts_transform_valid_ = false;
+ last_volume_ = 0xFF;
+
+ decoder_ = NULL;
+ format_ = NULL;
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::read(MediaBuffer **buffer,
+ const ReadOptions *options) {
+ if (!buffer) {
+ return BAD_VALUE;
+ }
+
+ *buffer = NULL;
+
+ // While its not time to shut down, and we have no data to process, wait.
+ AutoMutex lock(&thread_lock_);
+ while (!thread_->exitPending() && in_queue_.empty())
+ thread_cond_.wait(thread_lock_);
+
+ // At this point, if its not time to shutdown then we must have something to
+ // process. Go ahead and pop the front of the queue for processing.
+ if (!thread_->exitPending()) {
+ CHECK(!in_queue_.empty());
+
+ *buffer = *(in_queue_.begin());
+ in_queue_.erase(in_queue_.begin());
+ }
+
+ // If we managed to get a buffer, then everything must be OK. If not, then
+ // we must be shutting down.
+ return (NULL == *buffer) ? INVALID_OPERATION : OK;
+}
+
+AAH_DecoderPump::ThreadWrapper::ThreadWrapper(AAH_DecoderPump* owner)
+ : Thread(false /* canCallJava*/ )
+ , owner_(owner) {
+}
+
+bool AAH_DecoderPump::ThreadWrapper::threadLoop() {
+ CHECK(NULL != owner_);
+ owner_->workThread();
+ return false;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_decoder_pump.h b/media/libaah_rtp/aah_decoder_pump.h
new file mode 100644
index 0000000..c9b35cd
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.h
@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2011 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 __DECODER_PUMP_H__
+#define __DECODER_PUMP_H__
+
+#include <pthread.h>
+
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/List.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class MetaData;
+class OMXClient;
+class TimedAudioTrack;
+
+class AAH_DecoderPump : public MediaSource {
+ public:
+ explicit AAH_DecoderPump(OMXClient& omx);
+ status_t initCheck();
+
+ status_t queueForDecode(MediaBuffer* buf);
+
+ status_t init(sp<MetaData> params);
+ status_t shutdown();
+
+ void setRenderTSTransform(const LinearTransform& trans);
+ void setRenderVolume(uint8_t volume);
+ bool isAboutToUnderflow(int64_t threshold);
+
+ // MediaSource methods
+ virtual status_t start(MetaData *params) { return OK; }
+ virtual sp<MetaData> getFormat() { return format_; }
+ virtual status_t stop() { return OK; }
+ virtual status_t read(MediaBuffer **buffer,
+ const ReadOptions *options);
+
+ protected:
+ virtual ~AAH_DecoderPump();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_DecoderPump;
+ explicit ThreadWrapper(AAH_DecoderPump* owner);
+
+ private:
+ virtual bool threadLoop();
+ AAH_DecoderPump* owner_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+ void* workThread();
+ virtual status_t shutdown_l();
+ void queueToRenderer(MediaBuffer* decoded_sample);
+ void stopAndCleanupRenderer();
+
+ sp<MetaData> format_;
+ int32_t format_channels_;
+ int32_t format_sample_rate_;
+
+ sp<MediaSource> decoder_;
+ OMXClient& omx_;
+ Mutex init_lock_;
+
+ sp<ThreadWrapper> thread_;
+ Condition thread_cond_;
+ Mutex thread_lock_;
+ status_t thread_status_;
+
+ Mutex render_lock_;
+ TimedAudioTrack* renderer_;
+ bool last_queued_pts_valid_;
+ int64_t last_queued_pts_;
+ bool last_ts_transform_valid_;
+ LinearTransform last_ts_transform_;
+ uint8_t last_volume_;
+
+ // protected by the thread_lock_
+ typedef List<MediaBuffer*> MBQueue;
+ MBQueue in_queue_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_DecoderPump);
+};
+
+} // namespace android
+#endif // __DECODER_PUMP_H__
diff --git a/media/libaah_rtp/aah_rx_player.cpp b/media/libaah_rtp/aah_rx_player.cpp
new file mode 100644
index 0000000..1fe836c
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.cpp
@@ -0,0 +1,289 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <aah_timesrv/cc_helper.h>
+#include <binder/IServiceManager.h>
+#include <media/MediaPlayerInterface.h>
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRTPRingBufferSize = 1 << 10;
+
+sp<MediaPlayerBase> createAAH_RXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_RXPlayer();
+ return ret;
+}
+
+AAH_RXPlayer::AAH_RXPlayer()
+ : ring_buffer_(kRTPRingBufferSize)
+ , substreams_(NULL) {
+ thread_wrapper_ = new ThreadWrapper(*this);
+
+ is_playing_ = false;
+ multicast_joined_ = false;
+ transmitter_known_ = false;
+ current_epoch_known_ = false;
+ data_source_set_ = false;
+ sock_fd_ = -1;
+
+ substreams_.setCapacity(4);
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ memset(&transmitter_addr_, 0, sizeof(transmitter_addr_));
+
+ fetchAudioFlinger();
+}
+
+AAH_RXPlayer::~AAH_RXPlayer() {
+ reset_l();
+ CHECK(substreams_.size() == 0);
+ omx_.disconnect();
+}
+
+status_t AAH_RXPlayer::initCheck() {
+ if (thread_wrapper_ == NULL) {
+ LOGE("Failed to allocate thread wrapper!");
+ return NO_MEMORY;
+ }
+
+ if (!ring_buffer_.initCheck()) {
+ LOGE("Failed to allocate reassembly ring buffer!");
+ return NO_MEMORY;
+ }
+
+ // Check for the presense of the A@H common time service by attempting to
+ // query for CommonTime's frequency. If we get an error back, we cannot
+ // talk to the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = CCHelper::getCommonFreq(&freq);
+ if (OK != res) {
+ LOGE("Failed to connect to common time service!");
+ return res;
+ }
+
+ return omx_.connect();
+}
+
+status_t AAH_RXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ AutoMutex api_lock(&api_lock_);
+ uint32_t a, b, c, d;
+ uint16_t port;
+
+ if (data_source_set_) {
+ return INVALID_OPERATION;
+ }
+
+ if (NULL == url) {
+ return BAD_VALUE;
+ }
+
+ if (5 != sscanf(url, "%*[^:/]://%u.%u.%u.%u:%hu", &a, &b, &c, &d, &port)) {
+ LOGE("Failed to parse URL \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ if ((a > 255) || (b > 255) || (c > 255) || (d > 255) || (port == 0)) {
+ LOGE("Bad multicast address \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ LOGI("setDataSource :: %u.%u.%u.%u:%hu", a, b, c, d, port);
+
+ a = (a << 24) | (b << 16) | (c << 8) | d;
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ listen_addr_.sin_family = AF_INET;
+ listen_addr_.sin_port = htons(port);
+ listen_addr_.sin_addr.s_addr = htonl(a);
+ data_source_set_ = true;
+
+ return OK;
+}
+
+status_t AAH_RXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_RXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepare() {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepareAsync() {
+ sendEvent(MEDIA_PREPARED);
+ return OK;
+}
+
+status_t AAH_RXPlayer::start() {
+ AutoMutex api_lock(&api_lock_);
+
+ if (is_playing_) {
+ return OK;
+ }
+
+ status_t res = startWorkThread();
+ is_playing_ = (res == OK);
+ return res;
+}
+
+status_t AAH_RXPlayer::stop() {
+ return pause();
+}
+
+status_t AAH_RXPlayer::pause() {
+ AutoMutex api_lock(&api_lock_);
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ is_playing_ = false;
+ return OK;
+}
+
+bool AAH_RXPlayer::isPlaying() {
+ AutoMutex api_lock(&api_lock_);
+ return is_playing_;
+}
+
+status_t AAH_RXPlayer::seekTo(int msec) {
+ sendEvent(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_RXPlayer::getCurrentPosition(int *msec) {
+ if (NULL != msec) {
+ *msec = 0;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::getDuration(int *msec) {
+ if (NULL != msec) {
+ *msec = 1;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::reset() {
+ AutoMutex api_lock(&api_lock_);
+ reset_l();
+ return OK;
+}
+
+void AAH_RXPlayer::reset_l() {
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ CHECK(!multicast_joined_);
+ is_playing_ = false;
+ data_source_set_ = false;
+ transmitter_known_ = false;
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+}
+
+status_t AAH_RXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_RXPlayer::playerType() {
+ return AAH_RX_PLAYER;
+}
+
+status_t AAH_RXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ if (!reply) {
+ return BAD_VALUE;
+ }
+
+ int32_t magic;
+ status_t err = request.readInt32(&magic);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ if (magic != 0x12345) {
+ reply->writeInt32(BAD_VALUE);
+ return OK;
+ }
+
+ int32_t methodID;
+ err = request.readInt32(&methodID);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ switch (methodID) {
+ // Get Volume
+ case INVOKE_GET_MASTER_VOLUME: {
+ if (audio_flinger_ != NULL) {
+ reply->writeInt32(OK);
+ reply->writeFloat(audio_flinger_->masterVolume());
+ } else {
+ reply->writeInt32(UNKNOWN_ERROR);
+ }
+ } break;
+
+ // Set Volume
+ case INVOKE_SET_MASTER_VOLUME: {
+ float targetVol = request.readFloat();
+ reply->writeInt32(audio_flinger_->setMasterVolume(targetVol));
+ } break;
+
+ default: return BAD_VALUE;
+ }
+
+ return OK;
+}
+
+void AAH_RXPlayer::fetchAudioFlinger() {
+ if (audio_flinger_ == NULL) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ binder = sm->getService(String16("media.audio_flinger"));
+
+ if (binder == NULL) {
+ LOGW("AAH_RXPlayer failed to fetch handle to audio flinger."
+ " Master volume control will not be possible.");
+ }
+
+ audio_flinger_ = interface_cast<IAudioFlinger>(binder);
+ }
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player.h b/media/libaah_rtp/aah_rx_player.h
new file mode 100644
index 0000000..44aedc3
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.h
@@ -0,0 +1,289 @@
+/*
+ * Copyright (C) 2011 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 __AAH_RX_PLAYER_H__
+#define __AAH_RX_PLAYER_H__
+
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <netinet/in.h>
+#include <utils/KeyedVector.h>
+#include <utils/LinearTransform.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+#include "pipe_event.h"
+
+namespace android {
+
+class AAH_RXPlayer : public MediaPlayerInterface {
+ public:
+ AAH_RXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+
+ protected:
+ virtual ~AAH_RXPlayer();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_RXPlayer;
+ explicit ThreadWrapper(AAH_RXPlayer& player)
+ : Thread(false /* canCallJava */ )
+ , player_(player) { }
+
+ virtual bool threadLoop() { return player_.threadLoop(); }
+
+ private:
+ AAH_RXPlayer& player_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+#pragma pack(push, 1)
+ struct PacketBuffer {
+ ssize_t length_;
+ uint8_t data_[0];
+
+ // TODO : consider changing this to be some form of ring buffer or free
+ // pool system instead of just using the heap in order to avoid heap
+ // fragmentation.
+ static PacketBuffer* allocate(ssize_t length);
+ static void destroy(PacketBuffer* pb);
+
+ private:
+ // Force people to use allocate/destroy instead of new/delete.
+ PacketBuffer() { }
+ ~PacketBuffer() { }
+ };
+
+ struct RetransRequest {
+ uint32_t magic_;
+ uint32_t mcast_ip_;
+ uint16_t mcast_port_;
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+#pragma pack(pop)
+
+ enum GapStatus {
+ kGS_NoGap = 0,
+ kGS_NormalGap,
+ kGS_FastStartGap,
+ };
+
+ struct SeqNoGap {
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+
+ class RXRingBuffer {
+ public:
+ explicit RXRingBuffer(uint32_t capacity);
+ ~RXRingBuffer();
+
+ bool initCheck() const { return (ring_ != NULL); }
+ void reset();
+
+ // Push a packet buffer with a given sequence number into the ring
+ // buffer. pushBuffer will always consume the buffer pushed to it,
+ // either destroying it because it was a duplicate or overflow, or
+ // holding on to it in the ring. Callers should not hold any references
+ // to PacketBuffers after they have been pushed to the ring. Returns
+ // false in the case of a serious error (such as ring overflow).
+ // Callers should consider resetting the pipeline entirely in the event
+ // of a serious error.
+ bool pushBuffer(PacketBuffer* buf, uint16_t seq);
+
+ // Fetch the next buffer in the RTP sequence. Returns NULL if there is
+ // no buffer to fetch. If a non-NULL PacketBuffer is returned,
+ // is_discon will be set to indicate whether or not this PacketBuffer is
+ // discontiuous with any previously returned packet buffers. Packet
+ // buffers returned by fetchBuffer are the caller's responsibility; they
+ // must be certain to destroy the buffers when they are done.
+ PacketBuffer* fetchBuffer(bool* is_discon);
+
+ // Returns true and fills out the gap structure if the read pointer of
+ // the ring buffer is currently pointing to a gap which would stall a
+ // fetchBuffer operation. Returns false if the read pointer is not
+ // pointing to a gap in the sequence currently.
+ GapStatus fetchCurrentGap(SeqNoGap* gap);
+
+ // Causes the read pointer to skip over any portion of a gap indicated
+ // by nak. If nak is NULL, any gap currently blocking the read pointer
+ // will be completely skipped. If any portion of a gap is skipped, the
+ // next successful read from fetch buffer will indicate a discontinuity.
+ void processNAK(SeqNoGap* nak = NULL);
+
+ // Compute the number of milliseconds until the inactivity timer for
+ // this RTP stream. Returns -1 if there is no active timeout, or 0 if
+ // the system has already timed out.
+ int computeInactivityTimeout();
+
+ private:
+ Mutex lock_;
+ PacketBuffer** ring_;
+ uint32_t capacity_;
+ uint32_t rd_;
+ uint32_t wr_;
+
+ uint16_t rd_seq_;
+ bool rd_seq_known_;
+ bool waiting_for_fast_start_;
+ bool fetched_first_packet_;
+
+ uint64_t rtp_activity_timeout_;
+ bool rtp_activity_timeout_valid_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(RXRingBuffer);
+ };
+
+ class Substream : public virtual RefBase {
+ public:
+ Substream(uint32_t ssrc, OMXClient& omx);
+
+ void cleanupBufferInProgress();
+ void shutdown();
+ void processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower);
+ void processPayloadCont (uint8_t* buf,
+ uint32_t amt);
+ void processCompletedBuffer();
+ void processTSTransform(const LinearTransform& trans);
+
+ bool isAboutToUnderflow();
+ uint32_t getSSRC() const { return ssrc_; }
+ uint16_t getProgramID() const { return (ssrc_ >> 5) & 0x1F; }
+
+ protected:
+ virtual ~Substream() {
+ shutdown();
+ }
+
+ private:
+ void cleanupDecoder();
+ bool setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type);
+
+ uint32_t ssrc_;
+ bool waiting_for_rap_;
+
+ bool substream_details_known_;
+ uint8_t substream_type_;
+ uint8_t codec_type_;
+ sp<MetaData> substream_meta_;
+
+ MediaBuffer* buffer_in_progress_;
+ uint32_t expected_buffer_size_;
+ uint32_t buffer_filled_;
+
+ sp<AAH_DecoderPump> decoder_;
+
+ static int64_t kAboutToUnderflowThreshold;
+
+ DISALLOW_EVIL_CONSTRUCTORS(Substream);
+ };
+
+ typedef DefaultKeyedVector< uint32_t, sp<Substream> > SubstreamVec;
+
+ status_t startWorkThread();
+ void stopWorkThread();
+ virtual bool threadLoop();
+ bool setupSocket();
+ void cleanupSocket();
+ void resetPipeline();
+ void reset_l();
+ bool processRX(PacketBuffer* pb);
+ void processRingBuffer();
+ void processCommandPacket(PacketBuffer* pb);
+ bool processGaps();
+ int computeNextGapRetransmitTimeout();
+ void fetchAudioFlinger();
+
+ PipeEvent wakeup_work_thread_evt_;
+ sp<ThreadWrapper> thread_wrapper_;
+ Mutex api_lock_;
+ bool is_playing_;
+ bool data_source_set_;
+
+ struct sockaddr_in listen_addr_;
+ int sock_fd_;
+ bool multicast_joined_;
+
+ struct sockaddr_in transmitter_addr_;
+ bool transmitter_known_;
+
+ uint32_t current_epoch_;
+ bool current_epoch_known_;
+
+ SeqNoGap current_gap_;
+ GapStatus current_gap_status_;
+ uint64_t next_retrans_req_time_;
+
+ RXRingBuffer ring_buffer_;
+ SubstreamVec substreams_;
+ OMXClient omx_;
+
+ // Connection to audio flinger used to hack a path to setMasterVolume.
+ sp<IAudioFlinger> audio_flinger_;
+
+ static const uint32_t kRTPRingBufferSize;
+ static const uint32_t kRetransRequestMagic;
+ static const uint32_t kFastStartRequestMagic;
+ static const uint32_t kRetransNAKMagic;
+ static const uint32_t kGapRerequestTimeoutUSec;
+ static const uint32_t kFastStartTimeoutUSec;
+ static const uint32_t kRTPActivityTimeoutUSec;
+
+ static const uint32_t INVOKE_GET_MASTER_VOLUME = 3;
+ static const uint32_t INVOKE_SET_MASTER_VOLUME = 4;
+
+ static uint64_t monotonicUSecNow();
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_RXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_RX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_rx_player_core.cpp b/media/libaah_rtp/aah_rx_player_core.cpp
new file mode 100644
index 0000000..a86ea17
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_core.cpp
@@ -0,0 +1,791 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <fcntl.h>
+#include <poll.h>
+#include <sys/socket.h>
+#include <time.h>
+#include <utils/misc.h>
+
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRetransRequestMagic =
+ FOURCC('T','r','e','q');
+const uint32_t AAH_RXPlayer::kRetransNAKMagic =
+ FOURCC('T','n','a','k');
+const uint32_t AAH_RXPlayer::kFastStartRequestMagic =
+ FOURCC('T','f','s','t');
+const uint32_t AAH_RXPlayer::kGapRerequestTimeoutUSec = 75000;
+const uint32_t AAH_RXPlayer::kFastStartTimeoutUSec = 800000;
+const uint32_t AAH_RXPlayer::kRTPActivityTimeoutUSec = 10000000;
+
+static inline int16_t fetchInt16(uint8_t* data) {
+ return static_cast<int16_t>(U16_AT(data));
+}
+
+static inline int32_t fetchInt32(uint8_t* data) {
+ return static_cast<int32_t>(U32_AT(data));
+}
+
+static inline int64_t fetchInt64(uint8_t* data) {
+ return static_cast<int64_t>(U64_AT(data));
+}
+
+uint64_t AAH_RXPlayer::monotonicUSecNow() {
+ struct timespec now;
+ int res = clock_gettime(CLOCK_MONOTONIC, &now);
+ CHECK(res >= 0);
+
+ uint64_t ret = static_cast<uint64_t>(now.tv_sec) * 1000000;
+ ret += now.tv_nsec / 1000;
+
+ return ret;
+}
+
+status_t AAH_RXPlayer::startWorkThread() {
+ status_t res;
+ stopWorkThread();
+ res = thread_wrapper_->run("TRX_Player", PRIORITY_AUDIO);
+
+ if (res != OK) {
+ LOGE("Failed to start work thread (res = %d)", res);
+ }
+
+ return res;
+}
+
+void AAH_RXPlayer::stopWorkThread() {
+ thread_wrapper_->requestExit(); // set the exit pending flag
+ wakeup_work_thread_evt_.setEvent();
+
+ status_t res;
+ res = thread_wrapper_->requestExitAndWait(); // block until thread exit.
+ if (res != OK) {
+ LOGE("Failed to stop work thread (res = %d)", res);
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+}
+
+void AAH_RXPlayer::cleanupSocket() {
+ if (sock_fd_ >= 0) {
+ if (multicast_joined_) {
+ int res;
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_DROP_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ LOGW("Failed to leave multicast group. (%d, %d)", res, errno);
+ }
+ multicast_joined_ = false;
+ }
+
+ close(sock_fd_);
+ sock_fd_ = -1;
+ }
+
+ resetPipeline();
+}
+
+void AAH_RXPlayer::resetPipeline() {
+ ring_buffer_.reset();
+
+ // Explicitly shudown all of the active substreams, then call clear out the
+ // collection. Failure to clear out a substream can result in its decoder
+ // holding a reference to itself and therefor not going away when the
+ // collection is cleared.
+ for (size_t i = 0; i < substreams_.size(); ++i)
+ substreams_.valueAt(i)->shutdown();
+
+ substreams_.clear();
+
+ current_gap_status_ = kGS_NoGap;
+}
+
+bool AAH_RXPlayer::setupSocket() {
+ long flags;
+ int res, buf_size;
+ socklen_t opt_size;
+
+ cleanupSocket();
+ CHECK(sock_fd_ < 0);
+
+ // Make the socket
+ sock_fd_ = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sock_fd_ < 0) {
+ LOGE("Failed to create listen socket (errno %d)", errno);
+ goto bailout;
+ }
+
+ // Set non-blocking operation
+ flags = fcntl(sock_fd_, F_GETFL);
+ res = fcntl(sock_fd_, F_SETFL, flags | O_NONBLOCK);
+ if (res < 0) {
+ LOGE("Failed to set socket (%d) to non-blocking mode (errno %d)",
+ sock_fd_, errno);
+ goto bailout;
+ }
+
+ // Bind to our port
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ bind_addr.sin_addr.s_addr = INADDR_ANY;
+ bind_addr.sin_port = listen_addr_.sin_port;
+ res = bind(sock_fd_,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr));
+ if (res < 0) {
+ uint32_t a = ntohl(bind_addr.sin_addr.s_addr);
+ uint16_t p = ntohs(bind_addr.sin_port);
+ LOGE("Failed to bind socket (%d) to %d.%d.%d.%d:%hd. (errno %d)",
+ sock_fd_,
+ (a >> 24) & 0xFF,
+ (a >> 16) & 0xFF,
+ (a >> 8) & 0xFF,
+ (a ) & 0xFF,
+ p,
+ errno);
+
+ goto bailout;
+ }
+
+ buf_size = 1 << 16; // 64k
+ res = setsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, sizeof(buf_size));
+ if (res < 0) {
+ LOGW("Failed to increase socket buffer size to %d. (errno %d)",
+ buf_size, errno);
+ }
+
+ buf_size = 0;
+ opt_size = sizeof(buf_size);
+ res = getsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, &opt_size);
+ if (res < 0) {
+ LOGW("Failed to increase socket buffer size to %d. (errno %d)",
+ buf_size, errno);
+ } else {
+ LOGI("RX socket buffer size is now %d bytes", buf_size);
+ }
+
+ if (listen_addr_.sin_addr.s_addr) {
+ // Join the multicast group and we should be good to go.
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_ADD_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ LOGE("Failed to join multicast group. (errno %d)", errno);
+ goto bailout;
+ }
+ multicast_joined_ = true;
+ }
+
+ return true;
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::threadLoop() {
+ struct pollfd poll_fds[2];
+ bool process_more_right_now = false;
+
+ if (!setupSocket()) {
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+
+ while (!thread_wrapper_->exitPending()) {
+ // Step 1: Wait until there is something to do.
+ int gap_timeout = computeNextGapRetransmitTimeout();
+ int ring_timeout = ring_buffer_.computeInactivityTimeout();
+ int timeout = -1;
+
+ if (!ring_timeout) {
+ LOGW("RTP inactivity timeout reached, resetting pipeline.");
+ resetPipeline();
+ timeout = gap_timeout;
+ } else {
+ if (gap_timeout < 0) {
+ timeout = ring_timeout;
+ } else if (ring_timeout < 0) {
+ timeout = gap_timeout;
+ } else {
+ timeout = (gap_timeout < ring_timeout) ? gap_timeout
+ : ring_timeout;
+ }
+ }
+
+ if ((0 != timeout) && (!process_more_right_now)) {
+ // Set up the events to wait on. Start with the wakeup pipe.
+ memset(&poll_fds, 0, sizeof(poll_fds));
+ poll_fds[0].fd = wakeup_work_thread_evt_.getWakeupHandle();
+ poll_fds[0].events = POLLIN;
+
+ // Add the RX socket.
+ poll_fds[1].fd = sock_fd_;
+ poll_fds[1].events = POLLIN;
+
+ // Wait for something interesing to happen.
+ int poll_res = poll(poll_fds, NELEM(poll_fds), timeout);
+ if (poll_res < 0) {
+ LOGE("Fatal error (%d,%d) while waiting on events",
+ poll_res, errno);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+
+ if (thread_wrapper_->exitPending()) {
+ break;
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+ process_more_right_now = false;
+
+ // Step 2: Do we have data waiting in the socket? If so, drain the
+ // socket moving valid RTP information into the ring buffer to be
+ // processed.
+ if (poll_fds[1].revents) {
+ struct sockaddr_in from;
+ socklen_t from_len;
+
+ ssize_t res = 0;
+ while (!thread_wrapper_->exitPending()) {
+ // Check the size of any pending packet.
+ res = recv(sock_fd_, NULL, 0, MSG_PEEK | MSG_TRUNC);
+
+ // Error?
+ if (res < 0) {
+ // If the error is anything other than would block,
+ // something has gone very wrong.
+ if ((errno != EAGAIN) && (errno != EWOULDBLOCK)) {
+ LOGE("Fatal socket error during recvfrom (%d, %d)",
+ (int)res, errno);
+ goto bailout;
+ }
+
+ // Socket is out of data, just break out of processing and
+ // wait for more.
+ break;
+ }
+
+ // Allocate a payload.
+ PacketBuffer* pb = PacketBuffer::allocate(res);
+ if (NULL == pb) {
+ LOGE("Fatal error, failed to allocate packet buffer of"
+ " length %u", static_cast<uint32_t>(res));
+ goto bailout;
+ }
+
+ // Fetch the data.
+ from_len = sizeof(from);
+ res = recvfrom(sock_fd_, pb->data_, pb->length_, 0,
+ reinterpret_cast<struct sockaddr*>(&from),
+ &from_len);
+ if (res != pb->length_) {
+ LOGE("Fatal error, fetched packet length (%d) does not"
+ " match peeked packet length (%u). This should never"
+ " happen. (errno = %d)",
+ static_cast<int>(res),
+ static_cast<uint32_t>(pb->length_),
+ errno);
+ }
+
+ bool drop_packet = false;
+ if (transmitter_known_) {
+ if (from.sin_addr.s_addr !=
+ transmitter_addr_.sin_addr.s_addr) {
+ uint32_t a = ntohl(from.sin_addr.s_addr);
+ uint16_t p = ntohs(from.sin_port);
+ LOGV("Dropping packet from unknown transmitter"
+ " %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+
+ drop_packet = true;
+ } else {
+ transmitter_addr_.sin_port = from.sin_port;
+ }
+ } else {
+ memcpy(&transmitter_addr_, &from, sizeof(from));
+ transmitter_known_ = true;
+ }
+
+ if (!drop_packet) {
+ bool serious_error = !processRX(pb);
+
+ if (serious_error) {
+ // Something went "seriously wrong". Currently, the
+ // only trigger for this should be a ring buffer
+ // overflow. The current failsafe behavior for when
+ // something goes seriously wrong is to just reset the
+ // pipeline. The system should behave as if this
+ // AAH_RXPlayer was just set up for the first time.
+ LOGE("Something just went seriously wrong with the"
+ " pipeline. Resetting.");
+ resetPipeline();
+ }
+ } else {
+ PacketBuffer::destroy(pb);
+ }
+ }
+ }
+
+ // Step 3: Process any data we mave have accumulated in the ring buffer
+ // so far.
+ if (!thread_wrapper_->exitPending()) {
+ processRingBuffer();
+ }
+
+ // Step 4: At this point in time, the ring buffer should either be
+ // empty, or stalled in front of a gap caused by some dropped packets.
+ // Check on the current gap situation and deal with it in an appropriate
+ // fashion. If processGaps returns true, it means that it has given up
+ // on a gap and that we should try to process some more data
+ // immediately.
+ if (!thread_wrapper_->exitPending()) {
+ process_more_right_now = processGaps();
+ }
+ }
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::processRX(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+ uint32_t nak_magic;
+ uint16_t seq_no;
+ uint32_t epoch;
+
+ // Every packet either starts with an RTP header which is at least 12 bytes
+ // long or is a retry NAK which is 14 bytes long. If there are fewer than
+ // 12 bytes here, this cannot be a proper RTP packet.
+ if (amt < 12) {
+ LOGV("Dropping packet, too short to contain RTP header (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ // Check to see if this is the special case of a NAK packet.
+ nak_magic = ntohl(*(reinterpret_cast<uint32_t*>(data)));
+ if (nak_magic == kRetransNAKMagic) {
+ // Looks like a NAK packet; make sure its long enough.
+
+ if (amt < static_cast<ssize_t>(sizeof(RetransRequest))) {
+ LOGV("Dropping packet, too short to contain NAK payload (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ SeqNoGap gap;
+ RetransRequest* rtr = reinterpret_cast<RetransRequest*>(data);
+ gap.start_seq_ = ntohs(rtr->start_seq_);
+ gap.end_seq_ = ntohs(rtr->end_seq_);
+
+ LOGV("Process NAK for gap at [%hu, %hu]", gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK(&gap);
+
+ return true;
+ }
+
+ // According to the TRTP spec, version should be 2, padding should be 0,
+ // extension should be 0 and CSRCCnt should be 0. If any of these tests
+ // fail, we chuck the packet.
+ if (data[0] != 0x80) {
+ LOGV("Dropping packet, bad V/P/X/CSRCCnt field (0x%02x)",
+ data[0]);
+ goto drop_packet;
+ }
+
+ // Check the payload type. For TRTP, it should always be 100.
+ if ((data[1] & 0x7F) != 100) {
+ LOGV("Dropping packet, bad payload type. (%u)",
+ data[1] & 0x7F);
+ goto drop_packet;
+ }
+
+ // Check whether the transmitter has begun a new epoch.
+ epoch = (U32_AT(data + 8) >> 10) & 0x3FFFFF;
+ if (current_epoch_known_) {
+ if (epoch != current_epoch_) {
+ LOGV("%s: new epoch %u", __PRETTY_FUNCTION__, epoch);
+ current_epoch_ = epoch;
+ resetPipeline();
+ }
+ } else {
+ current_epoch_ = epoch;
+ current_epoch_known_ = true;
+ }
+
+ // Extract the sequence number and hand the packet off to the ring buffer
+ // for dropped packet detection and later processing.
+ seq_no = U16_AT(data + 2);
+ return ring_buffer_.pushBuffer(pb, seq_no);
+
+drop_packet:
+ PacketBuffer::destroy(pb);
+ return true;
+}
+
+void AAH_RXPlayer::processRingBuffer() {
+ PacketBuffer* pb;
+ bool is_discon;
+ sp<Substream> substream;
+ LinearTransform trans;
+ bool foundTrans = false;
+
+ while (NULL != (pb = ring_buffer_.fetchBuffer(&is_discon))) {
+ if (is_discon) {
+ // Abort all partially assembled payloads.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ CHECK(substreams_.valueAt(i) != NULL);
+ substreams_.valueAt(i)->cleanupBufferInProgress();
+ }
+ }
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // Should not have any non-RTP packets in the ring buffer. RTP packets
+ // must be at least 12 bytes long.
+ CHECK(amt >= 12);
+
+ // Extract the marker bit and the SSRC field.
+ bool marker = (data[1] & 0x80) != 0;
+ uint32_t ssrc = U32_AT(data + 8);
+
+ // Is this the start of a new TRTP payload? If so, the marker bit
+ // should be set and there are some things we should be checking for.
+ if (marker) {
+ // TRTP headers need to have at least a byte for version, a byte for
+ // payload type and flags, and 4 bytes for length.
+ if (amt < 18) {
+ LOGV("Dropping packet, too short to contain TRTP header"
+ " (%u bytes)", static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ // Check the TRTP version and extract the payload type/flags.
+ uint8_t trtp_version = data[12];
+ uint8_t payload_type = (data[13] >> 4) & 0xF;
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ LOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ goto process_next_packet;
+ }
+
+ // Is there a timestamp transformation present on this packet? If
+ // so, extract it and pass it to the appropriate substreams.
+ if (trtp_flags & 0x02) {
+ ssize_t offset = 18 + ((trtp_flags & 0x01) ? 4 : 0);
+ if (amt < (offset + 24)) {
+ LOGV("Dropping packet, too short to contain TRTP Timestamp"
+ " Transformation (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ trans.a_zero = fetchInt64(data + offset);
+ trans.b_zero = fetchInt64(data + offset + 16);
+ trans.a_to_b_numer = static_cast<int32_t>(
+ fetchInt32 (data + offset + 8));
+ trans.a_to_b_denom = U32_AT(data + offset + 12);
+ foundTrans = true;
+
+ uint32_t program_id = (ssrc >> 5) & 0x1F;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ CHECK(iter != NULL);
+
+ if (iter->getProgramID() == program_id) {
+ iter->processTSTransform(trans);
+ }
+ }
+ }
+
+ // Is this a command packet? If so, its not necessarily associate
+ // with one particular substream. Just give it to the command
+ // packet handler and then move on.
+ if (4 == payload_type) {
+ processCommandPacket(pb);
+ goto process_next_packet;
+ }
+ }
+
+ // If we got to here, then we are a normal packet. Find (or allocate)
+ // the substream we belong to and send the packet off to be processed.
+ substream = substreams_.valueFor(ssrc);
+ if (substream == NULL) {
+ substream = new Substream(ssrc, omx_);
+ if (substream == NULL) {
+ LOGE("Failed to allocate substream for SSRC 0x%08x", ssrc);
+ goto process_next_packet;
+ }
+ substreams_.add(ssrc, substream);
+
+ if (foundTrans) {
+ substream->processTSTransform(trans);
+ }
+ }
+
+ CHECK(substream != NULL);
+
+ if (marker) {
+ // Start of a new TRTP payload for this substream. Extract the
+ // lower 32 bits of the timestamp and hand the buffer to the
+ // substream for processing.
+ uint32_t ts_lower = U32_AT(data + 4);
+ substream->processPayloadStart(data + 12, amt - 12, ts_lower);
+ } else {
+ // Continuation of an existing TRTP payload. Just hand it off to
+ // the substream for processing.
+ substream->processPayloadCont(data + 12, amt - 12);
+ }
+
+process_next_packet:
+ PacketBuffer::destroy(pb);
+ } // end of main processing while loop.
+}
+
+void AAH_RXPlayer::processCommandPacket(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // verify that this packet meets the minimum length of a command packet
+ if (amt < 20) {
+ return;
+ }
+
+ uint8_t trtp_version = data[12];
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ LOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ return;
+ }
+
+ // calculate the start of the command payload
+ ssize_t offset = 18;
+ if (trtp_flags & 0x01) {
+ // timestamp is present (4 bytes)
+ offset += 4;
+ }
+ if (trtp_flags & 0x02) {
+ // transform is present (24 bytes)
+ offset += 24;
+ }
+
+ // the packet must contain 2 bytes of command payload beyond the TRTP header
+ if (amt < offset + 2) {
+ return;
+ }
+
+ uint16_t command_id = U16_AT(data + offset);
+
+ switch (command_id) {
+ case TRTPControlPacket::kCommandNop:
+ break;
+
+ case TRTPControlPacket::kCommandEOS:
+ case TRTPControlPacket::kCommandFlush: {
+ uint16_t program_id = (U32_AT(data + 8) >> 5) & 0x1F;
+ LOGI("*** %s flushing program_id=%d",
+ __PRETTY_FUNCTION__, program_id);
+
+ Vector<uint32_t> substreams_to_remove;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ if (iter->getProgramID() == program_id) {
+ iter->shutdown();
+ substreams_to_remove.add(iter->getSSRC());
+ }
+ }
+
+ for (size_t i = 0; i < substreams_to_remove.size(); ++i) {
+ substreams_.removeItem(substreams_to_remove[i]);
+ }
+ } break;
+ }
+}
+
+bool AAH_RXPlayer::processGaps() {
+ // Deal with the current gap situation. Specifically...
+ //
+ // 1) If a new gap has shown up, send a retransmit request to the
+ // transmitter.
+ // 2) If a gap we were working on has had a packet in the middle or at
+ // the end filled in, send another retransmit request for the begining
+ // portion of the gap. TRTP was designed for LANs where packet
+ // re-ordering is very unlikely; so see the middle or end of a gap
+ // filled in before the begining is an almost certain indication that
+ // a retransmission packet was also dropped.
+ // 3) If we have been working on a gap for a while and it still has not
+ // been filled in, send another retransmit request.
+ // 4) If the are no more gaps in the ring, clear the current_gap_status_
+ // flag to indicate that all is well again.
+
+ // Start by fetching the active gap status.
+ SeqNoGap gap;
+ bool send_retransmit_request = false;
+ bool ret_val = false;
+ GapStatus gap_status;
+ if (kGS_NoGap != (gap_status = ring_buffer_.fetchCurrentGap(&gap))) {
+ // Note: checking for a change in the end sequence number should cover
+ // moving on to an entirely new gap for case #1 as well as resending the
+ // begining of a gap range for case #2.
+ send_retransmit_request = (kGS_NoGap == current_gap_status_) ||
+ (current_gap_.end_seq_ != gap.end_seq_);
+
+ // If this is the same gap we have been working on, and it has timed
+ // out, then check to see if our substreams are about to underflow. If
+ // so, instead of sending another retransmit request, just give up on
+ // this gap and move on.
+ if (!send_retransmit_request &&
+ (kGS_NoGap != current_gap_status_) &&
+ (0 == computeNextGapRetransmitTimeout())) {
+
+ // If out current gap is the fast-start gap, don't bother to skip it
+ // because substreams look like the are about to underflow.
+ if ((kGS_FastStartGap != gap_status) ||
+ (current_gap_.end_seq_ != gap.end_seq_)) {
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ if (substreams_.valueAt(i)->isAboutToUnderflow()) {
+ LOGV("About to underflow, giving up on gap [%hu, %hu]",
+ gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+ }
+ }
+
+ // Looks like no one is about to underflow. Just go ahead and send
+ // the request.
+ send_retransmit_request = true;
+ }
+ } else {
+ current_gap_status_ = kGS_NoGap;
+ }
+
+ if (send_retransmit_request) {
+ // If we have been working on a fast start, and it is still not filled
+ // in, even after the extended retransmit time out, give up and skip it.
+ // The system should fall back into its normal slow-start behavior.
+ if ((kGS_FastStartGap == current_gap_status_) &&
+ (current_gap_.end_seq_ == gap.end_seq_)) {
+ LOGV("Fast start is taking forever; giving up.");
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+
+ // Send the request.
+ RetransRequest req;
+ uint32_t magic = (kGS_FastStartGap == gap_status)
+ ? kFastStartRequestMagic
+ : kRetransRequestMagic;
+ req.magic_ = htonl(magic);
+ req.mcast_ip_ = listen_addr_.sin_addr.s_addr;
+ req.mcast_port_ = listen_addr_.sin_port;
+ req.start_seq_ = htons(gap.start_seq_);
+ req.end_seq_ = htons(gap.end_seq_);
+
+ {
+ uint32_t a = ntohl(transmitter_addr_.sin_addr.s_addr);
+ uint16_t p = ntohs(transmitter_addr_.sin_port);
+ LOGV("Sending to transmitter %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+ }
+
+ int res = sendto(sock_fd_, &req, sizeof(req), 0,
+ reinterpret_cast<struct sockaddr*>(&transmitter_addr_),
+ sizeof(transmitter_addr_));
+ if (res < 0) {
+ LOGE("Error when sending retransmit request (%d)", errno);
+ } else {
+ LOGV("%s request for range [%hu, %hu] sent",
+ (kGS_FastStartGap == gap_status) ? "Fast Start" : "Retransmit",
+ gap.start_seq_, gap.end_seq_);
+ }
+
+ // Update the current gap info.
+ current_gap_ = gap;
+ current_gap_status_ = gap_status;
+ next_retrans_req_time_ = monotonicUSecNow() +
+ ((kGS_FastStartGap == current_gap_status_)
+ ? kFastStartTimeoutUSec
+ : kGapRerequestTimeoutUSec);
+ }
+
+ return false;
+}
+
+// Compute when its time to send the next gap retransmission in milliseconds.
+// Returns < 0 for an infinite timeout (no gap) and 0 if its time to retransmit
+// right now.
+int AAH_RXPlayer::computeNextGapRetransmitTimeout() {
+ if (kGS_NoGap == current_gap_status_) {
+ return -1;
+ }
+
+ int64_t timeout_delta = next_retrans_req_time_ - monotonicUSecNow();
+
+ timeout_delta /= 1000;
+ if (timeout_delta <= 0) {
+ return 0;
+ }
+
+ return static_cast<uint32_t>(timeout_delta);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_ring_buffer.cpp b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
new file mode 100644
index 0000000..22467f7
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
@@ -0,0 +1,366 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+AAH_RXPlayer::RXRingBuffer::RXRingBuffer(uint32_t capacity) {
+ capacity_ = capacity;
+ rd_ = wr_ = 0;
+ ring_ = new PacketBuffer*[capacity];
+ memset(ring_, 0, sizeof(PacketBuffer*) * capacity);
+ reset();
+}
+
+AAH_RXPlayer::RXRingBuffer::~RXRingBuffer() {
+ reset();
+ delete[] ring_;
+}
+
+void AAH_RXPlayer::RXRingBuffer::reset() {
+ AutoMutex lock(&lock_);
+
+ if (NULL != ring_) {
+ while (rd_ != wr_) {
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ PacketBuffer::destroy(ring_[rd_]);
+ ring_[rd_] = NULL;
+ }
+ rd_ = (rd_ + 1) % capacity_;
+ }
+ }
+
+ rd_ = wr_ = 0;
+ rd_seq_known_ = false;
+ waiting_for_fast_start_ = true;
+ fetched_first_packet_ = false;
+ rtp_activity_timeout_valid_ = false;
+}
+
+bool AAH_RXPlayer::RXRingBuffer::pushBuffer(PacketBuffer* buf,
+ uint16_t seq) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != buf);
+
+ rtp_activity_timeout_valid_ = true;
+ rtp_activity_timeout_ = monotonicUSecNow() + kRTPActivityTimeoutUSec;
+
+ // If the ring buffer is totally reset (we have never received a single
+ // payload) then we don't know the rd sequence number and this should be
+ // simple. We just store the payload, advance the wr pointer and record the
+ // initial sequence number.
+ if (!rd_seq_known_) {
+ CHECK(rd_ == wr_);
+ CHECK(NULL == ring_[wr_]);
+ CHECK(wr_ < capacity_);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+ rd_seq_ = seq;
+ rd_seq_known_ = true;
+ return true;
+ }
+
+ // Compute the seqence number of this payload and of the write pointer,
+ // normalized around the read pointer. IOW - transform the payload seq no
+ // and the wr pointer seq no into a space where the rd pointer seq no is
+ // zero. This will define 4 cases we can consider...
+ //
+ // 1) norm_seq == norm_wr_seq
+ // This payload is contiguous with the last. All is good.
+ //
+ // 2) ((norm_seq < norm_wr_seq) && (norm_seq >= norm_rd_seq)
+ // aka ((norm_seq < norm_wr_seq) && (norm_seq >= 0)
+ // This payload is in the past, in the unprocessed region of the ring
+ // buffer. It is probably a retransmit intended to fill in a dropped
+ // payload; it may be a duplicate.
+ //
+ // 3) ((norm_seq - norm_wr_seq) & 0x8000) != 0
+ // This payload is in the past compared to the write pointer (or so very
+ // far in the future that it has wrapped the seq no space), but not in
+ // the unprocessed region of the ring buffer. This could be a duplicate
+ // retransmit; we just drop these payloads unless we are waiting for our
+ // first fast start packet. If we are waiting for fast start, than this
+ // packet is probably the first packet of the fast start retransmission.
+ // If it will fit in the buffer, back up the read pointer to its position
+ // and clear the fast start flag, otherwise just drop it.
+ //
+ // 4) ((norm_seq - norm_wr_seq) & 0x8000) == 0
+ // This payload which is ahead of the next write pointer. This indicates
+ // that we have missed some payloads and need to request a retransmit.
+ // If norm_seq >= (capacity - 1), then the gap is so large that it would
+ // overflow the ring buffer and we should probably start to panic.
+
+ uint16_t norm_wr_seq = ((wr_ + capacity_ - rd_) % capacity_);
+ uint16_t norm_seq = seq - rd_seq_;
+
+ // Check for overflow first.
+ if ((!(norm_seq & 0x8000)) && (norm_seq >= (capacity_ - 1))) {
+ LOGW("Ring buffer overflow; cap = %u, [rd, wr] = [%hu, %hu], seq = %hu",
+ capacity_, rd_seq_, norm_wr_seq + rd_seq_, seq);
+ PacketBuffer::destroy(buf);
+ return false;
+ }
+
+ // Check for case #1
+ if (norm_seq == norm_wr_seq) {
+ CHECK(wr_ < capacity_);
+ CHECK(NULL == ring_[wr_]);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+
+ CHECK(wr_ != rd_);
+ return true;
+ }
+
+ // Check case #2
+ uint32_t ring_pos = (rd_ + norm_seq) % capacity_;
+ if ((norm_seq < norm_wr_seq) && (!(norm_seq & 0x8000))) {
+ // Do we already have a payload for this slot? If so, then this looks
+ // like a duplicate retransmit. Just ignore it.
+ if (NULL != ring_[ring_pos]) {
+ LOGD("RXed duplicate retransmit, seq = %hu", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like we were missing this payload. Go ahead and store it.
+ ring_[ring_pos] = buf;
+ }
+
+ return true;
+ }
+
+ // Check case #3
+ if ((norm_seq - norm_wr_seq) & 0x8000) {
+ if (!waiting_for_fast_start_) {
+ LOGD("RXed duplicate retransmit from before rd pointer, seq = %hu",
+ seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like a fast start fill-in. Go ahead and store it, assuming
+ // that we can fit it in the buffer.
+ uint32_t implied_ring_size = static_cast<uint32_t>(norm_wr_seq)
+ + (rd_seq_ - seq);
+
+ if (implied_ring_size >= (capacity_ - 1)) {
+ LOGD("RXed what looks like a fast start packet (seq = %hu),"
+ " but packet is too far in the past to fit into the ring"
+ " buffer. Dropping.", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ ring_pos = (rd_ + capacity_ + seq - rd_seq_) % capacity_;
+ rd_seq_ = seq;
+ rd_ = ring_pos;
+ waiting_for_fast_start_ = false;
+
+ CHECK(ring_pos < capacity_);
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ }
+
+ }
+ return true;
+ }
+
+ // Must be in case #4 with no overflow. This packet fits in the current
+ // ring buffer, but is discontiuguous. Advance the write pointer leaving a
+ // gap behind.
+ uint32_t gap_len = (ring_pos + capacity_ - wr_) % capacity_;
+ LOGD("Drop detected; %u packets, seq_range [%hu, %hu]",
+ gap_len,
+ rd_seq_ + norm_wr_seq,
+ rd_seq_ + norm_wr_seq + gap_len - 1);
+
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ wr_ = (ring_pos + 1) % capacity_;
+ CHECK(wr_ != rd_);
+
+ return true;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::RXRingBuffer::fetchBuffer(bool* is_discon) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != is_discon);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any packets to
+ // return. If we are still waiting for the first fast start packet to show
+ // up, we don't want to let any buffer be consumed yet because we expect to
+ // see a packet before the initial read sequence number show up shortly.
+ if (!rd_seq_known_ || waiting_for_fast_start_) {
+ *is_discon = false;
+ return NULL;
+ }
+
+ PacketBuffer* ret = NULL;
+ *is_discon = !fetched_first_packet_;
+
+ while ((rd_ != wr_) && (NULL == ret)) {
+ CHECK(rd_ < capacity_);
+
+ // If we hit a gap, stall and do not advance the read pointer. Let the
+ // higher level code deal with requesting retries and/or deciding to
+ // skip the current gap.
+ ret = ring_[rd_];
+ if (NULL == ret) {
+ break;
+ }
+
+ ring_[rd_] = NULL;
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+
+ if (NULL != ret) {
+ fetched_first_packet_ = true;
+ }
+
+ return ret;
+}
+
+AAH_RXPlayer::GapStatus
+AAH_RXPlayer::RXRingBuffer::fetchCurrentGap(SeqNoGap* gap) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != gap);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any gaps.
+ if (!rd_seq_known_) {
+ return kGS_NoGap;
+ }
+
+ // If we are waiting for fast start, then the current gap is a fast start
+ // gap and it includes all packets before the read sequence number.
+ if (waiting_for_fast_start_) {
+ gap->start_seq_ =
+ gap->end_seq_ = rd_seq_ - 1;
+ return kGS_FastStartGap;
+ }
+
+ // If rd == wr, then the buffer is empty and there cannot be any gaps.
+ if (rd_ == wr_) {
+ return kGS_NoGap;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // current gap.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return kGS_NoGap;
+ }
+
+ // Looks like there must be a gap here. The start of the gap is the current
+ // rd sequence number, all we need to do now is determine its length in
+ // order to compute the end sequence number.
+ gap->start_seq_ = rd_seq_;
+ uint16_t end = rd_seq_;
+ uint32_t tmp = (rd_ + 1) % capacity_;
+ while ((tmp != wr_) && (NULL == ring_[tmp])) {
+ ++end;
+ tmp = (tmp + 1) % capacity_;
+ }
+ gap->end_seq_ = end;
+
+ return kGS_NormalGap;
+}
+
+void AAH_RXPlayer::RXRingBuffer::processNAK(SeqNoGap* nak) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+
+ // If we were waiting for our first fast start fill-in packet, and we
+ // received a NAK, then apparantly we are not getting our fast start. Just
+ // clear the waiting flag and go back to normal behavior.
+ if (waiting_for_fast_start_) {
+ waiting_for_fast_start_ = false;
+ }
+
+ // If we have not received a packet since last reset, or there is no data in
+ // the ring, then there is nothing to skip.
+ if ((!rd_seq_known_) || (rd_ == wr_)) {
+ return;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // gap to skip.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return;
+ }
+
+ // Looks like there must be a gap here. Advance rd until we have passed
+ // over the portion of it indicated by nak (or all of the gap if nak is
+ // NULL). Then reset fetched_first_packet_ so that the next read will show
+ // up as being discontiguous.
+ uint16_t seq_after_gap = (NULL == nak) ? 0 : nak->end_seq_ + 1;
+ while ((rd_ != wr_) &&
+ (NULL == ring_[rd_]) &&
+ ((NULL == nak) || (seq_after_gap != rd_seq_))) {
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+ fetched_first_packet_ = false;
+}
+
+int AAH_RXPlayer::RXRingBuffer::computeInactivityTimeout() {
+ AutoMutex lock(&lock_);
+
+ if (!rtp_activity_timeout_valid_) {
+ return -1;
+ }
+
+ uint64_t now = monotonicUSecNow();
+ if (rtp_activity_timeout_ <= now) {
+ return 0;
+ }
+
+ return (rtp_activity_timeout_ - now) / 1000;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::PacketBuffer::allocate(ssize_t length) {
+ if (length <= 0) {
+ return NULL;
+ }
+
+ uint32_t alloc_len = sizeof(PacketBuffer) + length;
+ PacketBuffer* ret = reinterpret_cast<PacketBuffer*>(
+ new uint8_t[alloc_len]);
+
+ if (NULL != ret) {
+ ret->length_ = length;
+ }
+
+ return ret;
+}
+
+void AAH_RXPlayer::PacketBuffer::destroy(PacketBuffer* pb) {
+ uint8_t* kill_me = reinterpret_cast<uint8_t*>(pb);
+ delete[] kill_me;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_substream.cpp b/media/libaah_rtp/aah_rx_player_substream.cpp
new file mode 100644
index 0000000..708ea36
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_substream.cpp
@@ -0,0 +1,474 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include <include/avc_utils.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+int64_t AAH_RXPlayer::Substream::kAboutToUnderflowThreshold =
+ 50ull * 1000;
+
+AAH_RXPlayer::Substream::Substream(uint32_t ssrc, OMXClient& omx) {
+ ssrc_ = ssrc;
+ substream_details_known_ = false;
+ buffer_in_progress_ = NULL;
+
+ decoder_ = new AAH_DecoderPump(omx);
+ if (decoder_ == NULL) {
+ LOGE("%s failed to allocate decoder pump!", __PRETTY_FUNCTION__);
+ }
+ if (OK != decoder_->initCheck()) {
+ LOGE("%s failed to initialize decoder pump!", __PRETTY_FUNCTION__);
+ }
+
+ // cleanupBufferInProgress will reset most of the internal state variables.
+ // Just need to make sure that buffer_in_progress_ is NULL before calling.
+ cleanupBufferInProgress();
+}
+
+
+void AAH_RXPlayer::Substream::shutdown() {
+ substream_meta_ = NULL;
+ cleanupBufferInProgress();
+ cleanupDecoder();
+}
+
+void AAH_RXPlayer::Substream::cleanupBufferInProgress() {
+ if (NULL != buffer_in_progress_) {
+ buffer_in_progress_->release();
+ buffer_in_progress_ = NULL;
+ }
+
+ expected_buffer_size_ = 0;
+ buffer_filled_ = 0;
+ waiting_for_rap_ = true;
+}
+
+void AAH_RXPlayer::Substream::cleanupDecoder() {
+ if (decoder_ != NULL) {
+ decoder_->shutdown();
+ }
+}
+
+void AAH_RXPlayer::Substream::processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower) {
+ uint32_t min_length = 6;
+ // Do we have a buffer in progress already? If so, abort the buffer. In
+ // theory, this should never happen. If there were a discontinutity in the
+ // stream, the discon in the seq_nos at the RTP level should have already
+ // triggered a cleanup of the buffer in progress. To see a problem at this
+ // level is an indication either of a bug in the transmitter, or some form
+ // of terrible corruption/tampering on the wire.
+ if (NULL != buffer_in_progress_) {
+ LOGE("processPayloadStart is aborting payload already in progress.");
+ cleanupBufferInProgress();
+ }
+
+ // Parse enough of the header to know where we stand. Since this is a
+ // payload start, it should begin with a TRTP header which has to be at
+ // least 6 bytes long.
+ if (amt < min_length) {
+ LOGV("Discarding payload too short to contain TRTP header (len = %u)",
+ amt);
+ return;
+ }
+
+ // Check the TRTP version number.
+ if (0x01 != buf[0]) {
+ LOGV("Unexpected TRTP version (%u) in header. Expected %u.",
+ buf[0], 1);
+ return;
+ }
+
+ // Extract the substream type field and make sure its one we understand (and
+ // one that does not conflict with any previously received substream type.
+ uint8_t header_type = (buf[1] >> 4) & 0xF;
+ switch (header_type) {
+ case 0x01:
+ // Audio, yay! Just break. We understand audio payloads.
+ break;
+ case 0x02:
+ LOGV("RXed packet with unhandled TRTP header type (Video).");
+ return;
+ case 0x03:
+ LOGV("RXed packet with unhandled TRTP header type (Subpicture).");
+ return;
+ case 0x04:
+ LOGV("RXed packet with unhandled TRTP header type (Control).");
+ return;
+ default:
+ LOGV("RXed packet with unhandled TRTP header type (%u).",
+ header_type);
+ return;
+ }
+
+ if (substream_details_known_ && (header_type != substream_type_)) {
+ LOGV("RXed TRTP Payload for SSRC=0x%08x where header type (%u) does not"
+ " match previously received header type (%u)",
+ ssrc_, header_type, substream_type_);
+ return;
+ }
+
+ // Check the flags to see if there is another 32 bits of timestamp present.
+ uint32_t trtp_header_len = 6;
+ bool ts_valid = buf[1] & 0x1;
+ if (ts_valid) {
+ min_length += 4;
+ trtp_header_len += 4;
+ if (amt < min_length) {
+ LOGV("Discarding payload too short to contain TRTP timestamp"
+ " (len = %u)", amt);
+ return;
+ }
+ }
+
+ // Extract the TRTP length field and sanity check it.
+ uint32_t trtp_len;
+ trtp_len = (static_cast<uint32_t>(buf[2]) << 24) |
+ (static_cast<uint32_t>(buf[3]) << 16) |
+ (static_cast<uint32_t>(buf[4]) << 8) |
+ static_cast<uint32_t>(buf[5]);
+ if (trtp_len < min_length) {
+ LOGV("TRTP length (%u) is too short to be valid. Must be at least %u"
+ " bytes.", trtp_len, min_length);
+ return;
+ }
+
+ // Extract the rest of the timestamp field if valid.
+ int64_t ts = 0;
+ uint32_t parse_offset = 6;
+ if (ts_valid) {
+ ts = (static_cast<int64_t>(buf[parse_offset ]) << 56) |
+ (static_cast<int64_t>(buf[parse_offset + 1]) << 48) |
+ (static_cast<int64_t>(buf[parse_offset + 2]) << 40) |
+ (static_cast<int64_t>(buf[parse_offset + 3]) << 32);
+ ts |= ts_lower;
+ parse_offset += 4;
+ }
+
+ // Check the flags to see if there is another 24 bytes of timestamp
+ // transformation present.
+ if (buf[1] & 0x2) {
+ min_length += 24;
+ parse_offset += 24;
+ trtp_header_len += 24;
+ if (amt < min_length) {
+ LOGV("Discarding payload too short to contain TRTP timestamp"
+ " transformation (len = %u)", amt);
+ return;
+ }
+ }
+
+ // TODO : break the parsing into individual parsers for the different
+ // payload types (audio, video, etc).
+ //
+ // At this point in time, we know that this is audio. Go ahead and parse
+ // the basic header, check the codec type, and find the payload portion of
+ // the packet.
+ min_length += 3;
+ if (trtp_len < min_length) {
+ LOGV("TRTP length (%u) is too short to be a valid audio payload. Must"
+ " be at least %u bytes.", trtp_len, min_length);
+ return;
+ }
+
+ if (amt < min_length) {
+ LOGV("TRTP porttion of RTP payload (%u bytes) too small to contain"
+ " entire TRTP header. TRTP does not currently support fragmenting"
+ " TRTP headers across RTP payloads", amt);
+ return;
+ }
+
+ uint8_t codec_type = buf[parse_offset ];
+ uint8_t flags = buf[parse_offset + 1];
+ uint8_t volume = buf[parse_offset + 2];
+ parse_offset += 3;
+ trtp_header_len += 3;
+
+ if (!setupSubstreamType(header_type, codec_type)) {
+ return;
+ }
+
+ if (decoder_ != NULL) {
+ decoder_->setRenderVolume(volume);
+ }
+
+ // TODO : move all of the constant flag and offset definitions for TRTP up
+ // into some sort of common header file.
+ if (waiting_for_rap_ && !(flags & 0x08)) {
+ LOGV("Dropping non-RAP TRTP Audio Payload while waiting for RAP.");
+ return;
+ }
+
+ if (flags & 0x10) {
+ LOGV("Dropping TRTP Audio Payload with aux codec data present (only"
+ " handle MP3 right now, and it has no aux data)");
+ return;
+ }
+
+ // OK - everything left is just payload. Compute the payload size, start
+ // the buffer in progress and pack as much payload as we can into it. If
+ // the payload is finished once we are done, go ahead and send the payload
+ // to the decoder.
+ expected_buffer_size_ = trtp_len - trtp_header_len;
+ if (!expected_buffer_size_) {
+ LOGV("Dropping TRTP Audio Payload with 0 Access Unit length");
+ return;
+ }
+
+ CHECK(amt >= trtp_header_len);
+ uint32_t todo = amt - trtp_header_len;
+ if (expected_buffer_size_ < todo) {
+ LOGV("Extra data (%u > %u) present in initial TRTP Audio Payload;"
+ " dropping payload.", todo, expected_buffer_size_);
+ return;
+ }
+
+ buffer_filled_ = 0;
+ buffer_in_progress_ = new MediaBuffer(expected_buffer_size_);
+ if ((NULL == buffer_in_progress_) ||
+ (NULL == buffer_in_progress_->data())) {
+ LOGV("Failed to allocate MediaBuffer of length %u",
+ expected_buffer_size_);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ sp<MetaData> meta = buffer_in_progress_->meta_data();
+ if (meta == NULL) {
+ LOGV("Missing metadata structure in allocated MediaBuffer; dropping"
+ " payload");
+ cleanupBufferInProgress();
+ return;
+ }
+
+ // TODO : set this based on the codec type indicated in the TRTP stream.
+ // Right now, we only support MP3, so the choice is obvious.
+ meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+ if (ts_valid) {
+ meta->setInt64(kKeyTime, ts);
+ }
+
+ if (amt > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt + buffer_filled_, buf + trtp_header_len, todo);
+ buffer_filled_ += amt;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processPayloadCont(uint8_t* buf,
+ uint32_t amt) {
+ if (NULL == buffer_in_progress_) {
+ LOGV("TRTP Receiver skipping payload continuation; no buffer currently"
+ " in progress.");
+ return;
+ }
+
+ CHECK(buffer_filled_ < expected_buffer_size_);
+ uint32_t buffer_left = expected_buffer_size_ - buffer_filled_;
+ if (amt > buffer_left) {
+ LOGV("Extra data (%u > %u) present in continued TRTP Audio Payload;"
+ " dropping payload.", amt, buffer_left);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ if (amt > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt + buffer_filled_, buf, amt);
+ buffer_filled_ += amt;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processCompletedBuffer() {
+ const uint8_t* buffer_data = NULL;
+ int sample_rate;
+ int channel_count;
+ size_t frame_size;
+ status_t res;
+
+ CHECK(NULL != buffer_in_progress_);
+
+ if (decoder_ == NULL) {
+ LOGV("Dropping complete buffer, no decoder pump allocated");
+ goto bailout;
+ }
+
+ buffer_data = reinterpret_cast<const uint8_t*>(buffer_in_progress_->data());
+ if (buffer_in_progress_->size() < 4) {
+ LOGV("MP3 payload too short to contain header, dropping payload.");
+ goto bailout;
+ }
+
+ // Extract the channel count and the sample rate from the MP3 header. The
+ // stagefright MP3 requires that these be delivered before decoing can
+ // begin.
+ if (!GetMPEGAudioFrameSize(U32_AT(buffer_data),
+ &frame_size,
+ &sample_rate,
+ &channel_count,
+ NULL,
+ NULL)) {
+ LOGV("Failed to parse MP3 header in payload, droping payload.");
+ goto bailout;
+ }
+
+
+ // Make sure that our substream metadata is set up properly. If there has
+ // been a format change, be sure to reset the underlying decoder. In
+ // stagefright, it seems like the only way to do this is to destroy and
+ // recreate the decoder.
+ if (substream_meta_ == NULL) {
+ substream_meta_ = new MetaData();
+
+ if (substream_meta_ == NULL) {
+ LOGE("Failed to allocate MetaData structure for substream");
+ goto bailout;
+ }
+
+ substream_meta_->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+ substream_meta_->setInt32 (kKeyChannelCount, channel_count);
+ substream_meta_->setInt32 (kKeySampleRate, sample_rate);
+ } else {
+ int32_t prev_sample_rate;
+ int32_t prev_channel_count;
+ substream_meta_->findInt32(kKeySampleRate, &prev_sample_rate);
+ substream_meta_->findInt32(kKeyChannelCount, &prev_channel_count);
+
+ if ((prev_channel_count != channel_count) ||
+ (prev_sample_rate != sample_rate)) {
+ LOGW("Format change detected, forcing decoder reset.");
+ cleanupDecoder();
+
+ substream_meta_->setInt32(kKeyChannelCount, channel_count);
+ substream_meta_->setInt32(kKeySampleRate, sample_rate);
+ }
+ }
+
+ // If our decoder has not be set up, do so now.
+ res = decoder_->init(substream_meta_);
+ if (OK != res) {
+ LOGE("Failed to init decoder (res = %d)", res);
+ cleanupDecoder();
+ substream_meta_ = NULL;
+ goto bailout;
+ }
+
+ // Queue the payload for decode.
+ res = decoder_->queueForDecode(buffer_in_progress_);
+
+ if (res != OK) {
+ LOGD("Failed to queue payload for decode, resetting decoder pump!"
+ " (res = %d)", res);
+ cleanupDecoder();
+ cleanupBufferInProgress();
+ }
+
+ // NULL out buffer_in_progress before calling the cleanup helper.
+ //
+ // MediaBuffers use something of a hybrid ref-counting pattern which prevent
+ // the AAH_DecoderPump's input queue from adding their own reference to the
+ // MediaBuffer. MediaBuffers start life with a reference count of 0, as
+ // well as an observer which starts as NULL. Before being given an
+ // observer, the ref count cannot be allowed to become non-zero as it will
+ // cause calls to release() to assert. Basically, before a MediaBuffer has
+ // an observer, they behave like non-ref counted obects where release()
+ // serves the roll of delete. After a MediaBuffer has an observer, they
+ // become more like ref counted objects where add ref and release can be
+ // used, and when the ref count hits zero, the MediaBuffer is handed off to
+ // the observer.
+ //
+ // Given all of this, when we give the buffer to the decoder pump to wait in
+ // the to-be-processed queue, the decoder cannot add a ref to the buffer as
+ // it would in a traditional ref counting system. Instead it needs to
+ // "steal" the non-existent ref. In the case of queue failure, we need to
+ // make certain to release this non-existent reference so that the buffer is
+ // cleaned up during the cleanupBufferInProgress helper. In the case of a
+ // successful queue operation, we need to make certain that the
+ // cleanupBufferInProgress helper does not release the buffer since it needs
+ // to remain alive in the queue. We acomplish this by NULLing out the
+ // buffer pointer before calling the cleanup helper.
+ buffer_in_progress_ = NULL;
+
+bailout:
+ cleanupBufferInProgress();
+}
+
+
+void AAH_RXPlayer::Substream::processTSTransform(const LinearTransform& trans) {
+ if (decoder_ != NULL) {
+ decoder_->setRenderTSTransform(trans);
+ }
+}
+
+bool AAH_RXPlayer::Substream::isAboutToUnderflow() {
+ if (decoder_ == NULL) {
+ return false;
+ }
+
+ return decoder_->isAboutToUnderflow(kAboutToUnderflowThreshold);
+}
+
+bool AAH_RXPlayer::Substream::setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type) {
+ // Sanity check the codec type. Right now we only support MP3. Also check
+ // for conflicts with previously delivered codec types.
+ if (substream_details_known_ && (codec_type != codec_type_)) {
+ LOGV("RXed TRTP Payload for SSRC=0x%08x where codec type (%u) does not"
+ " match previously received codec type (%u)",
+ ssrc_, codec_type, codec_type_);
+ return false;
+ }
+
+ if (codec_type != 0x03) {
+ LOGV("RXed TRTP Audio Payload for SSRC=0x%08x with unsupported codec"
+ " type (%u)", ssrc_, codec_type);
+ return false;
+ }
+
+ if (!substream_details_known_) {
+ substream_type_ = substream_type;
+ codec_type_ = codec_type;
+ substream_details_known_ = true;
+ }
+
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.cpp b/media/libaah_rtp/aah_tx_packet.cpp
new file mode 100644
index 0000000..f3e15e2
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.cpp
@@ -0,0 +1,331 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <arpa/inet.h>
+#include <string.h>
+
+#include <media/stagefright/MediaDebug.h>
+
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const int TRTPPacket::kRTPHeaderLen;
+const uint32_t TRTPPacket::kTRTPEpochMask;
+
+TRTPPacket::~TRTPPacket() {
+ delete mPacket;
+}
+
+/*** TRTP packet properties ***/
+
+void TRTPPacket::setSeqNumber(uint16_t val) {
+ mSeqNumber = val;
+
+ if (mIsPacked) {
+ const int kTRTPSeqNumberOffset = 2;
+ uint16_t* buf = reinterpret_cast<uint16_t*>(
+ mPacket + kTRTPSeqNumberOffset);
+ *buf = htons(mSeqNumber);
+ }
+}
+
+uint16_t TRTPPacket::getSeqNumber() const {
+ return mSeqNumber;
+}
+
+void TRTPPacket::setPTS(int64_t val) {
+ CHECK(!mIsPacked);
+ mPTS = val;
+ mPTSValid = true;
+}
+
+int64_t TRTPPacket::getPTS() const {
+ return mPTS;
+}
+
+void TRTPPacket::setEpoch(uint32_t val) {
+ mEpoch = val;
+
+ if (mIsPacked) {
+ const int kTRTPEpochOffset = 8;
+ uint32_t* buf = reinterpret_cast<uint32_t*>(
+ mPacket + kTRTPEpochOffset);
+ uint32_t val = ntohl(*buf);
+ val &= ~(kTRTPEpochMask << kTRTPEpochShift);
+ val |= (mEpoch & kTRTPEpochMask) << kTRTPEpochShift;
+ *buf = htonl(val);
+ }
+}
+
+void TRTPPacket::setProgramID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mProgramID = val;
+}
+
+void TRTPPacket::setSubstreamID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mSubstreamID = val;
+}
+
+
+void TRTPPacket::setClockTransform(const LinearTransform& trans) {
+ CHECK(!mIsPacked);
+ mClockTranform = trans;
+ mClockTranformValid = true;
+}
+
+uint8_t* TRTPPacket::getPacket() const {
+ CHECK(mIsPacked);
+ return mPacket;
+}
+
+int TRTPPacket::getPacketLen() const {
+ CHECK(mIsPacked);
+ return mPacketLen;
+}
+
+void TRTPPacket::setExpireTime(nsecs_t val) {
+ CHECK(!mIsPacked);
+ mExpireTime = val;
+}
+
+nsecs_t TRTPPacket::getExpireTime() const {
+ return mExpireTime;
+}
+
+/*** TRTP audio packet properties ***/
+
+void TRTPAudioPacket::setCodecType(TRTPAudioCodecType val) {
+ CHECK(!mIsPacked);
+ mCodecType = val;
+}
+
+void TRTPAudioPacket::setRandomAccessPoint(bool val) {
+ CHECK(!mIsPacked);
+ mRandomAccessPoint = val;
+}
+
+void TRTPAudioPacket::setDropable(bool val) {
+ CHECK(!mIsPacked);
+ mDropable = val;
+}
+
+void TRTPAudioPacket::setDiscontinuity(bool val) {
+ CHECK(!mIsPacked);
+ mDiscontinuity = val;
+}
+
+void TRTPAudioPacket::setEndOfStream(bool val) {
+ CHECK(!mIsPacked);
+ mEndOfStream = val;
+}
+
+void TRTPAudioPacket::setVolume(uint8_t val) {
+ CHECK(!mIsPacked);
+ mVolume = val;
+}
+
+void TRTPAudioPacket::setAccessUnitData(void* data, int len) {
+ CHECK(!mIsPacked);
+ mAccessUnitData = data;
+ mAccessUnitLen = len;
+}
+
+/*** TRTP control packet properties ***/
+
+void TRTPControlPacket::setCommandID(TRTPCommandID val) {
+ CHECK(!mIsPacked);
+ mCommandID = val;
+}
+
+/*** TRTP packet serializers ***/
+
+void TRTPPacket::writeU8(uint8_t*& buf, uint8_t val) {
+ *buf = val;
+ buf++;
+}
+
+void TRTPPacket::writeU16(uint8_t*& buf, uint16_t val) {
+ *reinterpret_cast<uint16_t*>(buf) = htons(val);
+ buf += 2;
+}
+
+void TRTPPacket::writeU32(uint8_t*& buf, uint32_t val) {
+ *reinterpret_cast<uint32_t*>(buf) = htonl(val);
+ buf += 4;
+}
+
+void TRTPPacket::writeU64(uint8_t*& buf, uint64_t val) {
+ buf[0] = static_cast<uint8_t>(val >> 56);
+ buf[1] = static_cast<uint8_t>(val >> 48);
+ buf[2] = static_cast<uint8_t>(val >> 40);
+ buf[3] = static_cast<uint8_t>(val >> 32);
+ buf[4] = static_cast<uint8_t>(val >> 24);
+ buf[5] = static_cast<uint8_t>(val >> 16);
+ buf[6] = static_cast<uint8_t>(val >> 8);
+ buf[7] = static_cast<uint8_t>(val);
+ buf += 8;
+}
+
+void TRTPPacket::writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen) {
+ // RTP header
+ writeU8(buf,
+ ((mVersion & 0x03) << 6) |
+ (static_cast<int>(mPadding) << 5) |
+ (static_cast<int>(mExtension) << 4) |
+ (mCsrcCount & 0x0F));
+ writeU8(buf,
+ (static_cast<int>(isFirstFragment) << 7) |
+ (mPayloadType & 0x7F));
+ writeU16(buf, mSeqNumber);
+ if (isFirstFragment && mPTSValid) {
+ writeU32(buf, mPTS & 0xFFFFFFFF);
+ } else {
+ writeU32(buf, 0);
+ }
+ writeU32(buf,
+ ((mEpoch & kTRTPEpochMask) << kTRTPEpochShift) |
+ ((mProgramID & 0x1F) << 5) |
+ (mSubstreamID & 0x1F));
+
+ // TRTP header
+ writeU8(buf, mTRTPVersion);
+ writeU8(buf,
+ ((mTRTPHeaderType & 0x0F) << 4) |
+ (mClockTranformValid ? 0x02 : 0x00) |
+ (mPTSValid ? 0x01 : 0x00));
+ writeU32(buf, totalPacketLen - kRTPHeaderLen);
+ if (mPTSValid) {
+ writeU32(buf, mPTS >> 32);
+ }
+
+ if (mClockTranformValid) {
+ writeU64(buf, mClockTranform.a_zero);
+ writeU32(buf, mClockTranform.a_to_b_numer);
+ writeU32(buf, mClockTranform.a_to_b_denom);
+ writeU64(buf, mClockTranform.b_zero);
+ }
+}
+
+bool TRTPAudioPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ int packetLen = kRTPHeaderLen +
+ mAccessUnitLen +
+ TRTPHeaderLen();
+
+ // TODO : support multiple fragments
+ const int kMaxUDPPayloadLen = 65507;
+ if (packetLen > kMaxUDPPayloadLen) {
+ return false;
+ }
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU8(cur, mCodecType);
+ writeU8(cur,
+ (static_cast<int>(mRandomAccessPoint) << 3) |
+ (static_cast<int>(mDropable) << 2) |
+ (static_cast<int>(mDiscontinuity) << 1) |
+ (static_cast<int>(mEndOfStream)));
+ writeU8(cur, mVolume);
+
+ memcpy(cur, mAccessUnitData, mAccessUnitLen);
+
+ mIsPacked = true;
+ return true;
+}
+
+int TRTPPacket::TRTPHeaderLen() const {
+ // 6 bytes for version, payload type, flags and length. An additional 4 if
+ // there are upper timestamp bits present and another 24 if there is a clock
+ // transformation present.
+ return 6 +
+ (mClockTranformValid ? 24 : 0) +
+ (mPTSValid ? 4 : 0);
+}
+
+int TRTPAudioPacket::TRTPHeaderLen() const {
+ // TRTPPacket::TRTPHeaderLen() for the base TRTPHeader. 3 bytes for audio's
+ // codec type, flags and volume field. Another 5 bytes if the codec type is
+ // PCM and we are sending sample rate/channel count. as well as however long
+ // the aux data (if present) is.
+
+ int pcmParamLength;
+ switch(mCodecType) {
+ case kCodecPCMBigEndian:
+ case kCodecPCMLittleEndian:
+ pcmParamLength = 5;
+ break;
+
+ default:
+ pcmParamLength = 0;
+ break;
+ }
+
+
+ // TODO : properly compute aux data length. Currently, nothing
+ // uses aux data, so its length is always 0.
+ int auxDataLength = 0;
+ return TRTPPacket::TRTPHeaderLen() +
+ 3 +
+ auxDataLength +
+ pcmParamLength;
+}
+
+bool TRTPControlPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ // command packets contain a 2-byte command ID
+ int packetLen = kRTPHeaderLen +
+ TRTPHeaderLen() +
+ 2;
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU16(cur, mCommandID);
+
+ mIsPacked = true;
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.h b/media/libaah_rtp/aah_tx_packet.h
new file mode 100644
index 0000000..c5f6285
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2011 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 __AAH_TX_PACKET_H__
+#define __AAH_TX_PACKET_H__
+
+#include <utils/LinearTransform.h>
+#include <utils/RefBase.h>
+#include <utils/Timers.h>
+
+namespace android {
+
+class TRTPPacket : public RefBase {
+ protected:
+ enum TRTPHeaderType {
+ kHeaderTypeAudio = 1,
+ kHeaderTypeVideo = 2,
+ kHeaderTypeSubpicture = 3,
+ kHeaderTypeControl = 4,
+ };
+
+ TRTPPacket(TRTPHeaderType headerType)
+ : mIsPacked(false)
+ , mVersion(2)
+ , mPadding(false)
+ , mExtension(false)
+ , mCsrcCount(0)
+ , mPayloadType(100)
+ , mSeqNumber(0)
+ , mPTSValid(false)
+ , mPTS(0)
+ , mEpoch(0)
+ , mProgramID(0)
+ , mSubstreamID(0)
+ , mClockTranformValid(false)
+ , mTRTPVersion(1)
+ , mTRTPLength(0)
+ , mTRTPHeaderType(headerType)
+ , mPacket(NULL)
+ , mPacketLen(0) { }
+
+ public:
+ virtual ~TRTPPacket();
+
+ void setSeqNumber(uint16_t val);
+ uint16_t getSeqNumber() const;
+
+ void setPTS(int64_t val);
+ int64_t getPTS() const;
+
+ void setEpoch(uint32_t val);
+ void setProgramID(uint16_t val);
+ void setSubstreamID(uint16_t val);
+ void setClockTransform(const LinearTransform& trans);
+
+ uint8_t* getPacket() const;
+ int getPacketLen() const;
+
+ void setExpireTime(nsecs_t val);
+ nsecs_t getExpireTime() const;
+
+ virtual bool pack() = 0;
+
+ // mask for the number of bits in a TRTP epoch
+ static const uint32_t kTRTPEpochMask = (1 << 22) - 1;
+ static const int kTRTPEpochShift = 10;
+
+ protected:
+ static const int kRTPHeaderLen = 12;
+ virtual int TRTPHeaderLen() const;
+
+ void writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen);
+
+ void writeU8(uint8_t*& buf, uint8_t val);
+ void writeU16(uint8_t*& buf, uint16_t val);
+ void writeU32(uint8_t*& buf, uint32_t val);
+ void writeU64(uint8_t*& buf, uint64_t val);
+
+ bool mIsPacked;
+
+ uint8_t mVersion;
+ bool mPadding;
+ bool mExtension;
+ uint8_t mCsrcCount;
+ uint8_t mPayloadType;
+ uint16_t mSeqNumber;
+ bool mPTSValid;
+ int64_t mPTS;
+ uint32_t mEpoch;
+ uint16_t mProgramID;
+ uint16_t mSubstreamID;
+ LinearTransform mClockTranform;
+ bool mClockTranformValid;
+ uint8_t mTRTPVersion;
+ uint32_t mTRTPLength;
+ TRTPHeaderType mTRTPHeaderType;
+
+ uint8_t* mPacket;
+ int mPacketLen;
+
+ nsecs_t mExpireTime;
+};
+
+class TRTPAudioPacket : public TRTPPacket {
+ public:
+ TRTPAudioPacket()
+ : TRTPPacket(kHeaderTypeAudio)
+ , mCodecType(kCodecInvalid)
+ , mRandomAccessPoint(false)
+ , mDropable(false)
+ , mDiscontinuity(false)
+ , mEndOfStream(false)
+ , mVolume(0)
+ , mAccessUnitData(NULL) { }
+
+ enum TRTPAudioCodecType {
+ kCodecInvalid = 0,
+ kCodecPCMBigEndian = 1,
+ kCodecPCMLittleEndian = 2,
+ kCodecMPEG1Audio = 3,
+ };
+
+ void setCodecType(TRTPAudioCodecType val);
+ void setRandomAccessPoint(bool val);
+ void setDropable(bool val);
+ void setDiscontinuity(bool val);
+ void setEndOfStream(bool val);
+ void setVolume(uint8_t val);
+ void setAccessUnitData(void* data, int len);
+
+ virtual bool pack();
+
+ protected:
+ virtual int TRTPHeaderLen() const;
+
+ private:
+ TRTPAudioCodecType mCodecType;
+ bool mRandomAccessPoint;
+ bool mDropable;
+ bool mDiscontinuity;
+ bool mEndOfStream;
+ uint8_t mVolume;
+ void* mAccessUnitData;
+ int mAccessUnitLen;
+};
+
+class TRTPControlPacket : public TRTPPacket {
+ public:
+ TRTPControlPacket()
+ : TRTPPacket(kHeaderTypeControl)
+ , mCommandID(kCommandNop) {}
+
+ enum TRTPCommandID {
+ kCommandNop = 1,
+ kCommandFlush = 2,
+ kCommandEOS = 3,
+ };
+
+ void setCommandID(TRTPCommandID val);
+
+ virtual bool pack();
+
+ private:
+ TRTPCommandID mCommandID;
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_player.cpp b/media/libaah_rtp/aah_tx_player.cpp
new file mode 100644
index 0000000..e7020c1
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.cpp
@@ -0,0 +1,1111 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+#include <utils/Log.h>
+
+#define __STDC_FORMAT_MACROS
+#include <inttypes.h>
+#include <netdb.h>
+#include <netinet/ip.h>
+
+#include <aah_timesrv/cc_helper.h>
+#include <media/IMediaPlayer.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/Timers.h>
+
+#include "aah_tx_packet.h"
+#include "aah_tx_player.h"
+
+namespace android {
+
+static int64_t kLowWaterMarkUs = 2000000ll; // 2secs
+static int64_t kHighWaterMarkUs = 10000000ll; // 10secs
+static const size_t kLowWaterMarkBytes = 40000;
+static const size_t kHighWaterMarkBytes = 200000;
+
+// When we start up, how much lead time should we put on the first access unit?
+static const int64_t kAAHStartupLeadTimeUs = 300000LL;
+
+// How much time do we attempt to lead the clock by in steady state?
+static const int64_t kAAHBufferTimeUs = 1000000LL;
+
+// how long do we keep data in our retransmit buffer after sending it.
+const int64_t AAH_TXPlayer::kAAHRetryKeepAroundTimeNs =
+ kAAHBufferTimeUs * 1100;
+
+sp<MediaPlayerBase> createAAH_TXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_TXPlayer();
+ return ret;
+}
+
+template <typename T> static T clamp(T val, T min, T max) {
+ if (val < min) {
+ return min;
+ } else if (val > max) {
+ return max;
+ } else {
+ return val;
+ }
+}
+
+struct AAH_TXEvent : public TimedEventQueue::Event {
+ AAH_TXEvent(AAH_TXPlayer *player,
+ void (AAH_TXPlayer::*method)()) : mPlayer(player)
+ , mMethod(method) {}
+
+ protected:
+ virtual ~AAH_TXEvent() {}
+
+ virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
+ (mPlayer->*mMethod)();
+ }
+
+ private:
+ AAH_TXPlayer *mPlayer;
+ void (AAH_TXPlayer::*mMethod)();
+
+ AAH_TXEvent(const AAH_TXEvent &);
+ AAH_TXEvent& operator=(const AAH_TXEvent &);
+};
+
+AAH_TXPlayer::AAH_TXPlayer()
+ : mQueueStarted(false)
+ , mFlags(0)
+ , mExtractorFlags(0) {
+ DataSource::RegisterDefaultSniffers();
+
+ mBufferingEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onBufferingUpdate);
+ mBufferingEventPending = false;
+
+ mPumpAudioEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onPumpAudio);
+ mPumpAudioEventPending = false;
+
+ reset();
+}
+
+AAH_TXPlayer::~AAH_TXPlayer() {
+ if (mQueueStarted) {
+ mQueue.stop();
+ }
+
+ reset();
+}
+
+void AAH_TXPlayer::cancelPlayerEvents(bool keepBufferingGoing) {
+ if (!keepBufferingGoing) {
+ mQueue.cancelEvent(mBufferingEvent->eventID());
+ mBufferingEventPending = false;
+
+ mQueue.cancelEvent(mPumpAudioEvent->eventID());
+ mPumpAudioEventPending = false;
+ }
+}
+
+status_t AAH_TXPlayer::initCheck() {
+ // Check for the presense of the A@H common time service by attempting to
+ // query for CommonTime's frequency. If we get an error back, we cannot
+ // talk to the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = CCHelper::getCommonFreq(&freq);
+ if (OK != res) {
+ LOGE("Failed to connect to common time service!");
+ return res;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ Mutex::Autolock autoLock(mLock);
+ return setDataSource_l(url, headers);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ reset_l();
+
+ // the URL must consist of "aahTX://" followed by the real URL of
+ // the data source
+ const char *kAAHPrefix = "aahTX://";
+ if (strncasecmp(url, kAAHPrefix, strlen(kAAHPrefix))) {
+ return INVALID_OPERATION;
+ }
+
+ mUri.setTo(url + strlen(kAAHPrefix));
+
+ if (headers) {
+ mUriHeaders = *headers;
+
+ ssize_t index = mUriHeaders.indexOfKey(String8("x-hide-urls-from-log"));
+ if (index >= 0) {
+ // Browser is in "incognito" mode, suppress logging URLs.
+
+ // This isn't something that should be passed to the server.
+ mUriHeaders.removeItemsAt(index);
+
+ mFlags |= INCOGNITO;
+ }
+ }
+
+ // The URL may optionally contain a "#" character followed by a Skyjam
+ // cookie. Ideally the cookie header should just be passed in the headers
+ // argument, but the Java API for supplying headers is apparently not yet
+ // exposed in the SDK used by application developers.
+ const char kSkyjamCookieDelimiter = '#';
+ char* skyjamCookie = strrchr(mUri.string(), kSkyjamCookieDelimiter);
+ if (skyjamCookie) {
+ skyjamCookie++;
+ mUriHeaders.add(String8("Cookie"), String8(skyjamCookie));
+ mUri = String8(mUri.string(), skyjamCookie - mUri.string());
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_TXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::prepare() {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_TXPlayer::prepareAsync() {
+ Mutex::Autolock autoLock(mLock);
+
+ return prepareAsync_l();
+}
+
+status_t AAH_TXPlayer::prepareAsync_l() {
+ if (mFlags & PREPARING) {
+ return UNKNOWN_ERROR; // async prepare already pending
+ }
+
+ mAAH_Sender = AAH_TXSender::GetInstance();
+ if (mAAH_Sender == NULL) {
+ return NO_MEMORY;
+ }
+
+ if (!mQueueStarted) {
+ mQueue.start();
+ mQueueStarted = true;
+ }
+
+ mFlags |= PREPARING;
+ mAsyncPrepareEvent = new AAH_TXEvent(
+ this, &AAH_TXPlayer::onPrepareAsyncEvent);
+
+ mQueue.postEvent(mAsyncPrepareEvent);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::finishSetDataSource_l() {
+ sp<DataSource> dataSource;
+
+ if (!strncasecmp("http://", mUri.string(), 7) ||
+ !strncasecmp("https://", mUri.string(), 8)) {
+
+ mConnectingDataSource = HTTPBase::Create(
+ (mFlags & INCOGNITO)
+ ? HTTPBase::kFlagIncognito
+ : 0);
+
+ mLock.unlock();
+ status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders);
+ mLock.lock();
+
+ if (err != OK) {
+ mConnectingDataSource.clear();
+
+ LOGI("mConnectingDataSource->connect() returned %d", err);
+ return err;
+ }
+
+ mCachedSource = new NuCachedSource2(mConnectingDataSource);
+ mConnectingDataSource.clear();
+
+ dataSource = mCachedSource;
+
+ // We're going to prefill the cache before trying to instantiate
+ // the extractor below, as the latter is an operation that otherwise
+ // could block on the datasource for a significant amount of time.
+ // During that time we'd be unable to abort the preparation phase
+ // without this prefill.
+
+ mLock.unlock();
+
+ for (;;) {
+ status_t finalStatus;
+ size_t cachedDataRemaining =
+ mCachedSource->approxDataRemaining(&finalStatus);
+
+ if (finalStatus != OK ||
+ cachedDataRemaining >= kHighWaterMarkBytes ||
+ (mFlags & PREPARE_CANCELLED)) {
+ break;
+ }
+
+ usleep(200000);
+ }
+
+ mLock.lock();
+
+ if (mFlags & PREPARE_CANCELLED) {
+ LOGI("Prepare cancelled while waiting for initial cache fill.");
+ return UNKNOWN_ERROR;
+ }
+ } else {
+ dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+ }
+
+ if (dataSource == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
+
+ if (extractor == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ return setDataSource_l(extractor);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
+ // Attempt to approximate overall stream bitrate by summing all
+ // tracks' individual bitrates, if not all of them advertise bitrate,
+ // we have to fail.
+
+ int64_t totalBitRate = 0;
+
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ int32_t bitrate;
+ if (!meta->findInt32(kKeyBitRate, &bitrate)) {
+ totalBitRate = -1;
+ break;
+ }
+
+ totalBitRate += bitrate;
+ }
+
+ mBitrate = totalBitRate;
+
+ LOGV("mBitrate = %lld bits/sec", mBitrate);
+
+ bool haveAudio = false;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ if (!strncasecmp(mime, "audio/", 6)) {
+ mAudioSource = extractor->getTrack(i);
+ CHECK(mAudioSource != NULL);
+ haveAudio = true;
+ break;
+ }
+ }
+
+ if (!haveAudio) {
+ return UNKNOWN_ERROR;
+ }
+
+ mExtractorFlags = extractor->flags();
+
+ return OK;
+}
+
+void AAH_TXPlayer::abortPrepare(status_t err) {
+ CHECK(err != OK);
+
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
+
+ mPrepareResult = err;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mPreparedCondition.broadcast();
+}
+
+void AAH_TXPlayer::onPrepareAsyncEvent() {
+ Mutex::Autolock autoLock(mLock);
+
+ if (mFlags & PREPARE_CANCELLED) {
+ LOGI("prepare was cancelled before doing anything");
+ abortPrepare(UNKNOWN_ERROR);
+ return;
+ }
+
+ if (mUri.size() > 0) {
+ status_t err = finishSetDataSource_l();
+
+ if (err != OK) {
+ abortPrepare(err);
+ return;
+ }
+ }
+
+ mAudioSource->getFormat()->findInt64(kKeyDuration, &mDurationUs);
+
+ mAudioSource->start();
+
+ mFlags |= PREPARING_CONNECTED;
+
+ if (mCachedSource != NULL) {
+ postBufferingEvent_l();
+ } else {
+ finishAsyncPrepare_l();
+ }
+}
+
+void AAH_TXPlayer::finishAsyncPrepare_l() {
+ notifyListener_l(MEDIA_PREPARED);
+
+ mPrepareResult = OK;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mFlags |= PREPARED;
+ mPreparedCondition.broadcast();
+}
+
+status_t AAH_TXPlayer::start() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return play_l();
+}
+
+status_t AAH_TXPlayer::play_l() {
+ if (mFlags & PLAYING) {
+ return OK;
+ }
+
+ if (!(mFlags & PREPARED)) {
+ return INVALID_OPERATION;
+ }
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return INVALID_OPERATION;
+ }
+ if (!mEndpointRegistered) {
+ mProgramID = mAAH_Sender->registerEndpoint(mEndpoint);
+ mEndpointRegistered = true;
+ }
+ }
+
+ mFlags |= PLAYING;
+
+ updateClockTransform_l(false);
+
+ postPumpAudioEvent_l(-1);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::stop() {
+ status_t ret = pause();
+ sendEOS_l();
+ return ret;
+}
+
+status_t AAH_TXPlayer::pause() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return pause_l();
+}
+
+status_t AAH_TXPlayer::pause_l(bool doClockUpdate) {
+ if (!(mFlags & PLAYING)) {
+ return OK;
+ }
+
+ cancelPlayerEvents(true /* keepBufferingGoing */);
+
+ mFlags &= ~PLAYING;
+
+ if (doClockUpdate) {
+ updateClockTransform_l(true);
+ }
+
+ return OK;
+}
+
+void AAH_TXPlayer::updateClockTransform_l(bool pause) {
+ // record the new pause status so that onPumpAudio knows what rate to apply
+ // when it initializes the transform
+ mPlayRateIsPaused = pause;
+
+ // if we haven't yet established a valid clock transform, then we can't
+ // do anything here
+ if (!mCurrentClockTransformValid) {
+ return;
+ }
+
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != CCHelper::getCommonTime(&commonTimeNow)) {
+ LOGE("updateClockTransform_l get common time failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // convert the current common time to media time using the old
+ // transform
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(
+ commonTimeNow, &mediaTimeNow)) {
+ LOGE("updateClockTransform_l reverse transform failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // calculate a new transform that preserves the old transform's
+ // result for the current time
+ mCurrentClockTransform.a_zero = mediaTimeNow;
+ mCurrentClockTransform.b_zero = commonTimeNow;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = pause ? 0 : 1;
+
+ // send a packet announcing the new transform
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+ queuePacketToSender_l(packet);
+}
+
+void AAH_TXPlayer::sendEOS_l() {
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandEOS);
+ queuePacketToSender_l(packet);
+}
+
+bool AAH_TXPlayer::isPlaying() {
+ return (mFlags & PLAYING) || (mFlags & CACHE_UNDERRUN);
+}
+
+status_t AAH_TXPlayer::seekTo(int msec) {
+ if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
+ Mutex::Autolock autoLock(mLock);
+ return seekTo_l(static_cast<int64_t>(msec) * 1000);
+ }
+
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_TXPlayer::seekTo_l(int64_t timeUs) {
+ mIsSeeking = true;
+ mSeekTimeUs = timeUs;
+
+ mCurrentClockTransformValid = false;
+ mLastQueuedMediaTimePTSValid = false;
+
+ // send a flush command packet
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandFlush);
+ queuePacketToSender_l(packet);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getCurrentPosition(int *msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ int position;
+
+ if (mIsSeeking) {
+ position = mSeekTimeUs / 1000;
+ } else if (mCurrentClockTransformValid) {
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != CCHelper::getCommonTime(&commonTimeNow)) {
+ LOGE("getCurrentPosition get common time failed");
+ return INVALID_OPERATION;
+ }
+
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(commonTimeNow,
+ &mediaTimeNow)) {
+ LOGE("getCurrentPosition reverse transform failed");
+ return INVALID_OPERATION;
+ }
+
+ position = static_cast<int>(mediaTimeNow / 1000);
+ } else {
+ position = 0;
+ }
+
+ int duration;
+ if (getDuration_l(&duration) == OK) {
+ *msec = clamp(position, 0, duration);
+ } else {
+ *msec = (position >= 0) ? position : 0;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getDuration(int* msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ return getDuration_l(msec);
+}
+
+status_t AAH_TXPlayer::getDuration_l(int* msec) {
+ if (mDurationUs < 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ *msec = (mDurationUs + 500) / 1000;
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::reset() {
+ Mutex::Autolock autoLock(mLock);
+ reset_l();
+ return OK;
+}
+
+void AAH_TXPlayer::reset_l() {
+ if (mFlags & PREPARING) {
+ mFlags |= PREPARE_CANCELLED;
+ if (mConnectingDataSource != NULL) {
+ LOGI("interrupting the connection process");
+ mConnectingDataSource->disconnect();
+ }
+
+ if (mFlags & PREPARING_CONNECTED) {
+ // We are basically done preparing, we're just buffering
+ // enough data to start playback, we can safely interrupt that.
+ finishAsyncPrepare_l();
+ }
+ }
+
+ while (mFlags & PREPARING) {
+ mPreparedCondition.wait(mLock);
+ }
+
+ cancelPlayerEvents();
+
+ sendEOS_l();
+
+ mCachedSource.clear();
+
+ if (mAudioSource != NULL) {
+ mAudioSource->stop();
+ }
+ mAudioSource.clear();
+
+ mFlags = 0;
+ mExtractorFlags = 0;
+
+ mDurationUs = -1;
+ mIsSeeking = false;
+ mSeekTimeUs = 0;
+
+ mUri.setTo("");
+ mUriHeaders.clear();
+
+ mBitrate = -1;
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (mAAH_Sender != NULL && mEndpointRegistered) {
+ mAAH_Sender->unregisterEndpoint(mEndpoint);
+ }
+ mEndpointRegistered = false;
+ mEndpointValid = false;
+ }
+
+ mProgramID = 0;
+
+ mAAH_Sender.clear();
+ mLastQueuedMediaTimePTSValid = false;
+ mCurrentClockTransformValid = false;
+ mPlayRateIsPaused = false;
+
+ mTRTPVolume = 255;
+}
+
+status_t AAH_TXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_TXPlayer::playerType() {
+ return AAH_TX_PLAYER;
+}
+
+status_t AAH_TXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ if (!reply) {
+ return BAD_VALUE;
+ }
+
+ int32_t methodID;
+ status_t err = request.readInt32(&methodID);
+ if (err != android::OK) {
+ return err;
+ }
+
+ switch (methodID) {
+ case kInvokeSetAAHDstIPPort:
+ case kInvokeSetAAHConfigBlob: {
+ if (mEndpointValid) {
+ return INVALID_OPERATION;
+ }
+
+ String8 addr;
+ uint16_t port;
+
+ if (methodID == kInvokeSetAAHDstIPPort) {
+ addr = String8(request.readString16());
+
+ int32_t port32;
+ err = request.readInt32(&port32);
+ if (err != android::OK) {
+ return err;
+ }
+ port = static_cast<uint16_t>(port32);
+ } else {
+ String8 blob(request.readString16());
+
+ char addr_buf[101];
+ if (sscanf(blob.string(), "V1:%100s %" SCNu16,
+ addr_buf, &port) != 2) {
+ return BAD_VALUE;
+ }
+ if (addr.setTo(addr_buf) != OK) {
+ return NO_MEMORY;
+ }
+ }
+
+ struct hostent* ent = gethostbyname(addr.string());
+ if (ent == NULL) {
+ return ERROR_UNKNOWN_HOST;
+ }
+ if (!(ent->h_addrtype == AF_INET && ent->h_length == 4)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mEndpointLock);
+ mEndpoint = AAH_TXSender::Endpoint(
+ reinterpret_cast<struct in_addr*>(ent->h_addr)->s_addr,
+ port);
+ mEndpointValid = true;
+ return OK;
+ };
+
+ default:
+ return INVALID_OPERATION;
+ }
+}
+
+status_t AAH_TXPlayer::getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records) {
+ using media::Metadata;
+
+ Metadata metadata(records);
+
+ metadata.appendBool(Metadata::kPauseAvailable, true);
+ metadata.appendBool(Metadata::kSeekBackwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekForwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekAvailable, false);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVolume(float leftVolume, float rightVolume) {
+ if (leftVolume != rightVolume) {
+ LOGE("%s does not support per channel volume: %f, %f",
+ __PRETTY_FUNCTION__, leftVolume, rightVolume);
+ }
+
+ float volume = clamp(leftVolume, 0.0f, 1.0f);
+
+ Mutex::Autolock lock(mLock);
+ mTRTPVolume = static_cast<uint8_t>((leftVolume * 255.0) + 0.5);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setAudioStreamType(int streamType) {
+ return OK;
+}
+
+void AAH_TXPlayer::notifyListener_l(int msg, int ext1, int ext2) {
+ sendEvent(msg, ext1, ext2);
+}
+
+bool AAH_TXPlayer::getBitrate_l(int64_t *bitrate) {
+ off64_t size;
+ if (mDurationUs >= 0 &&
+ mCachedSource != NULL &&
+ mCachedSource->getSize(&size) == OK) {
+ *bitrate = size * 8000000ll / mDurationUs; // in bits/sec
+ return true;
+ }
+
+ if (mBitrate >= 0) {
+ *bitrate = mBitrate;
+ return true;
+ }
+
+ *bitrate = 0;
+
+ return false;
+}
+
+// Returns true iff cached duration is available/applicable.
+bool AAH_TXPlayer::getCachedDuration_l(int64_t *durationUs, bool *eos) {
+ int64_t bitrate;
+
+ if (mCachedSource != NULL && getBitrate_l(&bitrate)) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ *durationUs = cachedDataRemaining * 8000000ll / bitrate;
+ *eos = (finalStatus != OK);
+ return true;
+ }
+
+ return false;
+}
+
+void AAH_TXPlayer::ensureCacheIsFetching_l() {
+ if (mCachedSource != NULL) {
+ mCachedSource->resumeFetchingIfNecessary();
+ }
+}
+
+void AAH_TXPlayer::postBufferingEvent_l() {
+ if (mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = true;
+ mQueue.postEventWithDelay(mBufferingEvent, 1000000ll);
+}
+
+void AAH_TXPlayer::postPumpAudioEvent_l(int64_t delayUs) {
+ if (mPumpAudioEventPending) {
+ return;
+ }
+ mPumpAudioEventPending = true;
+ mQueue.postEventWithDelay(mPumpAudioEvent, delayUs < 0 ? 10000 : delayUs);
+}
+
+void AAH_TXPlayer::onBufferingUpdate() {
+ Mutex::Autolock autoLock(mLock);
+ if (!mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = false;
+
+ if (mCachedSource != NULL) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ bool eos = (finalStatus != OK);
+
+ if (eos) {
+ if (finalStatus == ERROR_END_OF_STREAM) {
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ }
+ if (mFlags & PREPARING) {
+ LOGV("cache has reached EOS, prepare is done.");
+ finishAsyncPrepare_l();
+ }
+ } else {
+ int64_t bitrate;
+ if (getBitrate_l(&bitrate)) {
+ size_t cachedSize = mCachedSource->cachedSize();
+ int64_t cachedDurationUs = cachedSize * 8000000ll / bitrate;
+
+ int percentage = (100.0 * (double) cachedDurationUs)
+ / mDurationUs;
+ if (percentage > 100) {
+ percentage = 100;
+ }
+
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, percentage);
+ } else {
+ // We don't know the bitrate of the stream, use absolute size
+ // limits to maintain the cache.
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDataRemaining < kLowWaterMarkBytes)) {
+ LOGI("cache is running low (< %d) , pausing.",
+ kLowWaterMarkBytes);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) {
+ if (mFlags & CACHE_UNDERRUN) {
+ LOGI("cache has filled up (> %d), resuming.",
+ kHighWaterMarkBytes);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ LOGV("cache has filled up (> %d), prepare is done",
+ kHighWaterMarkBytes);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+ }
+ }
+
+ int64_t cachedDurationUs;
+ bool eos;
+ if (getCachedDuration_l(&cachedDurationUs, &eos)) {
+ LOGV("cachedDurationUs = %.2f secs, eos=%d",
+ cachedDurationUs / 1E6, eos);
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDurationUs < kLowWaterMarkUs)) {
+ LOGI("cache is running low (%.2f secs) , pausing.",
+ cachedDurationUs / 1E6);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDurationUs > kHighWaterMarkUs) {
+ if (mFlags & CACHE_UNDERRUN) {
+ LOGI("cache has filled up (%.2f secs), resuming.",
+ cachedDurationUs / 1E6);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ LOGV("cache has filled up (%.2f secs), prepare is done",
+ cachedDurationUs / 1E6);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+
+ postBufferingEvent_l();
+}
+
+void AAH_TXPlayer::onPumpAudio() {
+ while (true) {
+ Mutex::Autolock autoLock(mLock);
+ // If this flag is clear, its because someone has externally canceled
+ // this pump operation (probably because we a resetting/shutting down).
+ // Get out immediately, do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Start by checking if there is still work to be doing. If we have
+ // never queued a payload (so we don't know what the last queued PTS is)
+ // or we have never established a MediaTime->CommonTime transformation,
+ // then we have work to do (one time through this loop should establish
+ // both). Otherwise, we want to keep a fixed amt of presentation time
+ // worth of data buffered. If we cannot get common time (service is
+ // unavailable, or common time is undefined)) then we don't have a lot
+ // of good options here. For now, signal an error up to the app level
+ // and shut down the transmission pump.
+ int64_t commonTimeNow;
+ if (OK != CCHelper::getCommonTime(&commonTimeNow)) {
+ // Failed to get common time; either the service is down or common
+ // time is not synced. Raise an error and shutdown the player.
+ LOGE("*** Cannot pump audio, unable to fetch common time."
+ " Shutting down.");
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+
+ if (mCurrentClockTransformValid && mLastQueuedMediaTimePTSValid) {
+ int64_t mediaTimeNow;
+ bool conversionResult = mCurrentClockTransform.doReverseTransform(
+ commonTimeNow,
+ &mediaTimeNow);
+ CHECK(conversionResult);
+
+ if ((mediaTimeNow +
+ kAAHBufferTimeUs -
+ mLastQueuedMediaTimePTS) <= 0) {
+ break;
+ }
+ }
+
+ MediaSource::ReadOptions options;
+ if (mIsSeeking) {
+ options.setSeekTo(mSeekTimeUs);
+ }
+
+ MediaBuffer* mediaBuffer;
+ status_t err = mAudioSource->read(&mediaBuffer, &options);
+ if (err != NO_ERROR) {
+ if (err == ERROR_END_OF_STREAM) {
+ LOGI("*** %s reached end of stream", __PRETTY_FUNCTION__);
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ notifyListener_l(MEDIA_PLAYBACK_COMPLETE);
+ pause_l(false);
+ sendEOS_l();
+ } else {
+ LOGE("*** %s read failed err=%d", __PRETTY_FUNCTION__, err);
+ }
+ return;
+ }
+
+ if (mIsSeeking) {
+ mIsSeeking = false;
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ }
+
+ uint8_t* data = (static_cast<uint8_t*>(mediaBuffer->data()) +
+ mediaBuffer->range_offset());
+ LOGV("*** %s got media buffer data=[%02hhx %02hhx %02hhx %02hhx]"
+ " offset=%d length=%d", __PRETTY_FUNCTION__,
+ data[0], data[1], data[2], data[3],
+ mediaBuffer->range_offset(), mediaBuffer->range_length());
+
+ int64_t mediaTimeUs;
+ CHECK(mediaBuffer->meta_data()->findInt64(kKeyTime, &mediaTimeUs));
+ LOGV("*** timeUs=%lld", mediaTimeUs);
+
+ if (!mCurrentClockTransformValid) {
+ if (OK == CCHelper::getCommonTime(&commonTimeNow)) {
+ mCurrentClockTransform.a_zero = mediaTimeUs;
+ mCurrentClockTransform.b_zero = commonTimeNow +
+ kAAHStartupLeadTimeUs;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = mPlayRateIsPaused ? 0 : 1;
+ mCurrentClockTransformValid = true;
+ } else {
+ // Failed to get common time; either the service is down or
+ // common time is not synced. Raise an error and shutdown the
+ // player.
+ LOGE("*** Cannot begin transmission, unable to fetch common"
+ " time. Dropping sample with pts=%lld", mediaTimeUs);
+ notifyListener_l(MEDIA_ERROR,
+ MEDIA_ERROR_UNKNOWN,
+ UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+ }
+
+ LOGV("*** transmitting packet with pts=%lld", mediaTimeUs);
+
+ sp<TRTPAudioPacket> packet = new TRTPAudioPacket();
+ packet->setPTS(mediaTimeUs);
+ packet->setSubstreamID(1);
+
+ packet->setCodecType(TRTPAudioPacket::kCodecMPEG1Audio);
+ packet->setVolume(mTRTPVolume);
+ // TODO : introduce a throttle for this so we can control the
+ // frequency with which transforms get sent.
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setAccessUnitData(data, mediaBuffer->range_length());
+ packet->setRandomAccessPoint(true);
+
+ queuePacketToSender_l(packet);
+ mediaBuffer->release();
+
+ mLastQueuedMediaTimePTSValid = true;
+ mLastQueuedMediaTimePTS = mediaTimeUs;
+ }
+
+ { // Explicit scope for the autolock pattern.
+ Mutex::Autolock autoLock(mLock);
+
+ // If someone externally has cleared this flag, its because we should be
+ // shutting down. Do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Looks like no one canceled us explicitly. Clear our flag and post a
+ // new event to ourselves.
+ mPumpAudioEventPending = false;
+ postPumpAudioEvent_l(10000);
+ }
+}
+
+void AAH_TXPlayer::queuePacketToSender_l(const sp<TRTPPacket>& packet) {
+ if (mAAH_Sender == NULL) {
+ return;
+ }
+
+ sp<AMessage> message = new AMessage(AAH_TXSender::kWhatSendPacket,
+ mAAH_Sender->handlerID());
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return;
+ }
+
+ message->setInt32(AAH_TXSender::kSendPacketIPAddr, mEndpoint.addr);
+ message->setInt32(AAH_TXSender::kSendPacketPort, mEndpoint.port);
+ }
+
+ packet->setProgramID(mProgramID);
+ packet->setExpireTime(systemTime() + kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ message->setObject(AAH_TXSender::kSendPacketTRTPPacket, packet);
+
+ message->post();
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_player.h b/media/libaah_rtp/aah_tx_player.h
new file mode 100644
index 0000000..8b1551f
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.h
@@ -0,0 +1,176 @@
+/*
+ * Copyright (C) 2011 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 __AAH_TX_PLAYER_H__
+#define __AAH_TX_PLAYER_H__
+
+#include <libstagefright/include/HTTPBase.h>
+#include <libstagefright/include/NuCachedSource2.h>
+#include <libstagefright/include/TimedEventQueue.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/String8.h>
+#include <utils/threads.h>
+
+#include "aah_tx_sender.h"
+
+namespace android {
+
+class AAH_TXPlayer : public MediaPlayerHWInterface {
+ public:
+ AAH_TXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+ virtual status_t getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records);
+ virtual status_t setVolume(float leftVolume, float rightVolume);
+ virtual status_t setAudioStreamType(int streamType);
+
+ // invoke method IDs
+ enum {
+ // set the IP address and port of the A@H receiver
+ kInvokeSetAAHDstIPPort = 1,
+
+ // set the destination IP address and port (and perhaps any additional
+ // parameters added in the future) packaged in one string
+ kInvokeSetAAHConfigBlob,
+ };
+
+ static const int64_t kAAHRetryKeepAroundTimeNs;
+
+ protected:
+ virtual ~AAH_TXPlayer();
+
+ private:
+ friend struct AwesomeEvent;
+
+ enum {
+ PLAYING = 1,
+ PREPARING = 8,
+ PREPARED = 16,
+ PREPARE_CANCELLED = 64,
+ CACHE_UNDERRUN = 128,
+
+ // We are basically done preparing but are currently buffering
+ // sufficient data to begin playback and finish the preparation
+ // phase for good.
+ PREPARING_CONNECTED = 2048,
+
+ INCOGNITO = 32768,
+ };
+
+ status_t setDataSource_l(const char *url,
+ const KeyedVector<String8, String8> *headers);
+ status_t setDataSource_l(const sp<MediaExtractor>& extractor);
+ status_t finishSetDataSource_l();
+ status_t prepareAsync_l();
+ void onPrepareAsyncEvent();
+ void finishAsyncPrepare_l();
+ void abortPrepare(status_t err);
+ status_t play_l();
+ status_t pause_l(bool doClockUpdate = true);
+ status_t seekTo_l(int64_t timeUs);
+ void updateClockTransform_l(bool pause);
+ void sendEOS_l();
+ void cancelPlayerEvents(bool keepBufferingGoing = false);
+ void reset_l();
+ void notifyListener_l(int msg, int ext1 = 0, int ext2 = 0);
+ bool getBitrate_l(int64_t* bitrate);
+ status_t getDuration_l(int* msec);
+ bool getCachedDuration_l(int64_t* durationUs, bool* eos);
+ void ensureCacheIsFetching_l();
+ void postBufferingEvent_l();
+ void postPumpAudioEvent_l(int64_t delayUs);
+ void onBufferingUpdate();
+ void onPumpAudio();
+ void queuePacketToSender_l(const sp<TRTPPacket>& packet);
+
+ Mutex mLock;
+
+ TimedEventQueue mQueue;
+ bool mQueueStarted;
+
+ sp<TimedEventQueue::Event> mBufferingEvent;
+ bool mBufferingEventPending;
+
+ uint32_t mFlags;
+ uint32_t mExtractorFlags;
+
+ String8 mUri;
+ KeyedVector<String8, String8> mUriHeaders;
+
+ sp<TimedEventQueue::Event> mAsyncPrepareEvent;
+ Condition mPreparedCondition;
+ status_t mPrepareResult;
+
+ bool mIsSeeking;
+ int64_t mSeekTimeUs;
+
+ sp<TimedEventQueue::Event> mPumpAudioEvent;
+ bool mPumpAudioEventPending;
+
+ sp<HTTPBase> mConnectingDataSource;
+ sp<NuCachedSource2> mCachedSource;
+
+ sp<MediaSource> mAudioSource;
+ int64_t mDurationUs;
+ int64_t mBitrate;
+
+ sp<AAH_TXSender> mAAH_Sender;
+ LinearTransform mCurrentClockTransform;
+ bool mCurrentClockTransformValid;
+ int64_t mLastQueuedMediaTimePTS;
+ bool mLastQueuedMediaTimePTSValid;
+ bool mPlayRateIsPaused;
+
+ Mutex mEndpointLock;
+ AAH_TXSender::Endpoint mEndpoint;
+ bool mEndpointValid;
+ bool mEndpointRegistered;
+ uint16_t mProgramID;
+
+ uint8_t mTRTPVolume;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_TXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_sender.cpp b/media/libaah_rtp/aah_tx_sender.cpp
new file mode 100644
index 0000000..d894b58
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.cpp
@@ -0,0 +1,591 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+#include <media/stagefright/foundation/ADebug.h>
+
+#include <netinet/in.h>
+#include <poll.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <unistd.h>
+
+#include <media/stagefright/foundation/AMessage.h>
+#include <utils/misc.h>
+
+#include "aah_tx_player.h"
+#include "aah_tx_sender.h"
+
+namespace android {
+
+const char* AAH_TXSender::kSendPacketIPAddr = "ipaddr";
+const char* AAH_TXSender::kSendPacketPort = "port";
+const char* AAH_TXSender::kSendPacketTRTPPacket = "trtp";
+
+const int AAH_TXSender::kRetryTrimIntervalUs = 100000;
+const int AAH_TXSender::kHeartbeatIntervalUs = 1000000;
+const int AAH_TXSender::kRetryBufferCapacity = 100;
+
+Mutex AAH_TXSender::sLock;
+wp<AAH_TXSender> AAH_TXSender::sInstance;
+uint32_t AAH_TXSender::sNextEpoch;
+bool AAH_TXSender::sNextEpochValid = false;
+
+AAH_TXSender::AAH_TXSender() : mSocket(-1) { }
+
+sp<AAH_TXSender> AAH_TXSender::GetInstance() {
+ Mutex::Autolock autoLock(sLock);
+
+ sp<AAH_TXSender> sender = sInstance.promote();
+
+ if (sender == NULL) {
+ sender = new AAH_TXSender();
+ if (sender == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper = new ALooper();
+ if (sender->mLooper == NULL) {
+ return NULL;
+ }
+
+ sender->mReflector = new AHandlerReflector<AAH_TXSender>(sender.get());
+ if (sender->mReflector == NULL) {
+ return NULL;
+ }
+
+ sender->mSocket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sender->mSocket == -1) {
+ LOGW("%s unable to create socket", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ if (bind(sender->mSocket,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr)) < 0) {
+ LOGW("%s unable to bind socket (errno %d)",
+ __PRETTY_FUNCTION__, errno);
+ return NULL;
+ }
+
+ sender->mRetryReceiver = new RetryReceiver(sender.get());
+ if (sender->mRetryReceiver == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper->setName("AAH_TXSender");
+ sender->mLooper->registerHandler(sender->mReflector);
+ sender->mLooper->start(false, false, PRIORITY_AUDIO);
+
+ if (sender->mRetryReceiver->run("AAH_TXSenderRetry", PRIORITY_AUDIO)
+ != OK) {
+ LOGW("%s unable to start retry thread", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ sInstance = sender;
+ }
+
+ return sender;
+}
+
+AAH_TXSender::~AAH_TXSender() {
+ mLooper->stop();
+ mLooper->unregisterHandler(mReflector->id());
+
+ if (mRetryReceiver != NULL) {
+ mRetryReceiver->requestExit();
+ mRetryReceiver->mWakeupEvent.setEvent();
+ if (mRetryReceiver->requestExitAndWait() != OK) {
+ LOGW("%s shutdown of retry receiver failed", __PRETTY_FUNCTION__);
+ }
+ mRetryReceiver->mSender = NULL;
+ mRetryReceiver.clear();
+ }
+
+ if (mSocket != -1) {
+ close(mSocket);
+ }
+}
+
+// Return the next epoch number usable for a newly instantiated endpoint.
+uint32_t AAH_TXSender::getNextEpoch() {
+ Mutex::Autolock autoLock(sLock);
+
+ if (sNextEpochValid) {
+ sNextEpoch = (sNextEpoch + 1) & TRTPPacket::kTRTPEpochMask;
+ } else {
+ sNextEpoch = ns2ms(systemTime()) & TRTPPacket::kTRTPEpochMask;
+ sNextEpochValid = true;
+ }
+
+ return sNextEpoch;
+}
+
+// Notify the sender that a player has started sending to this endpoint.
+// Returns a program ID for use by the calling player.
+uint16_t AAH_TXSender::registerEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount++;
+ } else {
+ eps = new EndpointState(getNextEpoch());
+ mEndpointMap.add(endpoint, eps);
+ }
+
+ // if this is the first registered endpoint, then send a message to start
+ // trimming retry buffers and a message to start sending heartbeats.
+ if (mEndpointMap.size() == 1) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+
+ eps->nextProgramID++;
+ return eps->nextProgramID;
+}
+
+// Notify the sender that a player has ceased sending to this endpoint.
+// An endpoint's state can not be deleted until all of the endpoint's
+// registered players have called unregisterEndpoint.
+void AAH_TXSender::unregisterEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount--;
+ CHECK(eps->playerRefCount >= 0);
+ }
+}
+
+void AAH_TXSender::onMessageReceived(const sp<AMessage>& msg) {
+ switch (msg->what()) {
+ case kWhatSendPacket:
+ onSendPacket(msg);
+ break;
+
+ case kWhatTrimRetryBuffers:
+ trimRetryBuffers();
+ break;
+
+ case kWhatSendHeartbeats:
+ sendHeartbeats();
+ break;
+
+ default:
+ TRESPASS();
+ break;
+ }
+}
+
+void AAH_TXSender::onSendPacket(const sp<AMessage>& msg) {
+ sp<RefBase> obj;
+ CHECK(msg->findObject(kSendPacketTRTPPacket, &obj));
+ sp<TRTPPacket> packet = static_cast<TRTPPacket*>(obj.get());
+
+ uint32_t ipAddr;
+ CHECK(msg->findInt32(kSendPacketIPAddr,
+ reinterpret_cast<int32_t*>(&ipAddr)));
+
+ int32_t port32;
+ CHECK(msg->findInt32(kSendPacketPort, &port32));
+ uint16_t port = port32;
+
+ Mutex::Autolock lock(mEndpointLock);
+ doSendPacket_l(packet, Endpoint(ipAddr, port));
+}
+
+void AAH_TXSender::doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint) {
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (!eps) {
+ // the endpoint state has disappeared, so the player that sent this
+ // packet must be dead.
+ return;
+ }
+
+ // assign the packet's sequence number
+ packet->setEpoch(eps->epoch);
+ packet->setSeqNumber(eps->trtpSeqNumber++);
+
+ // add the packet to the retry buffer
+ RetryBuffer& retry = eps->retry;
+ retry.push_back(packet);
+
+ // send the packet
+ struct sockaddr_in addr;
+ memset(&addr, 0, sizeof(addr));
+ addr.sin_family = AF_INET;
+ addr.sin_addr.s_addr = endpoint.addr;
+ addr.sin_port = htons(endpoint.port);
+
+ ssize_t result = sendto(mSocket,
+ packet->getPacket(),
+ packet->getPacketLen(),
+ 0,
+ (const struct sockaddr *) &addr,
+ sizeof(addr));
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+}
+
+void AAH_TXSender::trimRetryBuffers() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ nsecs_t localTimeNow = systemTime();
+
+ Vector<Endpoint> endpointsToRemove;
+
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ RetryBuffer& retry = eps->retry;
+
+ while (!retry.isEmpty()) {
+ if (retry[0]->getExpireTime() < localTimeNow) {
+ retry.pop_front();
+ } else {
+ break;
+ }
+ }
+
+ if (retry.isEmpty() && eps->playerRefCount == 0) {
+ endpointsToRemove.add(mEndpointMap.keyAt(i));
+ }
+ }
+
+ // remove the state for any endpoints that are no longer in use
+ for (size_t i = 0; i < endpointsToRemove.size(); i++) {
+ Endpoint& e = endpointsToRemove.editItemAt(i);
+ LOGD("*** %s removing endpoint addr=%08x", __PRETTY_FUNCTION__, e.addr);
+ size_t index = mEndpointMap.indexOfKey(e);
+ delete mEndpointMap.valueAt(index);
+ mEndpointMap.removeItemsAt(index);
+ }
+
+ // schedule the next trim
+ if (mEndpointMap.size()) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+ }
+}
+
+void AAH_TXSender::sendHeartbeats() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ const Endpoint& ep = mEndpointMap.keyAt(i);
+
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+
+ packet->setExpireTime(systemTime() +
+ AAH_TXPlayer::kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ doSendPacket_l(packet, ep);
+ }
+
+ // schedule the next heartbeat
+ if (mEndpointMap.size()) {
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+}
+
+// Receiver
+
+// initial 4-byte ID of a retry request packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryRequestID = 'Treq';
+
+// initial 4-byte ID of a retry NAK packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryNakID = 'Tnak';
+
+// initial 4-byte ID of a fast start request packet
+const uint32_t AAH_TXSender::RetryReceiver::kFastStartRequestID = 'Tfst';
+
+AAH_TXSender::RetryReceiver::RetryReceiver(AAH_TXSender* sender)
+ : Thread(false),
+ mSender(sender) {}
+
+ AAH_TXSender::RetryReceiver::~RetryReceiver() {
+ mWakeupEvent.clearPendingEvents();
+ }
+
+// Returns true if val is within the interval bounded inclusively by
+// start and end. Also handles the case where there is a rollover of the
+// range between start and end.
+template <typename T>
+static inline bool withinIntervalWithRollover(T val, T start, T end) {
+ return ((start <= end && val >= start && val <= end) ||
+ (start > end && (val >= start || val <= end)));
+}
+
+bool AAH_TXSender::RetryReceiver::threadLoop() {
+ struct pollfd pollFds[2];
+ pollFds[0].fd = mSender->mSocket;
+ pollFds[0].events = POLLIN;
+ pollFds[0].revents = 0;
+ pollFds[1].fd = mWakeupEvent.getWakeupHandle();
+ pollFds[1].events = POLLIN;
+ pollFds[1].revents = 0;
+
+ int pollResult = poll(pollFds, NELEM(pollFds), -1);
+ if (pollResult == -1) {
+ LOGE("%s poll failed", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (exitPending()) {
+ LOGI("*** %s exiting", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (pollFds[0].revents) {
+ handleRetryRequest();
+ }
+
+ return true;
+}
+
+void AAH_TXSender::RetryReceiver::handleRetryRequest() {
+ LOGV("*** RX %s start", __PRETTY_FUNCTION__);
+
+ RetryPacket request;
+ struct sockaddr requestSrcAddr;
+ socklen_t requestSrcAddrLen = sizeof(requestSrcAddr);
+
+ ssize_t result = recvfrom(mSender->mSocket, &request, sizeof(request), 0,
+ &requestSrcAddr, &requestSrcAddrLen);
+ if (result == -1) {
+ LOGE("%s recvfrom failed, errno=%d", __PRETTY_FUNCTION__, errno);
+ return;
+ }
+
+ if (static_cast<size_t>(result) < sizeof(RetryPacket)) {
+ LOGW("%s short packet received", __PRETTY_FUNCTION__);
+ return;
+ }
+
+ uint32_t host_request_id = ntohl(request.id);
+ if ((host_request_id != kRetryRequestID) &&
+ (host_request_id != kFastStartRequestID)) {
+ LOGW("%s received retry request with bogus ID (%08x)",
+ __PRETTY_FUNCTION__, host_request_id);
+ return;
+ }
+
+ Endpoint endpoint(request.endpointIP, ntohs(request.endpointPort));
+
+ Mutex::Autolock lock(mSender->mEndpointLock);
+
+ EndpointState* eps = mSender->mEndpointMap.valueFor(endpoint);
+
+ if (eps == NULL || eps->retry.isEmpty()) {
+ // we have no retry buffer or an empty retry buffer for this endpoint,
+ // so NAK the entire request
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ RetryBuffer& retry = eps->retry;
+
+ uint16_t startSeq = ntohs(request.seqStart);
+ uint16_t endSeq = ntohs(request.seqEnd);
+
+ uint16_t retryFirstSeq = retry[0]->getSeqNumber();
+ uint16_t retryLastSeq = retry[retry.size() - 1]->getSeqNumber();
+
+ // If this is a fast start, then force the start of the retry to match the
+ // start of the retransmit ring buffer (unless the end of the retransmit
+ // ring buffer is already past the point of fast start)
+ if ((host_request_id == kFastStartRequestID) &&
+ !((startSeq - retryFirstSeq) & 0x8000)) {
+ startSeq = retryFirstSeq;
+ }
+
+ int startIndex;
+ if (withinIntervalWithRollover(startSeq, retryFirstSeq, retryLastSeq)) {
+ startIndex = static_cast<uint16_t>(startSeq - retryFirstSeq);
+ } else {
+ startIndex = -1;
+ }
+
+ int endIndex;
+ if (withinIntervalWithRollover(endSeq, retryFirstSeq, retryLastSeq)) {
+ endIndex = static_cast<uint16_t>(endSeq - retryFirstSeq);
+ } else {
+ endIndex = -1;
+ }
+
+ if (startIndex == -1 && endIndex == -1) {
+ // no part of the request range is found in the retry buffer
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ if (startIndex == -1) {
+ // NAK a subrange at the front of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqEnd = htons(retryFirstSeq - 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ startIndex = 0;
+ } else if (endIndex == -1) {
+ // NAK a subrange at the back of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqStart = htons(retryLastSeq + 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ endIndex = retry.size() - 1;
+ }
+
+ // send the retry packets
+ for (int i = startIndex; i <= endIndex; i++) {
+ const sp<TRTPPacket>& replyPacket = retry[i];
+
+ result = sendto(mSender->mSocket,
+ replyPacket->getPacket(),
+ replyPacket->getPacketLen(),
+ 0,
+ &requestSrcAddr,
+ requestSrcAddrLen);
+
+ if (result == -1) {
+ LOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ }
+}
+
+// Endpoint
+
+AAH_TXSender::Endpoint::Endpoint()
+ : addr(0)
+ , port(0) { }
+
+AAH_TXSender::Endpoint::Endpoint(uint32_t a, uint16_t p)
+ : addr(a)
+ , port(p) {}
+
+bool AAH_TXSender::Endpoint::operator<(const Endpoint& other) const {
+ return ((addr < other.addr) ||
+ (addr == other.addr && port < other.port));
+}
+
+// EndpointState
+
+AAH_TXSender::EndpointState::EndpointState(uint32_t _epoch)
+ : retry(kRetryBufferCapacity)
+ , playerRefCount(1)
+ , trtpSeqNumber(0)
+ , nextProgramID(0)
+ , epoch(_epoch) { }
+
+// CircularBuffer
+
+template <typename T>
+CircularBuffer<T>::CircularBuffer(size_t capacity)
+ : mCapacity(capacity)
+ , mHead(0)
+ , mTail(0)
+ , mFillCount(0) {
+ mBuffer = new T[capacity];
+}
+
+template <typename T>
+CircularBuffer<T>::~CircularBuffer() {
+ delete [] mBuffer;
+}
+
+template <typename T>
+void CircularBuffer<T>::push_back(const T& item) {
+ if (this->isFull()) {
+ this->pop_front();
+ }
+ mBuffer[mHead] = item;
+ mHead = (mHead + 1) % mCapacity;
+ mFillCount++;
+}
+
+template <typename T>
+void CircularBuffer<T>::pop_front() {
+ CHECK(!isEmpty());
+ mBuffer[mTail] = T();
+ mTail = (mTail + 1) % mCapacity;
+ mFillCount--;
+}
+
+template <typename T>
+size_t CircularBuffer<T>::size() const {
+ return mFillCount;
+}
+
+template <typename T>
+bool CircularBuffer<T>::isFull() const {
+ return (mFillCount == mCapacity);
+}
+
+template <typename T>
+bool CircularBuffer<T>::isEmpty() const {
+ return (mFillCount == 0);
+}
+
+template <typename T>
+const T& CircularBuffer<T>::itemAt(size_t index) const {
+ CHECK(index < mFillCount);
+ return mBuffer[(mTail + index) % mCapacity];
+}
+
+template <typename T>
+const T& CircularBuffer<T>::operator[](size_t index) const {
+ return itemAt(index);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_sender.h b/media/libaah_rtp/aah_tx_sender.h
new file mode 100644
index 0000000..df4d162
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (C) 2011 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 __AAH_TX_SENDER_H__
+#define __AAH_TX_SENDER_H__
+
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+#include "aah_tx_packet.h"
+#include "pipe_event.h"
+
+namespace android {
+
+template <typename T> class CircularBuffer {
+ public:
+ CircularBuffer(size_t capacity);
+ ~CircularBuffer();
+ void push_back(const T& item);;
+ void pop_front();
+ size_t size() const;
+ bool isFull() const;
+ bool isEmpty() const;
+ const T& itemAt(size_t index) const;
+ const T& operator[](size_t index) const;
+
+ private:
+ T* mBuffer;
+ size_t mCapacity;
+ size_t mHead;
+ size_t mTail;
+ size_t mFillCount;
+};
+
+class AAH_TXSender : public virtual RefBase {
+ public:
+ ~AAH_TXSender();
+
+ static sp<AAH_TXSender> GetInstance();
+
+ ALooper::handler_id handlerID() { return mReflector->id(); }
+
+ // an IP address and port
+ struct Endpoint {
+ Endpoint();
+ Endpoint(uint32_t a, uint16_t p);
+ bool operator<(const Endpoint& other) const;
+
+ uint32_t addr;
+ uint16_t port;
+ };
+
+ uint16_t registerEndpoint(const Endpoint& endpoint);
+ void unregisterEndpoint(const Endpoint& endpoint);
+
+ enum {
+ kWhatSendPacket,
+ kWhatTrimRetryBuffers,
+ kWhatSendHeartbeats,
+ };
+
+ // fields for SendPacket messages
+ static const char* kSendPacketIPAddr;
+ static const char* kSendPacketPort;
+ static const char* kSendPacketTRTPPacket;
+
+ private:
+ AAH_TXSender();
+
+ static Mutex sLock;
+ static wp<AAH_TXSender> sInstance;
+ static uint32_t sNextEpoch;
+ static bool sNextEpochValid;
+
+ static uint32_t getNextEpoch();
+
+ typedef CircularBuffer<sp<TRTPPacket> > RetryBuffer;
+
+ // state maintained on a per-endpoint basis
+ struct EndpointState {
+ EndpointState(uint32_t epoch);
+ RetryBuffer retry;
+ int playerRefCount;
+ uint16_t trtpSeqNumber;
+ uint16_t nextProgramID;
+ uint32_t epoch;
+ };
+
+ friend class AHandlerReflector<AAH_TXSender>;
+ void onMessageReceived(const sp<AMessage>& msg);
+ void onSendPacket(const sp<AMessage>& msg);
+ void doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint);
+ void trimRetryBuffers();
+ void sendHeartbeats();
+
+ sp<ALooper> mLooper;
+ sp<AHandlerReflector<AAH_TXSender> > mReflector;
+
+ int mSocket;
+
+ DefaultKeyedVector<Endpoint, EndpointState*> mEndpointMap;
+ Mutex mEndpointLock;
+
+ static const int kRetryTrimIntervalUs;
+ static const int kHeartbeatIntervalUs;
+ static const int kRetryBufferCapacity;
+
+ class RetryReceiver : public Thread {
+ private:
+ friend class AAH_TXSender;
+
+ RetryReceiver(AAH_TXSender* sender);
+ virtual ~RetryReceiver();
+ virtual bool threadLoop();
+ void handleRetryRequest();
+
+ static const int kMaxReceiverPacketLen;
+ static const uint32_t kRetryRequestID;
+ static const uint32_t kFastStartRequestID;
+ static const uint32_t kRetryNakID;
+
+ AAH_TXSender* mSender;
+ PipeEvent mWakeupEvent;
+ };
+
+ sp<RetryReceiver> mRetryReceiver;
+};
+
+struct RetryPacket {
+ uint32_t id;
+ uint32_t endpointIP;
+ uint16_t endpointPort;
+ uint16_t seqStart;
+ uint16_t seqEnd;
+} __attribute__((packed));
+
+} // namespace android
+
+#endif // __AAH_TX_SENDER_H__
diff --git a/media/libaah_rtp/pipe_event.cpp b/media/libaah_rtp/pipe_event.cpp
new file mode 100644
index 0000000..96c674b
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.cpp
@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2011 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 "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <unistd.h>
+
+#include "pipe_event.h"
+
+namespace android {
+
+PipeEvent::PipeEvent() {
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+
+ // Create the pipe.
+ if (pipe(pipe_) >= 0) {
+ // Set non-blocking mode on the read side of the pipe so we can
+ // easily drain it whenever we wakeup.
+ fcntl(pipe_[0], F_SETFL, O_NONBLOCK);
+ } else {
+ LOGE("Failed to create pipe event %d %d %d",
+ pipe_[0], pipe_[1], errno);
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+ }
+}
+
+PipeEvent::~PipeEvent() {
+ if (pipe_[0] >= 0) {
+ close(pipe_[0]);
+ }
+
+ if (pipe_[1] >= 0) {
+ close(pipe_[1]);
+ }
+}
+
+void PipeEvent::clearPendingEvents() {
+ char drain_buffer[16];
+ while (read(pipe_[0], drain_buffer, sizeof(drain_buffer)) > 0) {
+ // No body.
+ }
+}
+
+bool PipeEvent::wait(int timeout) {
+ struct pollfd wait_fd;
+
+ wait_fd.fd = getWakeupHandle();
+ wait_fd.events = POLLIN;
+ wait_fd.revents = 0;
+
+ int res = poll(&wait_fd, 1, timeout);
+
+ if (res < 0) {
+ LOGE("Wait error in PipeEvent; sleeping to prevent overload!");
+ usleep(1000);
+ }
+
+ return (res > 0);
+}
+
+void PipeEvent::setEvent() {
+ char foo = 'q';
+ write(pipe_[1], &foo, 1);
+}
+
+} // namespace android
+
diff --git a/media/libaah_rtp/pipe_event.h b/media/libaah_rtp/pipe_event.h
new file mode 100644
index 0000000..637bfef
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2011 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 __PIPE_EVENT_H__
+#define __PIPE_EVENT_H__
+
+namespace android {
+
+class PipeEvent {
+ public:
+ PipeEvent();
+ ~PipeEvent();
+
+ bool initCheck() const {
+ return ((pipe_[0] >= 0) && (pipe_[1] >= 0));
+ }
+
+ int getWakeupHandle() const { return pipe_[0]; }
+
+ // block until the event fires; returns true if the event fired and false if
+ // the wait timed out. Timeout is expressed in milliseconds; negative
+ // values mean wait forever.
+ bool wait(int timeout = -1);
+
+ void clearPendingEvents();
+ void setEvent();
+
+ private:
+ int pipe_[2];
+};
+
+} // namespace android
+
+#endif // __PIPE_EVENT_H__
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 8ebb652..4bc534b 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -79,7 +79,8 @@
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false)
{
}
@@ -94,7 +95,8 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
@@ -112,7 +114,8 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0, flags, cbf, user, notificationFrames,
@@ -520,6 +523,10 @@
{
int afSamplingRate;
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
@@ -533,6 +540,10 @@
uint32_t AudioTrack::getSampleRate()
{
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
AutoMutex lock(mLock);
return mCblk->sampleRate;
}
@@ -558,6 +569,10 @@
return NO_ERROR;
}
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount ||
cblk->server > loopStart) {
@@ -641,6 +656,8 @@
status_t AudioTrack::setPosition(uint32_t position)
{
+ if (mIsTimed) return INVALID_OPERATION;
+
AutoMutex lock(mLock);
Mutex::Autolock _l(mCblk->lock);
@@ -790,6 +807,7 @@
((uint16_t)flags) << 16,
sharedBuffer,
output,
+ mIsTimed,
&mSessionId,
&status);
@@ -958,6 +976,7 @@
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
+ if (mIsTimed) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
// sanity-check. user is most-likely passing an error code.
@@ -1020,6 +1039,36 @@
// -------------------------------------------------------------------------
+TimedAudioTrack::TimedAudioTrack() {
+ mIsTimed = true;
+}
+
+status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
+{
+ return mAudioTrack->allocateTimedBuffer(size, buffer);
+}
+
+status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts)
+{
+ // restart track if it was disabled by audioflinger due to previous underrun
+ if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
+ mCblk->flags &= ~CBLK_DISABLED_ON;
+ LOGW("queueTimedBuffer() track %p disabled, restarting", this);
+ mAudioTrack->start();
+ }
+
+ return mAudioTrack->queueTimedBuffer(buffer, pts);
+}
+
+status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target)
+{
+ return mAudioTrack->setMediaTimeTransform(xform, target);
+}
+
+// -------------------------------------------------------------------------
+
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
@@ -1462,4 +1511,3 @@
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index d58834b..c7b49cd 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -90,6 +90,7 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
@@ -105,6 +106,7 @@
data.writeInt32(flags);
data.writeStrongBinder(sharedBuffer->asBinder());
data.writeInt32(output);
+ data.writeInt32(isTimed);
int lSessionId = 0;
if (sessionId != NULL) {
lSessionId = *sessionId;
@@ -684,11 +686,12 @@
uint32_t flags = data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
int output = data.readInt32();
+ bool isTimed = data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioTrack> track = createTrack(pid,
streamType, sampleRate, format,
- channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
+ channelCount, bufferCount, flags, buffer, output, isTimed, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(track->asBinder());
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index bc8ff34..a496fb8 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -35,7 +35,10 @@
FLUSH,
MUTE,
PAUSE,
- ATTACH_AUX_EFFECT
+ ATTACH_AUX_EFFECT,
+ ALLOCATE_TIMED_BUFFER,
+ QUEUE_TIMED_BUFFER,
+ SET_MEDIA_TIME_TRANSFORM,
};
class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -113,6 +116,52 @@
}
return status;
}
+
+ virtual status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt32(size);
+ status_t status = remote()->transact(ALLOCATE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ if (status == NO_ERROR) {
+ *buffer = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeStrongBinder(buffer->asBinder());
+ data.writeInt64(pts);
+ status_t status = remote()->transact(QUEUE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt64(xform.a_zero);
+ data.writeInt64(xform.b_zero);
+ data.writeInt32(xform.a_to_b_numer);
+ data.writeInt32(xform.a_to_b_denom);
+ data.writeInt32(target);
+ status_t status = remote()->transact(SET_MEDIA_TIME_TRANSFORM,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -158,10 +207,38 @@
reply->writeInt32(attachAuxEffect(data.readInt32()));
return NO_ERROR;
} break;
+ case ALLOCATE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer;
+ status_t status = allocateTimedBuffer(data.readInt32(), &buffer);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeStrongBinder(buffer->asBinder());
+ }
+ return NO_ERROR;
+ } break;
+ case QUEUE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer = interface_cast<IMemory>(
+ data.readStrongBinder());
+ uint64_t pts = data.readInt64();
+ reply->writeInt32(queueTimedBuffer(buffer, pts));
+ return NO_ERROR;
+ } break;
+ case SET_MEDIA_TIME_TRANSFORM: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ LinearTransform xform;
+ xform.a_zero = data.readInt64();
+ xform.b_zero = data.readInt64();
+ xform.a_to_b_numer = data.readInt32();
+ xform.a_to_b_denom = data.readInt32();
+ int target = data.readInt32();
+ reply->writeInt32(setMediaTimeTransform(xform, target));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
}
}; // namespace android
-
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index a3e2517..e521648 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -29,7 +29,8 @@
libstagefright_omx \
libstagefright_foundation \
libgui \
- libdl
+ libdl \
+ libaah_rtp
LOCAL_STATIC_LIBRARIES := \
libstagefright_nuplayer \
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index e8d0f0c..5b1d6c7 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -70,6 +70,11 @@
#include <OMX.h>
+namespace android {
+sp<MediaPlayerBase> createAAH_TXPlayer();
+sp<MediaPlayerBase> createAAH_RXPlayer();
+}
+
namespace {
using android::media::Metadata;
using android::status_t;
@@ -593,6 +598,14 @@
return NU_PLAYER;
}
+ if (!strncasecmp("aahRX://", url, 8)) {
+ return AAH_RX_PLAYER;
+ }
+
+ if (!strncasecmp("aahTX://", url, 8)) {
+ return AAH_TX_PLAYER;
+ }
+
// use MidiFile for MIDI extensions
int lenURL = strlen(url);
for (int i = 0; i < NELEM(FILE_EXTS); ++i) {
@@ -629,6 +642,14 @@
LOGV("Create Test Player stub");
p = new TestPlayerStub();
break;
+ case AAH_RX_PLAYER:
+ LOGV(" create A@H RX Player");
+ p = createAAH_RXPlayer();
+ break;
+ case AAH_TX_PLAYER:
+ LOGV(" create A@H TX Player");
+ p = createAAH_TXPlayer();
+ break;
default:
LOGE("Unknown player type: %d", playerType);
return NULL;
@@ -1031,9 +1052,21 @@
status_t MediaPlayerService::Client::setVolume(float leftVolume, float rightVolume)
{
LOGV("[%d] setVolume(%f, %f)", mConnId, leftVolume, rightVolume);
- // TODO: for hardware output, call player instead
- Mutex::Autolock l(mLock);
- if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+
+ // for hardware output, call player instead
+ sp<MediaPlayerBase> p = getPlayer();
+ {
+ Mutex::Autolock l(mLock);
+ if (p != 0 && p->hardwareOutput()) {
+ MediaPlayerHWInterface* hwp =
+ reinterpret_cast<MediaPlayerHWInterface*>(p.get());
+ return hwp->setVolume(leftVolume, rightVolume);
+ } else {
+ if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+ return NO_ERROR;
+ }
+ }
+
return NO_ERROR;
}
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fa49592..d4eea3d 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -8,7 +8,8 @@
AudioResampler.cpp.arm \
AudioResamplerSinc.cpp.arm \
AudioResamplerCubic.cpp.arm \
- AudioPolicyService.cpp
+ AudioPolicyService.cpp \
+ AudioBufferProvider.cpp
LOCAL_C_INCLUDES := \
system/media/audio_effects/include
@@ -22,7 +23,8 @@
libhardware_legacy \
libeffects \
libdl \
- libpowermanager
+ libpowermanager \
+ libaah_timesrv_client
LOCAL_STATIC_LIBRARIES := \
libcpustats \
diff --git a/services/audioflinger/AudioBufferProvider.cpp b/services/audioflinger/AudioBufferProvider.cpp
new file mode 100644
index 0000000..678fd58
--- /dev/null
+++ b/services/audioflinger/AudioBufferProvider.cpp
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2011 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.
+ */
+
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include "AudioBufferProvider.h"
+
+namespace android {
+
+const int64_t AudioBufferProvider::kInvalidPTS = INT64_MAX;
+
+}; // namespace android
diff --git a/services/audioflinger/AudioBufferProvider.h b/services/audioflinger/AudioBufferProvider.h
index 81c5c39..62ad6bd 100644
--- a/services/audioflinger/AudioBufferProvider.h
+++ b/services/audioflinger/AudioBufferProvider.h
@@ -38,8 +38,15 @@
};
virtual ~AudioBufferProvider() {}
-
- virtual status_t getNextBuffer(Buffer* buffer) = 0;
+
+ // value representing an invalid presentation timestamp
+ static const int64_t kInvalidPTS;
+
+ // pts is the local time when the next sample yielded by getNextBuffer
+ // will be rendered.
+ // Pass kInvalidPTS if the PTS is unknown or not applicable.
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts) = 0;
+
virtual void releaseBuffer(Buffer* buffer) = 0;
};
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index ff262f1..f00e732 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -58,6 +58,9 @@
#include <powermanager/PowerManager.h>
// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
+#include <aah_timesrv/cc_helper.h>
+#include <aah_timesrv/local_clock.h>
+
// ----------------------------------------------------------------------------
@@ -152,8 +155,9 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
- mBtNrecIsOff(false)
+ mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterVolumeSW(1.0f),
+ mMasterVolumeSupportLvl(MVS_NONE), mMasterMute(false), mNextUniqueId(1),
+ mBtNrecIsOff(false)
{
}
@@ -192,6 +196,33 @@
return;
}
+ // Determine the level of master volume support the primary audio HAL has,
+ // and set the initial master volume at the same time.
+ float initialVolume = 1.0;
+ mMasterVolumeSupportLvl = MVS_NONE;
+ if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
+ AutoMutex lock(mHardwareLock);
+ audio_hw_device_t *dev = mPrimaryHardwareDev;
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ if ((NULL != dev->get_master_volume) &&
+ (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_FULL;
+ } else {
+ mMasterVolumeSupportLvl = MVS_SETONLY;
+ initialVolume = 1.0;
+ }
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if ((NULL == dev->set_master_volume) ||
+ (NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_NONE;
+ }
+ mHardwareStatus = AUDIO_HW_INIT;
+ }
+
+ // Set the mode for each audio HAL, and try to set the initial volume (if
+ // supported) for all of the non-primary audio HALs.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
@@ -203,11 +234,22 @@
mMode = AUDIO_MODE_NORMAL;
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- dev->set_master_volume(dev, 1.0f);
- mHardwareStatus = AUDIO_HW_IDLE;
+
+ if ((dev != mPrimaryHardwareDev) &&
+ (NULL != dev->set_master_volume)) {
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ dev->set_master_volume(dev, initialVolume);
+ }
+
+ mHardwareStatus = AUDIO_HW_INIT;
}
}
+
+ mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
+ ? initialVolume
+ : 1.0;
+ mMasterVolume = initialVolume;
+ mHardwareStatus = AUDIO_HW_IDLE;
}
status_t AudioFlinger::initCheck() const
@@ -378,6 +420,7 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
@@ -441,7 +484,7 @@
LOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -534,20 +577,29 @@
return PERMISSION_DENIED;
}
+ float swmv = value;
+
// when hw supports master volume, don't scale in sw mixer
- { // scope for the lock
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
- value = 1.0f;
+ if (MVS_NONE != mMasterVolumeSupportLvl) {
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ AutoMutex lock(mHardwareLock);
+ audio_hw_device_t *dev = mAudioHwDevs[i];
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (NULL != dev->set_master_volume) {
+ dev->set_master_volume(dev, value);
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
}
- mHardwareStatus = AUDIO_HW_IDLE;
+
+ swmv = 1.0;
}
Mutex::Autolock _l(mLock);
- mMasterVolume = value;
+ mMasterVolume = value;
+ mMasterVolumeSW = swmv;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterVolume(value);
+ mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
return NO_ERROR;
}
@@ -635,9 +687,27 @@
float AudioFlinger::masterVolume() const
{
+ if (MVS_FULL == mMasterVolumeSupportLvl) {
+ float ret_val;
+ AutoMutex lock(mHardwareLock);
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ assert(NULL != mPrimaryHardwareDev);
+ assert(NULL != mPrimaryHardwareDev->get_master_volume);
+
+ mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret_val;
+ }
+
return mMasterVolume;
}
+float AudioFlinger::masterVolumeSW() const
+{
+ return mMasterVolumeSW;
+}
+
bool AudioFlinger::masterMute() const
{
return mMasterMute;
@@ -1368,7 +1438,8 @@
readOutputParameters();
- mMasterVolume = mAudioFlinger->masterVolume();
+ mMasterVolume = mAudioFlinger->masterVolumeSW();
+
mMasterMute = mAudioFlinger->masterMute();
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
@@ -1479,6 +1550,7 @@
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status)
{
sp<Track> track;
@@ -1528,9 +1600,14 @@
}
}
- track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId);
- if (track->getCblk() == NULL || track->name() < 0) {
+ if (!isTimed) {
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ } else {
+ track = TimedTrack::create(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ }
+ if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -1854,6 +1931,10 @@
acquireWakeLock();
+ LocalClock lc;
+ uint64_t localTimeFreq;
+ localTimeFreq = lc.getLocalFreq();
+
while (!exitPending())
{
#ifdef DEBUG_CPU_USAGE
@@ -1950,8 +2031,21 @@
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ // obtain the presentation timestamp of the next output buffer
+ int64_t pts;
+ status_t status = INVALID_OPERATION;
+
+ if (NULL != mOutput->stream->get_next_write_timestamp) {
+ status = mOutput->stream->get_next_write_timestamp(
+ mOutput->stream, &pts);
+ }
+
+ if (status != NO_ERROR) {
+ pts = AudioBufferProvider::kInvalidPTS;
+ }
+
// mix buffers...
- mAudioMixer->process();
+ mAudioMixer->process(pts);
sleepTime = 0;
standbyTime = systemTime() + kStandbyTimeInNsecs;
//TODO: delay standby when effects have a tail
@@ -2080,7 +2174,7 @@
minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1;
}
}
- if ((cblk->framesReady() >= minFrames) && track->isReady() &&
+ if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
@@ -2736,7 +2830,8 @@
// output audio to hardware
while (frameCount) {
buffer.frameCount = frameCount;
- activeTrack->getNextBuffer(&buffer);
+ activeTrack->getNextBuffer(&buffer,
+ AudioBufferProvider::kInvalidPTS);
if (UNLIKELY(buffer.raw == 0)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
@@ -3005,7 +3100,7 @@
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
- mAudioMixer->process();
+ mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
} else {
memset(mMixBuffer, 0, mixBufferSize);
}
@@ -3400,7 +3495,8 @@
(int)mAuxBuffer);
}
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
@@ -3441,10 +3537,14 @@
return NOT_ENOUGH_DATA;
}
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
+ return mCblk->framesReady();
+}
+
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
- if (mCblk->framesReady() >= mCblk->frameCount ||
+ if (framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
@@ -3613,6 +3713,393 @@
mAuxBuffer = buffer;
}
+// timed audio tracks
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+ const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ uint32_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId) {
+ if (!client->reserveTimedTrack())
+ return NULL;
+
+ sp<TimedTrack> track = new TimedTrack(
+ thread, client, streamType, sampleRate, format, channelMask, frameCount,
+ sharedBuffer, sessionId);
+
+ if (track == NULL) {
+ client->releaseTimedTrack();
+ return NULL;
+ }
+
+ return track;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+ const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ uint32_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : Track(thread, client, streamType, sampleRate, format, channelMask,
+ frameCount, sharedBuffer, sessionId),
+ mTimedSilenceBuffer(NULL),
+ mTimedSilenceBufferSize(0),
+ mTimedAudioOutputOnTime(false),
+ mMediaTimeTransformValid(false)
+{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
+ mLocalTimeToSampleTransform.a_zero = 0;
+ mLocalTimeToSampleTransform.b_zero = 0;
+ mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+ mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+ LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+ &mLocalTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+ mClient->releaseTimedTrack();
+ delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+ size_t size, sp<IMemory>* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ trimTimedBufferQueue_l();
+
+ // lazily initialize the shared memory heap for timed buffers
+ if (mTimedMemoryDealer == NULL) {
+ const int kTimedBufferHeapSize = 512 << 10;
+
+ mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+ "AudioFlingerTimed");
+ if (mTimedMemoryDealer == NULL)
+ return NO_MEMORY;
+ }
+
+ sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL) {
+ newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL)
+ return NO_MEMORY;
+ }
+
+ *buffer = newBuffer;
+ return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+ int64_t mediaTimeNow;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return;
+
+ int64_t targetTimeNow;
+ status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+ ? CCHelper::getCommonTime(&targetTimeNow)
+ : CCHelper::getLocalTime(&targetTimeNow);
+
+ if (OK != res)
+ return;
+
+ if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+ &mediaTimeNow)) {
+ return;
+ }
+ }
+
+ size_t trimIndex;
+ for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
+ if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
+ break;
+ }
+
+ if (trimIndex) {
+ mTimedBufferQueue.removeItemsAt(0, trimIndex);
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts) {
+
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+ const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+ LOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
+ xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+ target);
+
+ if (!(target == TimedAudioTrack::LOCAL_TIME ||
+ target == TimedAudioTrack::COMMON_TIME)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMediaTimeTransformLock);
+ mMediaTimeTransform = xform;
+ mMediaTimeTransformTarget = target;
+ mMediaTimeTransformValid = true;
+
+ return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ if (pts == AudioBufferProvider::kInvalidPTS) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+
+ // get ahold of the output stream that these samples will be written to
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread == NULL) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+ PlaybackThread* playbackThread = static_cast<PlaybackThread*>(thread.get());
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ while (true) {
+
+ // if we have no timed buffers, then fail
+ if (mTimedBufferQueue.isEmpty()) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ // calculate the PTS of the head of the timed buffer queue expressed in
+ // local time
+ int64_t headLocalPTS;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+ assert(mMediaTimeTransformValid);
+
+ if (mMediaTimeTransform.a_to_b_denom == 0) {
+ // the transform represents a pause, so yield silence
+ timedYieldSilence(buffer->frameCount, buffer);
+ return NO_ERROR;
+ }
+
+ int64_t transformedPTS;
+ if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+ &transformedPTS)) {
+ // the transform failed. this shouldn't happen, but if it does
+ // then just drop this buffer
+ LOGW("timedGetNextBuffer transform failed");
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ mTimedBufferQueue.removeAt(0);
+ return NO_ERROR;
+ }
+
+ if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+ if (OK != CCHelper::commonTimeToLocalTime(transformedPTS,
+ &headLocalPTS)) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+ } else {
+ headLocalPTS = transformedPTS;
+ }
+ }
+
+ // adjust the head buffer's PTS to reflect the portion of the head buffer
+ // that has already been consumed
+ int64_t effectivePTS = headLocalPTS +
+ ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
+
+ // Calculate the delta in samples between the head of the input buffer
+ // queue and the start of the next output buffer that will be written.
+ // If the transformation fails because of over or underflow, it means
+ // that the sample's position in the output stream is so far out of
+ // whack that it should just be dropped.
+ int64_t sampleDelta;
+ if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+ LOGV("*** head buffer is too far from PTS: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+ if (!mLocalTimeToSampleTransform.doForwardTransform(
+ (effectivePTS - pts) << 32, &sampleDelta)) {
+ LOGV("*** too late during sample rate transform: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+
+ LOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
+ __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
+ static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
+ static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+ // if the delta between the ideal placement for the next input sample and
+ // the current output position is within this threshold, then we will
+ // concatenate the next input samples to the previous output
+ const int64_t kSampleContinuityThreshold =
+ (static_cast<int64_t>(sampleRate()) << 32) / 10;
+
+ // if this is the first buffer of audio that we're emitting from this track
+ // then it should be almost exactly on time.
+ const int64_t kSampleStartupThreshold = 1LL << 32;
+
+ if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+ (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+ // the next input is close enough to being on time, so concatenate it
+ // with the last output
+ timedYieldSamples(buffer);
+
+ LOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ } else if (sampleDelta > 0) {
+ // the gap between the current output position and the proper start of
+ // the next input sample is too big, so fill it with silence
+ uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+ timedYieldSilence(framesUntilNextInput, buffer);
+ LOGV("*** silence: frameCount=%u", buffer->frameCount);
+ return NO_ERROR;
+ } else {
+ // the next input sample is late
+ uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+ size_t onTimeSamplePosition =
+ head.position() + lateFrames * mCblk->frameSize;
+
+ if (onTimeSamplePosition > head.buffer()->size()) {
+ // all the remaining samples in the head are too late, so
+ // drop it and move on
+ LOGV("*** too late: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ } else {
+ // skip over the late samples
+ head.setPosition(onTimeSamplePosition);
+
+ // yield the available samples
+ timedYieldSamples(buffer);
+
+ LOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ }
+ }
+ }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
+ AudioBufferProvider::Buffer* buffer) {
+
+ const TimedBuffer& head = mTimedBufferQueue[0];
+
+ buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+ head.position());
+
+ uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+ mCblk->frameSize);
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+ mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
+ uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+ // lazily allocate a buffer filled with silence
+ if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+ delete [] mTimedSilenceBuffer;
+ mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+ mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+ memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+ }
+
+ buffer->raw = mTimedSilenceBuffer;
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(numFrames, framesRequested);
+
+ mTimedAudioOutputOnTime = false;
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+ AudioBufferProvider::Buffer* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ if (buffer->raw != mTimedSilenceBuffer) {
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+ head.setPosition(head.position() + buffer->frameCount * mCblk->frameSize);
+ if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
+ mTimedBufferQueue.removeAt(0);
+ }
+ }
+
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+}
+
+uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ uint32_t frames = 0;
+ for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
+ const TimedBuffer& tb = mTimedBufferQueue[i];
+ frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize;
+ }
+
+ return frames;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+ : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts)
+ : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
@@ -3649,7 +4136,7 @@
}
}
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
@@ -3971,7 +4458,8 @@
: RefBase(),
mAudioFlinger(audioFlinger),
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
- mPid(pid)
+ mPid(pid),
+ mTimedTrackCount(0)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
@@ -3987,6 +4475,31 @@
return mMemoryDealer;
}
+// Reserve one of the limited slots for a timed audio track associated
+// with this client
+bool AudioFlinger::Client::reserveTimedTrack()
+{
+ const int kMaxTimedTracksPerClient = 4;
+
+ Mutex::Autolock _l(mTimedTrackLock);
+
+ if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
+ LOGW("can not create timed track - pid %d has exceeded the limit",
+ mPid);
+ return false;
+ }
+
+ mTimedTrackCount++;
+ return true;
+}
+
+// Release a slot for a timed audio track
+void AudioFlinger::Client::releaseTimedTrack()
+{
+ Mutex::Autolock _l(mTimedTrackLock);
+ mTimedTrackCount--;
+}
+
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
@@ -4058,6 +4571,38 @@
return mTrack->attachAuxEffect(EffectId);
}
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+ const LinearTransform& xform, int target) {
+
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->setMediaTimeTransform(
+ xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -4295,7 +4840,8 @@
}
buffer.frameCount = mFrameCount;
- if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ if (LIKELY(mActiveTrack->getNextBuffer(
+ &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == 0) {
// no resampling
@@ -4574,7 +5120,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 4b794ef..6bac595 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -84,6 +84,7 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
+ bool isTimed,
int *sessionId,
status_t *status);
@@ -97,6 +98,7 @@
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
+ virtual float masterVolumeSW() const;
virtual bool masterMute() const;
virtual status_t setStreamVolume(int stream, float value, int output);
@@ -188,6 +190,7 @@
AUDIO_HW_SET_MIC_MUTE,
AUDIO_SET_VOICE_VOLUME,
AUDIO_SET_PARAMETER,
+ AUDIO_HW_GET_MASTER_VOLUME,
};
// record interface
@@ -235,12 +238,18 @@
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
+ bool reserveTimedTrack();
+ void releaseTimedTrack();
+
private:
Client(const Client&);
Client& operator = (const Client&);
sp<AudioFlinger> mAudioFlinger;
sp<MemoryDealer> mMemoryDealer;
pid_t mPid;
+
+ Mutex mTimedTrackLock;
+ int mTimedTrackCount;
};
// --- Notification Client ---
@@ -346,7 +355,9 @@
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
uint32_t format() const {
@@ -611,7 +622,6 @@
int16_t *mainBuffer() { return mMainBuffer; }
int auxEffectId() { return mAuxEffectId; }
-
protected:
friend class ThreadBase;
friend class TrackHandle;
@@ -622,7 +632,11 @@
Track(const Track&);
Track& operator = (const Track&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual uint32_t framesReady() const;
+
bool isMuted() { return mMute; }
bool isPausing() const {
return mState == PAUSING;
@@ -638,6 +652,8 @@
return (mStreamType == AUDIO_STREAM_CNT);
}
+ virtual bool isTimedTrack() const { return false; }
+
// we don't really need a lock for these
float mVolume[2];
volatile bool mMute;
@@ -655,6 +671,78 @@
bool mHasVolumeController;
}; // end of Track
+ class TimedTrack : public Track {
+ public:
+ static sp<TimedTrack> create(const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ uint32_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+ ~TimedTrack();
+
+ class TimedBuffer {
+ public:
+ TimedBuffer();
+ TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+ const sp<IMemory>& buffer() const { return mBuffer; }
+ int64_t pts() const { return mPTS; }
+ int position() const { return mPosition; }
+ void setPosition(int pos) { mPosition = pos; }
+ private:
+ sp<IMemory> mBuffer;
+ int64_t mPTS;
+ int mPosition;
+ };
+
+ virtual bool isTimedTrack() const { return true; }
+
+ virtual uint32_t framesReady() const;
+
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSilence(uint32_t numFrames,
+ AudioBufferProvider::Buffer* buffer);
+
+ status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TimedAudioTrack::TargetTimeline target);
+ void trimTimedBufferQueue_l();
+
+ private:
+ TimedTrack(const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ uint32_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+
+ uint64_t mLocalTimeFreq;
+ LinearTransform mLocalTimeToSampleTransform;
+ sp<MemoryDealer> mTimedMemoryDealer;
+ Vector<TimedBuffer> mTimedBufferQueue;
+ uint8_t* mTimedSilenceBuffer;
+ uint32_t mTimedSilenceBufferSize;
+ mutable Mutex mTimedBufferQueueLock;
+ bool mTimedAudioOutputOnTime;
+
+ Mutex mMediaTimeTransformLock;
+ LinearTransform mMediaTimeTransform;
+ bool mMediaTimeTransformValid;
+ TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+ };
+
// playback track
class OutputTrack : public Track {
@@ -728,6 +816,7 @@
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status);
AudioStreamOut* getOutput();
@@ -916,6 +1005,12 @@
virtual void setVolume(float left, float right);
virtual sp<IMemory> getCblk() const;
virtual status_t attachAuxEffect(int effectId);
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
@@ -963,7 +1058,9 @@
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
bool mOverflow;
};
@@ -999,7 +1096,8 @@
AudioStreamIn* clearInput();
virtual audio_stream_t* stream();
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
@@ -1376,6 +1474,12 @@
friend class RecordThread;
friend class PlaybackThread;
+ enum master_volume_support {
+ MVS_NONE,
+ MVS_SETONLY,
+ MVS_FULL,
+ };
+
mutable Mutex mLock;
DefaultKeyedVector< pid_t, wp<Client> > mClients;
@@ -1389,6 +1493,8 @@
DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
PlaybackThread::stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
float mMasterVolume;
+ float mMasterVolumeSW;
+ master_volume_support mMasterVolumeSupportLvl;
bool mMasterMute;
DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 6e9319d..760981b 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <assert.h>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -30,6 +31,9 @@
#include <system/audio.h>
+#include <aah_timesrv/local_clock.h>
+#include <aah_timesrv/cc_helper.h>
+
#include "AudioMixer.h"
namespace android {
@@ -47,6 +51,9 @@
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
: mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
@@ -74,6 +81,7 @@
t->in = 0;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->localTimeFreq = mLocalTimeFreq;
t++;
}
}
@@ -293,6 +301,7 @@
if (resampler == 0) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
+ resampler->setLocalTimeFreq(localTimeFreq);
}
return true;
}
@@ -340,13 +349,13 @@
-void AudioMixer::process()
+void AudioMixer::process(int64_t pts)
{
- mState.hook(&mState);
+ mState.hook(&mState, pts);
}
-void AudioMixer::process__validate(state_t* state)
+void AudioMixer::process__validate(state_t* state, int64_t pts)
{
LOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -450,7 +459,7 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
+ state->hook(state, pts);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
@@ -859,7 +868,7 @@
}
// no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
@@ -889,7 +898,9 @@
size_t outFrames = state->frameCount;
while (outFrames) {
t1.buffer.frameCount = outFrames;
- t1.bufferProvider->getNextBuffer(&t1.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t1, pts, state->frameCount - outFrames);
+ t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS);
if (!t1.buffer.raw) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
@@ -899,7 +910,7 @@
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
@@ -911,7 +922,7 @@
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
@@ -965,7 +976,9 @@
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t, pts, numFrames + (BLOCKSIZE - outFrames));
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
@@ -994,7 +1007,7 @@
// generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1033,6 +1046,7 @@
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+ t.resampler->setPTS(pts);
(t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
@@ -1040,7 +1054,8 @@
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
@@ -1060,7 +1075,8 @@
}
// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
const int i = 31 - __builtin_clz(state->enabledTracks);
const track_t& t = state->tracks[i];
@@ -1075,7 +1091,8 @@
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
- t.bufferProvider->getNextBuffer(&b);
+ int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
+ t.bufferProvider->getNextBuffer(&b, outputPTS);
int16_t const *in = b.i16;
// in == NULL can happen if the track was flushed just after having
@@ -1118,7 +1135,8 @@
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
int i;
uint32_t en = state->enabledTracks;
@@ -1152,7 +1170,9 @@
if (frameCount0 == 0) {
b0.frameCount = numFrames;
- t0.bufferProvider->getNextBuffer(&b0);
+ int64_t outputPTS = calculateOutputPTS(t0, pts,
+ out - t0.mainBuffer);
+ t0.bufferProvider->getNextBuffer(&b0, outputPTS);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1166,7 +1186,9 @@
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
- t1.bufferProvider->getNextBuffer(&b1);
+ int64_t outputPTS = calculateOutputPTS(t1, pts,
+ out - t0.mainBuffer);
+ t1.bufferProvider->getNextBuffer(&b1, outputPTS);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1213,6 +1235,11 @@
}
}
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex)
+{
+ return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+}
+
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 75c9170..d32a366 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -85,7 +85,7 @@
status_t setParameter(int target, int name, void *value);
status_t setBufferProvider(AudioBufferProvider* bufferProvider);
- void process();
+ void process(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
@@ -125,7 +125,7 @@
struct state_t;
struct track_t;
- typedef void (*mix_t)(state_t* state);
+ typedef void (*mix_t)(state_t* state, int64_t pts);
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
static const int BLOCKSIZE = 16; // 4 cache lines
@@ -163,6 +163,8 @@
int32_t* mainBuffer;
int32_t* auxBuffer;
+ uint64_t localTimeFreq;
+
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const;
void resetResampler();
@@ -185,6 +187,8 @@
uint32_t mTrackNames;
const uint32_t mSampleRate;
+ int64_t mLocalTimeFreq;
+
state_t mState __attribute__((aligned(32)));
void invalidateState(uint32_t mask);
@@ -196,12 +200,17 @@
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
- static void process__validate(state_t* state);
- static void process__nop(state_t* state);
- static void process__genericNoResampling(state_t* state);
- static void process__genericResampling(state_t* state);
- static void process__OneTrack16BitsStereoNoResampling(state_t* state);
- static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
+ static void process__validate(state_t* state, int64_t pts);
+ static void process__nop(state_t* state, int64_t pts);
+ static void process__genericNoResampling(state_t* state, int64_t pts);
+ static void process__genericResampling(state_t* state, int64_t pts);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+ static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+
+ static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex);
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 9ee5a30..cd2857f 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -59,10 +59,12 @@
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
- uint32_t &phaseFraction, uint32_t phaseIncrement);
+ uint32_t &phaseFraction, uint32_t phaseIncrement)
+ __attribute__((noinline));
void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
- uint32_t &phaseFraction, uint32_t phaseIncrement);
+ uint32_t &phaseFraction, uint32_t phaseIncrement)
+ __attribute__((noinline));
#endif // ASM_ARM_RESAMP1
static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
@@ -118,7 +120,8 @@
int32_t sampleRate) :
mBitDepth(bitDepth), mChannelCount(inChannelCount),
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
- mPhaseFraction(0) {
+ mPhaseFraction(0), mLocalTimeFreq(0),
+ mPTS(AudioBufferProvider::kInvalidPTS) {
// sanity check on format
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
LOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
@@ -152,6 +155,23 @@
mVolume[1] = right;
}
+void AudioResampler::setLocalTimeFreq(uint64_t freq) {
+ mLocalTimeFreq = freq;
+}
+
+void AudioResampler::setPTS(int64_t pts) {
+ mPTS = pts;
+}
+
+int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
+
+ if (mPTS == AudioBufferProvider::kInvalidPTS) {
+ return AudioBufferProvider::kInvalidPTS;
+ } else {
+ return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
+ }
+}
+
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
@@ -198,7 +218,8 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto resampleStereo16_exit;
}
@@ -292,7 +313,8 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
@@ -602,4 +624,3 @@
// ----------------------------------------------------------------------------
}
; // namespace android
-
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 9f06c1c..da736ab 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -49,6 +49,10 @@
virtual void init() = 0;
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(int16_t left, int16_t right);
+ virtual void setLocalTimeFreq(uint64_t freq);
+
+ // set the PTS of the next buffer output by the resampler
+ virtual void setPTS(int64_t pts);
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
@@ -72,6 +76,8 @@
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
+ int64_t calculateOutputPTS(int outputFrameIndex);
+
int32_t mBitDepth;
int32_t mChannelCount;
int32_t mSampleRate;
@@ -86,6 +92,8 @@
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
+ uint64_t mLocalTimeFreq;
+ int64_t mPTS;
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 4d721f6..e864b30 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -65,7 +65,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
// LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -95,7 +95,8 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
in = mBuffer.i16;
@@ -130,7 +131,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
// LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
@@ -160,7 +161,8 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
// LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -181,4 +183,3 @@
// ----------------------------------------------------------------------------
}
; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 9e5e254..6184795 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -204,7 +204,8 @@
// buffer is empty, fetch a new one
while (buffer.frameCount == 0) {
buffer.frameCount = inFrameCount;
- provider->getNextBuffer(&buffer);
+ provider->getNextBuffer(&buffer,
+ calculateOutputPTS(outputIndex / 2));
if (buffer.raw == NULL) {
goto resample_exit;
}
@@ -355,4 +356,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-