Upintegreate AAH TX and RX players from ICS_AAH

Upintegrate the android at home TX and RX players developed in the
ICS_AAH branch.

Change-Id: I8247d3702e30d8b0e215b31a92675d8ab28dccbb
Signed-off-by: John Grossman <johngro@google.com>

1 files changed, 520 insertions, 0 deletions

diff --git a/media/libaah_rtp/aah_decoder_pump.cpp b/media/libaah_rtp/aah_decoder_pump.cpp
new file mode 100644
index 0000000..72fe43b
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.cpp
@@ -0,0 +1,520 @@
+/*
+ * 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 <common_time/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/Timers.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+
+namespace android {
+
+static const long long kLongDecodeErrorThreshold = 1000000ll;
+static const uint32_t kMaxLongErrorsBeforeFatal = 3;
+static const uint32_t kMaxErrorsBeforeFatal = 60;
+
+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) {
+        ALOGE("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))) {
+        ALOGV("Decoded sample missing timestamp, cannot render.");
+        CHECK(false);
+    } 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) {
+                    ALOGE("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) {
+                        ALOGE("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) {
+                            ALOGW("%s: setVolume failed", __FUNCTION__);
+                        }
+
+                        renderer_->start();
+                    }
+                }
+            } else {
+                ALOGE("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) {
+                ALOGE("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) {
+                    ALOGE("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 {
+            ALOGE("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) {
+            ALOGE("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) {
+            ALOGW("%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 != cc_helper_.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_) {
+        ALOGE("AAH_DecoderPump's work thread failed to start decoder (res = %d)",
+                thread_status_);
+        return NULL;
+    }
+
+    DurationTimer decode_timer;
+    uint32_t consecutive_long_errors = 0;
+    uint32_t consecutive_errors = 0;
+
+    while (!thread_->exitPending()) {
+        status_t res;
+        MediaBuffer* bufOut = NULL;
+
+        decode_timer.start();
+        res = decoder_->read(&bufOut);
+        decode_timer.stop();
+
+        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;
+        }
+
+        // Try to be a little nuanced in our handling of actual decode errors.
+        // Errors could happen because of minor stream corruption or because of
+        // transient resource limitations.  In these cases, we would rather drop
+        // a little bit of output and ride out the unpleasantness then throw up
+        // our hands and abort everything.
+        //
+        // OTOH - When things are really bad (like we have a non-transient
+        // resource or bookkeeping issue, or the stream being fed to us is just
+        // complete and total garbage) we really want to terminate playback and
+        // raise an error condition all the way up to the application level so
+        // they can deal with it.
+        //
+        // Unfortunately, the error codes returned by the decoder can be a
+        // little non-specific.  For example, if an OMXCodec times out
+        // attempting to obtain an output buffer, the error we get back is a
+        // generic -1.  Try to distinguish between this resource timeout error
+        // and ES corruption error by timing how long the decode operation
+        // takes.  Maintain accounting for both errors and "long errors".  If we
+        // get more than a certain number consecutive errors of either type,
+        // consider it fatal and shutdown (which will cause the error to
+        // propagate all of the way up to the application level).  The threshold
+        // for "long errors" is deliberately much lower than that of normal
+        // decode errors, both because of how long they take to happen and
+        // because they generally indicate resource limitation errors which are
+        // unlikely to go away in pathologically bad cases (in contrast to
+        // stream corruption errors which might happen 20 times in a row and
+        // then be suddenly OK again)
+        if (res != OK) {
+            consecutive_errors++;
+            if (decode_timer.durationUsecs() >= kLongDecodeErrorThreshold)
+                consecutive_long_errors++;
+
+            CHECK(NULL == bufOut);
+
+            ALOGW("%s: Failed to decode data (res = %d)",
+                    __PRETTY_FUNCTION__, res);
+
+            if ((consecutive_errors      >= kMaxErrorsBeforeFatal) ||
+                (consecutive_long_errors >= kMaxLongErrorsBeforeFatal)) {
+                ALOGE("%s: Maximum decode error threshold has been reached."
+                      " There have been %d consecutive decode errors, and %d"
+                      " consecutive decode operations which resulted in errors"
+                      " and took more than %lld uSec to process.  The last"
+                      " decode operation took %lld uSec.",
+                      __PRETTY_FUNCTION__,
+                      consecutive_errors, consecutive_long_errors,
+                      kLongDecodeErrorThreshold, decode_timer.durationUsecs());
+                thread_status_ = res;
+                break;
+            }
+
+            continue;
+        }
+
+        if (NULL == bufOut) {
+            ALOGW("%s: Successful decode, but no buffer produced",
+                    __PRETTY_FUNCTION__);
+            continue;
+        }
+
+        // Successful decode (with actual output produced).  Clear the error
+        // counters.
+        consecutive_errors = 0;
+        consecutive_long_errors = 0;
+
+        queueToRenderer(bufOut);
+        bufOut->release();
+    }
+
+    decoder_->stop();
+    stopAndCleanupRenderer();
+
+    return NULL;
+}
+
+status_t AAH_DecoderPump::init(const 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) {
+      ALOGE("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) {
+        ALOGE("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();
+
+    for (MBQueue::iterator iter = in_queue_.begin();
+         iter != in_queue_.end();
+         ++iter) {
+        (*iter)->release();
+    }
+    in_queue_.clear();
+
+    last_queued_pts_valid_   = false;
+    last_ts_transform_valid_ = false;
+    last_volume_             = 0xFF;
+    thread_status_           = OK;
+
+    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