Add media/libaah_rtp (033e42b7) · Commits · e / os / android_frameworks_av

include/media/MediaPlayerInterface.h

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		@@ -46,6 +46,9 @@ enum player_type {
		// 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,
		};

media/libaah_rtp/Android.mk

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		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)

media/libaah_rtp/aah_decoder_pump.cpp

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		/*
		* 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_(this)
		, thread_status_(OK)
		, renderer_(NULL)
		, last_queued_pts_valid_(false)
		, last_queued_pts_(0)
		, last_ts_transform_valid_(false)
		, last_volume_(0xFF) {
		}

		AAH_DecoderPump::~AAH_DecoderPump() {
		shutdown();
		}

		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

media/libaah_rtp/aah_decoder_pump.h

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		/*
		* 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 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_;

		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__

media/libaah_rtp/aah_rx_player.cpp

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