summaryrefslogtreecommitdiffstats
path: root/media/libaah_rtp/aah_rx_player_core.cpp
diff options
context:
space:
mode:
authorJohn Grossman <johngro@google.com>2012-02-09 15:09:05 -0800
committerJohn Grossman <johngro@google.com>2012-02-16 13:45:12 -0800
commit761defc341c5ce9019a42919c441f035f665ec0d (patch)
tree23b76ab41456c90e80559ffbb7d8e31a38104dc4 /media/libaah_rtp/aah_rx_player_core.cpp
parentef7740be67a4d7b6b033ebed59c3d4a9c74a2c18 (diff)
downloadframeworks_av-761defc341c5ce9019a42919c441f035f665ec0d.zip
frameworks_av-761defc341c5ce9019a42919c441f035f665ec0d.tar.gz
frameworks_av-761defc341c5ce9019a42919c441f035f665ec0d.tar.bz2
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>
Diffstat (limited to 'media/libaah_rtp/aah_rx_player_core.cpp')
-rw-r--r--media/libaah_rtp/aah_rx_player_core.cpp807
1 files changed, 807 insertions, 0 deletions
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..d2b3386
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_core.cpp
@@ -0,0 +1,807 @@
+/*
+ * 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) {
+ ALOGE("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) {
+ ALOGE("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) {
+ ALOGW("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) {
+ ALOGE("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) {
+ ALOGE("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);
+ ALOGE("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) {
+ ALOGW("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) {
+ ALOGW("Failed to fetch socket buffer size. (errno %d)", errno);
+ } else {
+ ALOGI("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) {
+ ALOGE("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) {
+ ALOGW("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) {
+ ALOGE("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)) {
+ ALOGE("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) {
+ ALOGE("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_) {
+ ALOGE("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);
+ ALOGV("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.
+ ALOGE("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();
+ }
+
+ // Step 5: Check for fatal errors. If any of our substreams has
+ // encountered a fatal, unrecoverable, error, then propagate the error
+ // up to user level and shut down.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ status_t status;
+ CHECK(substreams_.valueAt(i) != NULL);
+
+ status = substreams_.valueAt(i)->getStatus();
+ if (OK != status) {
+ ALOGE("Substream index %d has encountered an unrecoverable"
+ " error (%d). Signalling application level and shutting"
+ " down.", i, status);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+ }
+
+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) {
+ ALOGV("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))) {
+ ALOGV("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_);
+
+ ALOGV("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) {
+ ALOGV("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) {
+ ALOGV("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_) {
+ ALOGV("%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) {
+ ALOGV("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) {
+ ALOGV("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)) {
+ ALOGV("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) {
+ ALOGE("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) {
+ ALOGV("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;
+ ALOGI("*** %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()) {
+ ALOGV("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_)) {
+ ALOGV("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);
+ ALOGV("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) {
+ ALOGE("Error when sending retransmit request (%d)", errno);
+ } else {
+ ALOGV("%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