diff options
Diffstat (limited to 'media/libaah_rtp/aah_rx_player_substream.cpp')
| -rw-r--r-- | media/libaah_rtp/aah_rx_player_substream.cpp | 677 |
1 files changed, 0 insertions, 677 deletions
diff --git a/media/libaah_rtp/aah_rx_player_substream.cpp b/media/libaah_rtp/aah_rx_player_substream.cpp deleted file mode 100644 index 18b0e2b..0000000 --- a/media/libaah_rtp/aah_rx_player_substream.cpp +++ /dev/null @@ -1,677 +0,0 @@ -/* - * 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" -#include "aah_tx_packet.h" - -inline uint32_t min(uint32_t a, uint32_t b) { - return (a < b ? a : b); -} - -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; - status_ = OK; - codec_mime_type_ = ""; - - decoder_ = new AAH_DecoderPump(omx); - if (decoder_ == NULL) { - ALOGE("%s failed to allocate decoder pump!", __PRETTY_FUNCTION__); - } - if (OK != decoder_->initCheck()) { - ALOGE("%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(); -} - -AAH_RXPlayer::Substream::~Substream() { - shutdown(); -} - -void AAH_RXPlayer::Substream::shutdown() { - substream_meta_ = NULL; - status_ = OK; - 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; - - aux_data_in_progress_.clear(); - aux_data_expected_size_ = 0; -} - -void AAH_RXPlayer::Substream::cleanupDecoder() { - if (decoder_ != NULL) { - decoder_->shutdown(); - } -} - -bool AAH_RXPlayer::Substream::shouldAbort(const char* log_tag) { - // If we have already encountered a fatal error, do nothing. We are just - // waiting for our owner to shut us down now. - if (OK != status_) { - ALOGV("Skipping %s, substream has encountered fatal error (%d).", - log_tag, status_); - return true; - } - - return false; -} - -void AAH_RXPlayer::Substream::processPayloadStart(uint8_t* buf, - uint32_t amt, - int32_t ts_lower) { - uint32_t min_length = 6; - - if (shouldAbort(__PRETTY_FUNCTION__)) { - return; - } - - // 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_) { - ALOGE("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) { - ALOGV("Discarding payload too short to contain TRTP header (len = %u)", - amt); - return; - } - - // Check the TRTP version number. - if (0x01 != buf[0]) { - ALOGV("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 TRTPPacket::kHeaderTypeAudio: - // Audio, yay! Just break. We understand audio payloads. - break; - case TRTPPacket::kHeaderTypeVideo: - ALOGV("RXed packet with unhandled TRTP header type (Video)."); - return; - case TRTPPacket::kHeaderTypeSubpicture: - ALOGV("RXed packet with unhandled TRTP header type (Subpicture)."); - return; - case TRTPPacket::kHeaderTypeControl: - ALOGV("RXed packet with unhandled TRTP header type (Control)."); - return; - default: - ALOGV("RXed packet with unhandled TRTP header type (%u).", - header_type); - return; - } - - if (substream_details_known_ && (header_type != substream_type_)) { - ALOGV("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] & TRTPPacket::kFlag_TSValid; - if (ts_valid) { - min_length += 4; - trtp_header_len += 4; - if (amt < min_length) { - ALOGV("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 = U32_AT(buf + 2); - if (trtp_len < min_length) { - ALOGV("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) { - uint32_t ts_upper = U32_AT(buf + parse_offset); - parse_offset += 4; - ts = (static_cast<int64_t>(ts_upper) << 32) | ts_lower; - } - - // Check the flags to see if there is another 24 bytes of timestamp - // transformation present. - if (buf[1] & TRTPPacket::kFlag_TSTransformPresent) { - min_length += 24; - parse_offset += 24; - trtp_header_len += 24; - if (amt < min_length) { - ALOGV("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) { - ALOGV("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) { - ALOGV("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); - } - - if (waiting_for_rap_ && !(flags & TRTPAudioPacket::kFlag_RandomAccessPoint)) { - ALOGV("Dropping non-RAP TRTP Audio Payload while waiting for RAP."); - return; - } - - // Check for the presence of codec aux data. - if (flags & TRTPAudioPacket::kFlag_AuxLengthPresent) { - min_length += 4; - trtp_header_len += 4; - - if (trtp_len < min_length) { - ALOGV("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) { - ALOGV("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; - } - - aux_data_expected_size_ = U32_AT(buf + parse_offset); - aux_data_in_progress_.clear(); - if (aux_data_in_progress_.capacity() < aux_data_expected_size_) { - aux_data_in_progress_.setCapacity(aux_data_expected_size_); - } - } else { - aux_data_expected_size_ = 0; - } - - if ((aux_data_expected_size_ + trtp_header_len) > trtp_len) { - ALOGV("Expected codec aux data length (%u) and TRTP header overhead" - " (%u) too large for total TRTP payload length (%u).", - aux_data_expected_size_, trtp_header_len, trtp_len); - 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 - - aux_data_expected_size_; - if (!expected_buffer_size_) { - ALOGV("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_ + aux_data_expected_size_) < todo) { - ALOGV("Extra data (%u > %u) present in initial TRTP Audio Payload;" - " dropping payload.", todo, - expected_buffer_size_ + aux_data_expected_size_); - return; - } - - buffer_filled_ = 0; - buffer_in_progress_ = new MediaBuffer(expected_buffer_size_); - if ((NULL == buffer_in_progress_) || - (NULL == buffer_in_progress_->data())) { - ALOGV("Failed to allocate MediaBuffer of length %u", - expected_buffer_size_); - cleanupBufferInProgress(); - return; - } - - sp<MetaData> meta = buffer_in_progress_->meta_data(); - if (meta == NULL) { - ALOGV("Missing metadata structure in allocated MediaBuffer; dropping" - " payload"); - cleanupBufferInProgress(); - return; - } - - meta->setCString(kKeyMIMEType, codec_mime_type_); - if (ts_valid) { - meta->setInt64(kKeyTime, ts); - } - - // Skip over the header we have already extracted. - amt -= trtp_header_len; - buf += trtp_header_len; - - // Extract as much of the expected aux data as we can. - todo = min(aux_data_expected_size_, amt); - if (todo) { - aux_data_in_progress_.appendArray(buf, todo); - buf += todo; - amt -= todo; - } - - // Extract as much of the expected payload as we can. - todo = min(expected_buffer_size_, amt); - if (todo > 0) { - uint8_t* tgt = - reinterpret_cast<uint8_t*>(buffer_in_progress_->data()); - memcpy(tgt, buf, todo); - buffer_filled_ = amt; - buf += todo; - amt -= todo; - } - - if (buffer_filled_ >= expected_buffer_size_) { - processCompletedBuffer(); - } -} - -void AAH_RXPlayer::Substream::processPayloadCont(uint8_t* buf, - uint32_t amt) { - if (shouldAbort(__PRETTY_FUNCTION__)) { - return; - } - - if (NULL == buffer_in_progress_) { - ALOGV("TRTP Receiver skipping payload continuation; no buffer currently" - " in progress."); - return; - } - - CHECK(aux_data_in_progress_.size() <= aux_data_expected_size_); - uint32_t aux_left = aux_data_expected_size_ - aux_data_in_progress_.size(); - if (aux_left) { - uint32_t todo = min(aux_left, amt); - aux_data_in_progress_.appendArray(buf, todo); - amt -= todo; - buf += todo; - - if (!amt) - return; - } - - CHECK(buffer_filled_ < expected_buffer_size_); - uint32_t buffer_left = expected_buffer_size_ - buffer_filled_; - if (amt > buffer_left) { - ALOGV("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() { - status_t res; - - CHECK(NULL != buffer_in_progress_); - - if (decoder_ == NULL) { - ALOGV("Dropping complete buffer, no decoder pump allocated"); - goto bailout; - } - - // Make sure our metadata used to initialize the decoder has been properly - // set up. - if (!setupSubstreamMeta()) - goto bailout; - - // If our decoder has not be set up, do so now. - res = decoder_->init(substream_meta_); - if (OK != res) { - ALOGE("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) { - ALOGD("Failed to queue payload for decode, resetting decoder pump!" - " (res = %d)", res); - status_ = 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(); -} - -bool AAH_RXPlayer::Substream::setupSubstreamMeta() { - switch (codec_type_) { - case TRTPAudioPacket::kCodecMPEG1Audio: - codec_mime_type_ = MEDIA_MIMETYPE_AUDIO_MPEG; - return setupMP3SubstreamMeta(); - - case TRTPAudioPacket::kCodecAACAudio: - codec_mime_type_ = MEDIA_MIMETYPE_AUDIO_AAC; - return setupAACSubstreamMeta(); - - default: - ALOGV("Failed to setup substream metadata for unsupported codec" - " type (%u)", codec_type_); - break; - } - - return false; -} - -bool AAH_RXPlayer::Substream::setupMP3SubstreamMeta() { - const uint8_t* buffer_data = NULL; - int sample_rate; - int channel_count; - size_t frame_size; - status_t res; - - buffer_data = reinterpret_cast<const uint8_t*>(buffer_in_progress_->data()); - if (buffer_in_progress_->size() < 4) { - ALOGV("MP3 payload too short to contain header, dropping payload."); - return false; - } - - // 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)) { - ALOGV("Failed to parse MP3 header in payload, droping payload."); - return false; - } - - - // 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) { - ALOGE("Failed to allocate MetaData structure for MP3 substream"); - return false; - } - - 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)) { - ALOGW("MP3 format change detected, forcing decoder reset."); - cleanupDecoder(); - - substream_meta_->setInt32(kKeyChannelCount, channel_count); - substream_meta_->setInt32(kKeySampleRate, sample_rate); - } - } - - return true; -} - -bool AAH_RXPlayer::Substream::setupAACSubstreamMeta() { - int32_t sample_rate, channel_cnt; - static const size_t overhead = sizeof(sample_rate) - + sizeof(channel_cnt); - - if (aux_data_in_progress_.size() < overhead) { - ALOGE("Not enough aux data (%u) to initialize AAC substream decoder", - aux_data_in_progress_.size()); - return false; - } - - const uint8_t* aux_data = aux_data_in_progress_.array(); - size_t aux_data_size = aux_data_in_progress_.size(); - sample_rate = U32_AT(aux_data); - channel_cnt = U32_AT(aux_data + sizeof(sample_rate)); - - const uint8_t* esds_data = NULL; - size_t esds_data_size = 0; - if (aux_data_size > overhead) { - esds_data = aux_data + overhead; - esds_data_size = aux_data_size - overhead; - } - - // Do we already have metadata? If so, has it changed at all? If not, then - // there should be nothing else to do. Otherwise, release our old stream - // metadata and make new metadata. - if (substream_meta_ != NULL) { - uint32_t type; - const void* data; - size_t size; - int32_t prev_sample_rate; - int32_t prev_channel_count; - bool res; - - res = substream_meta_->findInt32(kKeySampleRate, &prev_sample_rate); - CHECK(res); - res = substream_meta_->findInt32(kKeyChannelCount, &prev_channel_count); - CHECK(res); - - // If nothing has changed about the codec aux data (esds, sample rate, - // channel count), then we can just do nothing and get out. Otherwise, - // we will need to reset the decoder and make a new metadata object to - // deal with the format change. - bool hasData = (esds_data != NULL); - bool hadData = substream_meta_->findData(kKeyESDS, &type, &data, &size); - bool esds_change = (hadData != hasData); - - if (!esds_change && hasData) - esds_change = ((size != esds_data_size) || - memcmp(data, esds_data, size)); - - if (!esds_change && - (prev_sample_rate == sample_rate) && - (prev_channel_count == channel_cnt)) { - return true; // no change, just get out. - } - - ALOGW("AAC format change detected, forcing decoder reset."); - cleanupDecoder(); - substream_meta_ = NULL; - } - - CHECK(substream_meta_ == NULL); - - substream_meta_ = new MetaData(); - if (substream_meta_ == NULL) { - ALOGE("Failed to allocate MetaData structure for AAC substream"); - return false; - } - - substream_meta_->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC); - substream_meta_->setInt32 (kKeySampleRate, sample_rate); - substream_meta_->setInt32 (kKeyChannelCount, channel_cnt); - - if (esds_data) { - substream_meta_->setData(kKeyESDS, kTypeESDS, - esds_data, esds_data_size); - } - - return true; -} - -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 and AAC. - // Also check for conflicts with previously delivered codec types. - if (substream_details_known_) { - if (codec_type != codec_type_) { - ALOGV("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; - } - - return true; - } - - switch (codec_type) { - // MP3 and AAC are all we support right now. - case TRTPAudioPacket::kCodecMPEG1Audio: - case TRTPAudioPacket::kCodecAACAudio: - break; - - default: - ALOGV("RXed TRTP Audio Payload for SSRC=0x%08x with unsupported" - " codec type (%u)", ssrc_, codec_type); - return false; - } - - substream_type_ = substream_type; - codec_type_ = codec_type; - substream_details_known_ = true; - - return true; -} - -} // namespace android |