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Added a ParsePayload method to AudioDecoder.

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It allows the decoder to split the input up into usable chunks before
they are put into NetEq's PacketBuffer. Eventually, all packet splitting
will move into ParsePayload.

There's currently a base implementation of ParsePayload. It will
generate a single Frame that calls the underlying AudioDecoder for
getting Duration() and to Decode.

BUG=webrtc:5805
BUG=chromium:428099

Review-Url: https://codereview.webrtc.org/2326953003
Cr-Commit-Position: refs/heads/master@{#14300}
This commit is contained in:
ossu
2016-09-20 01:38:00 -07:00
committed by Commit bot
parent 02bd5125e9
commit 61a208b1b8
8 changed files with 268 additions and 115 deletions
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36
webrtc/modules/audio_coding/neteq/packet_buffer.cc
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@ -68,7 +68,7 @@ bool PacketBuffer::Empty() const {
}
int PacketBuffer::InsertPacket(Packet* packet) {
if (!packet || packet->payload.empty()) {
if (!packet || packet->empty()) {
if (packet) {
delete packet;
}
@ -209,7 +209,7 @@ Packet* PacketBuffer::GetNextPacket(size_t* discard_count) {
Packet* packet = buffer_.front();
// Assert that the packet sanity checks in InsertPacket method works.
assert(packet && !packet->payload.empty());
RTC_DCHECK(packet && !packet->empty());
buffer_.pop_front();
// Discard other packets with the same timestamp. These are duplicates or
@ -237,8 +237,8 @@ int PacketBuffer::DiscardNextPacket() {
return kBufferEmpty;
}
// Assert that the packet sanity checks in InsertPacket method works.
assert(buffer_.front());
assert(!buffer_.front()->payload.empty());
RTC_DCHECK(buffer_.front());
RTC_DCHECK(!buffer_.front()->empty());
DeleteFirstPacket(&buffer_);
return kOK;
}
@ -260,26 +260,32 @@ int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
return DiscardOldPackets(timestamp_limit, 0);
}
void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type) {
for (auto it = buffer_.begin(); it != buffer_.end(); /* */) {
Packet *packet = *it;
if (packet->header.payloadType == payload_type) {
delete packet;
it = buffer_.erase(it);
} else {
++it;
}
}
}
size_t PacketBuffer::NumPacketsInBuffer() const {
return buffer_.size();
}
size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
size_t last_decoded_length) const {
PacketList::const_iterator it;
size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
size_t num_samples = 0;
size_t last_duration = last_decoded_length;
for (it = buffer_.begin(); it != buffer_.end(); ++it) {
Packet* packet = (*it);
AudioDecoder* decoder =
decoder_database->GetDecoder(packet->header.payloadType);
if (decoder) {
for (Packet* packet : buffer_) {
if (packet->frame) {
if (!packet->primary) {
continue;
}
int duration = decoder->PacketDuration(packet->payload.data(),
packet->payload.size());
if (duration >= 0) {
size_t duration = packet->frame->Duration();
if (duration > 0) {
last_duration = duration; // Save the most up-to-date (valid) duration.
}
}