update neteq 4 to facilitate NACK
BUG=
R=turaj@webrtc.org, turajs@google.com
Review URL: https://webrtc-codereview.appspot.com/2008004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4637 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/interface/neteq.h b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
index 824b17b..1f1ee67 100644
--- a/webrtc/modules/audio_coding/neteq4/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
@@ -233,6 +233,10 @@
int* current_memory_size_bytes,
int* max_memory_size_bytes) const = 0;
+ // Get sequence number and timestamp of the latest RTP.
+ // This method is to facilitate NACK.
+ virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) = 0;
+
protected:
NetEq() {}
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 403fccb..279401a 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -87,7 +87,9 @@
first_packet_(true),
error_code_(0),
decoder_error_code_(0),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
+ crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+ decoded_packet_sequence_number_(-1),
+ decoded_packet_timestamp_(0) {
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " <<
"Changing to 8000 Hz.";
@@ -352,6 +354,15 @@
first_packet_ = true;
}
+int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) {
+ CriticalSectionScoped lock(crit_sect_);
+ if (decoded_packet_sequence_number_ < 0)
+ return -1;
+ *sequence_number = decoded_packet_sequence_number_;
+ *timestamp = decoded_packet_timestamp_;
+ return 0;
+}
+
// Methods below this line are private.
@@ -1662,8 +1673,9 @@
if (first_packet) {
first_packet = false;
- prev_sequence_number = packet->header.sequenceNumber;
- prev_timestamp = packet->header.timestamp;
+ decoded_packet_sequence_number_ = prev_sequence_number =
+ packet->header.sequenceNumber;
+ decoded_packet_timestamp_ = prev_timestamp = packet->header.timestamp;
prev_payload_type = packet->header.payloadType;
}
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
index 0d81208..56dde77 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
@@ -167,6 +167,10 @@
int* current_memory_size_bytes,
int* max_memory_size_bytes) const;
+ // Get sequence number and timestamp of the latest RTP.
+ // This method is to facilitate NACK.
+ virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp);
+
private:
static const int kOutputSizeMs = 10;
static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
@@ -318,6 +322,16 @@
int decoder_error_code_;
CriticalSectionWrapper* crit_sect_;
+ // These values are used by NACK module to estimate time-to-play of
+ // a missing packet. Occasionally, NetEq might decide to decode more
+ // than one packet. Therefore, these values store sequence number and
+ // timestamp of the first packet pulled from the packet buffer. In
+ // such cases, these values do not exactly represent the sequence number
+ // or timestamp associated with a 10ms audio pulled from NetEq. NACK
+ // module is designed to compensate for this.
+ int decoded_packet_sequence_number_;
+ uint32_t decoded_packet_timestamp_;
+
DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
};