NetEq: Use TickTimer in PacketBuffer
This change makes use of the TickTimer::Stopwatch in Packets. When a
packet is inserted into the PacketBuffer, a Stopwatch object is
attached to it. When the packet is extracted from the buffer, the
Stopwatch is read to know how long the packet waited in the buffer.
BUG=webrtc:5608
Review URL: https://codereview.webrtc.org/1917913002
Cr-Commit-Position: refs/heads/master@{#12508}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 99fa140..3797818 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -791,6 +791,8 @@
"neteq/neteq_impl.h",
"neteq/normal.cc",
"neteq/normal.h",
+ "neteq/packet.cc",
+ "neteq/packet.h",
"neteq/packet_buffer.cc",
"neteq/packet_buffer.h",
"neteq/payload_splitter.cc",
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
index 499f946..ff8a0c4 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -17,6 +17,7 @@
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
namespace webrtc {
@@ -24,7 +25,8 @@
int fs_hz = 8000;
int output_size_samples = fs_hz / 100; // Samples per 10 ms.
DecoderDatabase decoder_database;
- PacketBuffer packet_buffer(10);
+ TickTimer tick_timer;
+ PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector;
DelayManager delay_manager(240, &delay_peak_detector);
BufferLevelFilter buffer_level_filter;
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index 97e54d8..6bb9590 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -19,8 +19,8 @@
class MockPacketBuffer : public PacketBuffer {
public:
- MockPacketBuffer(size_t max_number_of_packets)
- : PacketBuffer(max_number_of_packets) {}
+ MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+ : PacketBuffer(max_number_of_packets, tick_timer) {}
virtual ~MockPacketBuffer() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Flush,
diff --git a/webrtc/modules/audio_coding/neteq/neteq.cc b/webrtc/modules/audio_coding/neteq/neteq.cc
index bc6319d..1af7966 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq.cc
@@ -54,7 +54,8 @@
delay_manager->SetMaximumDelay(config.max_delay_ms);
DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator;
- PacketBuffer* packet_buffer = new PacketBuffer(config.max_packets_in_buffer);
+ PacketBuffer* packet_buffer =
+ new PacketBuffer(config.max_packets_in_buffer, tick_timer.get());
PayloadSplitter* payload_splitter = new PayloadSplitter;
TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database);
AccelerateFactory* accelerate_factory = new AccelerateFactory;
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
index 7e0f558..fee51df 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -105,6 +105,8 @@
'statistics_calculator.h',
'normal.cc',
'normal.h',
+ 'packet.cc',
+ 'packet.h',
'packet_buffer.cc',
'packet_buffer.h',
'payload_splitter.cc',
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index ef470e5..cca1c4c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -536,7 +536,8 @@
packet->header.numCSRCs = 0;
packet->payload_length = payload.size();
packet->primary = true;
- packet->waiting_time = 0;
+ // Waiting time will be set upon inserting the packet in the buffer.
+ RTC_DCHECK(!packet->waiting_time);
packet->payload = new uint8_t[packet->payload_length];
packet->sync_packet = is_sync_packet;
if (!packet->payload) {
@@ -1002,7 +1003,6 @@
*operation = kUndefined;
// Increment time counters.
- packet_buffer_->IncrementWaitingTimes();
stats_.IncreaseCounter(output_size_samples_, fs_hz_);
assert(sync_buffer_.get());
@@ -1931,8 +1931,7 @@
return -1;
}
stats_.PacketsDiscarded(discard_count);
- // Store waiting time in ms; packets->waiting_time is in "output blocks".
- stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
+ stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
assert(packet->payload_length > 0);
packet_list->push_back(packet); // Store packet in list.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index e95522b..4433d20 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -130,10 +130,12 @@
dtmf_tone_generator_ = new DtmfToneGenerator;
}
if (use_mock_packet_buffer_) {
- mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer);
+ mock_packet_buffer_ =
+ new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_);
packet_buffer_ = mock_packet_buffer_;
} else {
- packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer);
+ packet_buffer_ =
+ new PacketBuffer(config_.max_packets_in_buffer, tick_timer_);
}
if (use_mock_payload_splitter_) {
mock_payload_splitter_ = new MockPayloadSplitter;
diff --git a/webrtc/modules/audio_coding/neteq/packet.cc b/webrtc/modules/audio_coding/neteq/packet.cc
new file mode 100644
index 0000000..8a19fe4
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/packet.cc
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+Packet::Packet() = default;
+
+Packet::~Packet() = default;
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h
index 64b325e..d6f64c7 100644
--- a/webrtc/modules/audio_coding/neteq/packet.h
+++ b/webrtc/modules/audio_coding/neteq/packet.h
@@ -12,7 +12,9 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
#include <list>
+#include <memory>
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
@@ -21,20 +23,15 @@
// Struct for holding RTP packets.
struct Packet {
RTPHeader header;
- uint8_t* payload; // Datagram excluding RTP header and header extension.
- size_t payload_length;
- bool primary; // Primary, i.e., not redundant payload.
- int waiting_time;
- bool sync_packet;
+ // Datagram excluding RTP header and header extension.
+ uint8_t* payload = nullptr;
+ size_t payload_length = 0;
+ bool primary = true; // Primary, i.e., not redundant payload.
+ bool sync_packet = false;
+ std::unique_ptr<TickTimer::Stopwatch> waiting_time;
- // Constructor.
- Packet()
- : payload(NULL),
- payload_length(0),
- primary(true),
- waiting_time(0),
- sync_packet(false) {
- }
+ Packet();
+ ~Packet();
// Comparison operators. Establish a packet ordering based on (1) timestamp,
// (2) sequence number, (3) regular packet vs sync-packet and (4) redundancy.
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index c89de12..f1b898e 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
namespace webrtc {
@@ -37,8 +38,9 @@
const Packet* new_packet_;
};
-PacketBuffer::PacketBuffer(size_t max_number_of_packets)
- : max_number_of_packets_(max_number_of_packets) {}
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
+ const TickTimer* tick_timer)
+ : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
// Destructor. All packets in the buffer will be destroyed.
PacketBuffer::~PacketBuffer() {
@@ -65,6 +67,8 @@
int return_val = kOK;
+ packet->waiting_time = tick_timer_->GetNewStopwatch();
+
if (buffer_.size() >= max_number_of_packets_) {
// Buffer is full. Flush it.
Flush();
@@ -268,13 +272,6 @@
return num_samples;
}
-void PacketBuffer::IncrementWaitingTimes(int inc) {
- PacketList::iterator it;
- for (it = buffer_.begin(); it != buffer_.end(); ++it) {
- (*it)->waiting_time += inc;
- }
-}
-
bool PacketBuffer::DeleteFirstPacket(PacketList* packet_list) {
if (packet_list->empty()) {
return false;
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.h b/webrtc/modules/audio_coding/neteq/packet_buffer.h
index 03c11e6..6867b4c 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -17,8 +17,8 @@
namespace webrtc {
-// Forward declaration.
class DecoderDatabase;
+class TickTimer;
// This is the actual buffer holding the packets before decoding.
class PacketBuffer {
@@ -34,7 +34,7 @@
// Constructor creates a buffer which can hold a maximum of
// |max_number_of_packets| packets.
- PacketBuffer(size_t max_number_of_packets);
+ PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
// Deletes all packets in the buffer before destroying the buffer.
virtual ~PacketBuffer();
@@ -116,10 +116,6 @@
virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
size_t last_decoded_length) const;
- // Increase the waiting time counter for every packet in the buffer by |inc|.
- // The default value for |inc| is 1.
- virtual void IncrementWaitingTimes(int inc = 1);
-
virtual void BufferStat(int* num_packets, int* max_num_packets) const;
// Static method that properly deletes the first packet, and its payload
@@ -148,6 +144,7 @@
private:
size_t max_number_of_packets_;
PacketList buffer_;
+ const TickTimer* tick_timer_;
RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
};
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 435b6c8..da35301 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -16,6 +16,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
using ::testing::Return;
using ::testing::_;
@@ -80,13 +81,15 @@
// Start of test definitions.
TEST(PacketBuffer, CreateAndDestroy) {
- PacketBuffer* buffer = new PacketBuffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer* buffer = new PacketBuffer(10, &tick_timer); // 10 packets.
EXPECT_TRUE(buffer->Empty());
delete buffer;
}
TEST(PacketBuffer, InsertPacket) {
- PacketBuffer buffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(10, &tick_timer); // 10 packets.
PacketGenerator gen(17u, 4711u, 0, 10);
const int payload_len = 100;
@@ -107,7 +110,8 @@
// Test to flush buffer.
TEST(PacketBuffer, FlushBuffer) {
- PacketBuffer buffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(10, &tick_timer); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
const int payload_len = 10;
@@ -127,7 +131,8 @@
// Test to fill the buffer over the limits, and verify that it flushes.
TEST(PacketBuffer, OverfillBuffer) {
- PacketBuffer buffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(10, &tick_timer); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
// Insert 10 small packets; should be ok.
@@ -156,7 +161,8 @@
// Test inserting a list of packets.
TEST(PacketBuffer, InsertPacketList) {
- PacketBuffer buffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(10, &tick_timer); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
PacketList list;
const int payload_len = 10;
@@ -192,7 +198,8 @@
// Expecting the buffer to flush.
// TODO(hlundin): Remove this test when legacy operation is no longer needed.
TEST(PacketBuffer, InsertPacketListChangePayloadType) {
- PacketBuffer buffer(10); // 10 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(10, &tick_timer); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
PacketList list;
const int payload_len = 10;
@@ -230,7 +237,8 @@
}
TEST(PacketBuffer, ExtractOrderRedundancy) {
- PacketBuffer buffer(100); // 100 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(100, &tick_timer); // 100 packets.
const int kPackets = 18;
const int kFrameSize = 10;
const int kPayloadLength = 10;
@@ -289,7 +297,8 @@
}
TEST(PacketBuffer, DiscardPackets) {
- PacketBuffer buffer(100); // 100 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(100, &tick_timer); // 100 packets.
const uint16_t start_seq_no = 17;
const uint32_t start_ts = 4711;
const uint32_t ts_increment = 10;
@@ -318,7 +327,8 @@
}
TEST(PacketBuffer, Reordering) {
- PacketBuffer buffer(100); // 100 packets.
+ TickTimer tick_timer;
+ PacketBuffer buffer(100, &tick_timer); // 100 packets.
const uint16_t start_seq_no = 17;
const uint32_t start_ts = 4711;
const uint32_t ts_increment = 10;
@@ -373,8 +383,9 @@
const uint32_t ts_increment = 10;
int payload_len = 100;
PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+ TickTimer tick_timer;
- PacketBuffer* buffer = new PacketBuffer(100); // 100 packets.
+ PacketBuffer* buffer = new PacketBuffer(100, &tick_timer); // 100 packets.
Packet* packet = NULL;
EXPECT_EQ(PacketBuffer::kInvalidPacket, buffer->InsertPacket(packet));
packet = gen.NextPacket(payload_len);
@@ -404,7 +415,7 @@
// Insert packet list of three packets, where the second packet has an invalid
// payload. Expect first packet to be inserted, and the remaining two to be
// discarded.
- buffer = new PacketBuffer(100); // 100 packets.
+ buffer = new PacketBuffer(100, &tick_timer); // 100 packets.
PacketList list;
list.push_back(gen.NextPacket(payload_len)); // Valid packet.
packet = gen.NextPacket(payload_len);
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter.cc b/webrtc/modules/audio_coding/neteq/payload_splitter.cc
index 5428639..530e9d0 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter.cc
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter.cc
@@ -12,6 +12,7 @@
#include <assert.h>
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
@@ -168,8 +169,9 @@
memcpy(new_packet->payload, packet->payload, packet->payload_length);
new_packet->payload_length = packet->payload_length;
new_packet->primary = false;
- new_packet->waiting_time = packet->waiting_time;
new_packet->sync_packet = packet->sync_packet;
+ // Waiting time should not be set here.
+ RTC_DCHECK(!packet->waiting_time);
packet_list->insert(it, new_packet);
break;