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Add PeerConnection option to configure minimum audio jitter buffer delay.

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Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
This commit is contained in:
Jakob Ivarsson
2018-11-27 15:45:20 +01:00
committed by Commit Bot
parent c7f1a0af92
commit 10403ae87c
18 changed files with 66 additions and 13 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_coding/neteq/decision_logic_unittest.cc
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@ -31,7 +31,7 @@ TEST(DecisionLogic, CreateAndDestroy) {
TickTimer tick_timer;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer);
DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(
fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
@ -48,7 +48,7 @@ TEST(DecisionLogic, PostponeDecodingAfterExpansionSettings) {
TickTimer tick_timer;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer);
DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
{
test::ScopedFieldTrials field_trial(

8
modules/audio_coding/neteq/delay_manager.cc
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@ -62,6 +62,7 @@ absl::optional<int> GetForcedLimitProbability() {
namespace webrtc {
DelayManager::DelayManager(size_t max_packets_in_buffer,
int base_min_target_delay_ms,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer)
: first_packet_received_(false),
@ -69,13 +70,14 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
iat_vector_(kMaxIat + 1, 0),
iat_factor_(0),
tick_timer_(tick_timer),
base_min_target_delay_ms_(base_min_target_delay_ms),
base_target_level_(4), // In Q0 domain.
target_level_(base_target_level_ << 8), // In Q8 domain.
packet_len_ms_(0),
streaming_mode_(false),
last_seq_no_(0),
last_timestamp_(0),
minimum_delay_ms_(0),
minimum_delay_ms_(base_min_target_delay_ms_),
maximum_delay_ms_(target_level_),
iat_cumulative_sum_(0),
max_iat_cumulative_sum_(0),
@ -85,6 +87,8 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
forced_limit_probability_(GetForcedLimitProbability()) {
assert(peak_detector); // Should never be NULL.
RTC_DCHECK_GE(base_min_target_delay_ms_, 0);
RTC_DCHECK_LE(minimum_delay_ms_, maximum_delay_ms_);
Reset();
}
@ -485,7 +489,7 @@ bool DelayManager::SetMinimumDelay(int delay_ms) {
static_cast<int>(3 * max_packets_in_buffer_ * packet_len_ms_ / 4))) {
return false;
}
minimum_delay_ms_ = delay_ms;
minimum_delay_ms_ = std::max(delay_ms, base_min_target_delay_ms_);
return true;
}

8
modules/audio_coding/neteq/delay_manager.h
View File

@ -31,9 +31,11 @@ class DelayManager {
// Create a DelayManager object. Notify the delay manager that the packet
// buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
// is the number of packet slots in the buffer). Supply a PeakDetector
// object to the DelayManager.
// is the number of packet slots in the buffer) and that the target delay
// should be greater than or equal to |base_min_target_delay_ms|. Supply a
// PeakDetector object to the DelayManager.
DelayManager(size_t max_packets_in_buffer,
int base_min_target_delay_ms,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer);
@ -144,6 +146,8 @@ class DelayManager {
IATVector iat_vector_; // Histogram of inter-arrival times.
int iat_factor_; // Forgetting factor for updating the IAT histogram (Q15).
const TickTimer* tick_timer_;
const int base_min_target_delay_ms_; // Lower bound for target_level_ and
// minimum_delay_ms_.
// Time elapsed since last packet.
std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
int base_target_level_; // Currently preferred buffer level before peak

4
modules/audio_coding/neteq/delay_manager_unittest.cc
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@ -27,6 +27,7 @@ using ::testing::_;
class DelayManagerTest : public ::testing::Test {
protected:
static const int kMaxNumberOfPackets = 240;
static const int kMinDelayMs = 0;
static const int kTimeStepMs = 10;
static const int kFs = 8000;
static const int kFrameSizeMs = 20;
@ -56,7 +57,8 @@ void DelayManagerTest::SetUp() {
void DelayManagerTest::RecreateDelayManager() {
EXPECT_CALL(detector_, Reset()).Times(1);
dm_.reset(new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_));
dm_.reset(new DelayManager(kMaxNumberOfPackets, kMinDelayMs, &detector_,
&tick_timer_));
}
void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {

1
modules/audio_coding/neteq/include/neteq.h
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@ -113,6 +113,7 @@ class NetEq {
bool enable_post_decode_vad = false;
size_t max_packets_in_buffer = 50;
int max_delay_ms = 2000;
int min_delay_ms = 0;
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
absl::optional<AudioCodecPairId> codec_pair_id;

6
modules/audio_coding/neteq/mock/mock_delay_manager.h
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@ -20,9 +20,13 @@ namespace webrtc {
class MockDelayManager : public DelayManager {
public:
MockDelayManager(size_t max_packets_in_buffer,
int base_min_target_delay_ms,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer)
: DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
: DelayManager(max_packets_in_buffer,
base_min_target_delay_ms,
peak_detector,
tick_timer) {}
virtual ~MockDelayManager() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_CONST_METHOD0(iat_vector, const IATVector&());

1
modules/audio_coding/neteq/neteq_impl.cc
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@ -63,6 +63,7 @@ NetEqImpl::Dependencies::Dependencies(
new DecoderDatabase(decoder_factory, config.codec_pair_id)),
delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
delay_manager(new DelayManager(config.max_packets_in_buffer,
config.min_delay_ms,
delay_peak_detector.get(),
tick_timer.get())),
dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),

3
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -92,7 +92,8 @@ class NetEqImplTest : public ::testing::Test {
if (use_mock_delay_manager_) {
std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
config_.max_packets_in_buffer, delay_peak_detector_, tick_timer_));
config_.max_packets_in_buffer, config_.min_delay_ms,
delay_peak_detector_, tick_timer_));
mock_delay_manager_ = mock.get();
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
deps.delay_manager = std::move(mock);