webrtc/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
henrik.lundin@webrtc.org 612171527e Ensure that NetEq recovers after a large timestamp jump
Before this change it could happen that a large jump in timestamp (a
jump not correlated to wall-clock change) caused the audio to go silent
without recovering. The reason was that all incoming packets after the
jump were considered too old compared to the last decoded packet, and
were deleted. With CL changes two things:

1. If the only available packet in the buffer is an old packet, NetEq
will do Expand instead of immediate reset. This is to avoid that one
late packet triggers a reset.

2. Old packets are discarded only when the decision to decode a packet
has been taken. This is to allow the buffer to grow and eventually
flush if no decodable packet has been found for some time.

This CL also includes a new unit test for this situation.

BUG=3785
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7255 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 08:30:07 +00:00

237 lines
8.4 KiB
C++

/*
* Copyright (c) 2013 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/decision_logic_normal.h"
#include <assert.h>
#include <algorithm>
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/interface/module_common_types.h"
namespace webrtc {
Operations DecisionLogicNormal::GetDecisionSpecialized(
const SyncBuffer& sync_buffer,
const Expand& expand,
int decoder_frame_length,
const RTPHeader* packet_header,
Modes prev_mode,
bool play_dtmf,
bool* reset_decoder) {
assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
// Guard for errors, to avoid getting stuck in error mode.
if (prev_mode == kModeError) {
if (!packet_header) {
return kExpand;
} else {
return kUndefined; // Use kUndefined to flag for a reset.
}
}
uint32_t target_timestamp = sync_buffer.end_timestamp();
uint32_t available_timestamp = 0;
bool is_cng_packet = false;
if (packet_header) {
available_timestamp = packet_header->timestamp;
is_cng_packet =
decoder_database_->IsComfortNoise(packet_header->payloadType);
}
if (is_cng_packet) {
return CngOperation(prev_mode, target_timestamp, available_timestamp);
}
// Handle the case with no packet at all available (except maybe DTMF).
if (!packet_header) {
return NoPacket(play_dtmf);
}
// If the expand period was very long, reset NetEQ since it is likely that the
// sender was restarted.
if (num_consecutive_expands_ > kReinitAfterExpands) {
*reset_decoder = true;
return kNormal;
}
// Check if the required packet is available.
if (target_timestamp == available_timestamp) {
return ExpectedPacketAvailable(prev_mode, play_dtmf);
} else if (IsNewerTimestamp(available_timestamp, target_timestamp)) {
return FuturePacketAvailable(sync_buffer, expand, decoder_frame_length,
prev_mode, target_timestamp,
available_timestamp, play_dtmf);
} else {
// This implies that available_timestamp < target_timestamp, which can
// happen when a new stream or codec is received. Do Expand instead, and
// wait for a newer packet to arrive, or for the buffer to flush (resulting
// in a master reset).
return kExpand;
}
}
Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp) {
// Signed difference between target and available timestamp.
int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
available_timestamp;
int32_t optimal_level_samp =
(delay_manager_->TargetLevel() * packet_length_samples_) >> 8;
int32_t excess_waiting_time_samp = -timestamp_diff - optimal_level_samp;
if (excess_waiting_time_samp > optimal_level_samp / 2) {
// The waiting time for this packet will be longer than 1.5
// times the wanted buffer delay. Advance the clock to cut
// waiting time down to the optimal.
generated_noise_samples_ += excess_waiting_time_samp;
timestamp_diff += excess_waiting_time_samp;
}
if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
// Not time to play this packet yet. Wait another round before using this
// packet. Keep on playing CNG from previous CNG parameters.
return kRfc3389CngNoPacket;
} else {
// Otherwise, go for the CNG packet now.
return kRfc3389Cng;
}
}
Operations DecisionLogicNormal::NoPacket(bool play_dtmf) {
if (cng_state_ == kCngRfc3389On) {
// Keep on playing comfort noise.
return kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
// Keep on playing codec internal comfort noise.
return kCodecInternalCng;
} else if (play_dtmf) {
return kDtmf;
} else {
// Nothing to play, do expand.
return kExpand;
}
}
Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
bool play_dtmf) {
if (prev_mode != kModeExpand && !play_dtmf) {
// Check criterion for time-stretching.
int low_limit, high_limit;
delay_manager_->BufferLimits(&low_limit, &high_limit);
if ((buffer_level_filter_->filtered_current_level() >= high_limit &&
TimescaleAllowed()) ||
buffer_level_filter_->filtered_current_level() >= high_limit << 2) {
// Buffer level higher than limit and time-scaling allowed,
// or buffer level really high.
return kAccelerate;
} else if ((buffer_level_filter_->filtered_current_level() < low_limit)
&& TimescaleAllowed()) {
return kPreemptiveExpand;
}
}
return kNormal;
}
Operations DecisionLogicNormal::FuturePacketAvailable(
const SyncBuffer& sync_buffer,
const Expand& expand,
int decoder_frame_length,
Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
bool play_dtmf) {
// Required packet is not available, but a future packet is.
// Check if we should continue with an ongoing expand because the new packet
// is too far into the future.
uint32_t timestamp_leap = available_timestamp - target_timestamp;
if ((prev_mode == kModeExpand) &&
!ReinitAfterExpands(timestamp_leap) &&
!MaxWaitForPacket() &&
PacketTooEarly(timestamp_leap) &&
UnderTargetLevel()) {
if (play_dtmf) {
// Still have DTMF to play, so do not do expand.
return kDtmf;
} else {
// Nothing to play.
return kExpand;
}
}
const int samples_left = static_cast<int>(sync_buffer.FutureLength() -
expand.overlap_length());
const int cur_size_samples = samples_left +
packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
// If previous was comfort noise, then no merge is needed.
if (prev_mode == kModeRfc3389Cng ||
prev_mode == kModeCodecInternalCng) {
// Keep the same delay as before the CNG (or maximum 70 ms in buffer as
// safety precaution), but make sure that the number of samples in buffer
// is no higher than 4 times the optimal level. (Note that TargetLevel()
// is in Q8.)
int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
available_timestamp;
if (timestamp_diff >= 0 ||
cur_size_samples >
4 * ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8)) {
// Time to play this new packet.
return kNormal;
} else {
// Too early to play this new packet; keep on playing comfort noise.
if (prev_mode == kModeRfc3389Cng) {
return kRfc3389CngNoPacket;
} else { // prevPlayMode == kModeCodecInternalCng.
return kCodecInternalCng;
}
}
}
// Do not merge unless we have done an expand before.
// (Convert kAllowMergeWithoutExpand from ms to samples by multiplying with
// fs_mult_ * 8 = fs / 1000.)
if (prev_mode == kModeExpand ||
(decoder_frame_length < output_size_samples_ &&
cur_size_samples > kAllowMergeWithoutExpandMs * fs_mult_ * 8)) {
return kMerge;
} else if (play_dtmf) {
// Play DTMF instead of expand.
return kDtmf;
} else {
return kExpand;
}
}
bool DecisionLogicNormal::UnderTargetLevel() const {
return buffer_level_filter_->filtered_current_level() <=
delay_manager_->TargetLevel();
}
bool DecisionLogicNormal::ReinitAfterExpands(uint32_t timestamp_leap) const {
return timestamp_leap >=
static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
}
bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
return timestamp_leap >
static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
}
bool DecisionLogicNormal::MaxWaitForPacket() const {
return num_consecutive_expands_ >= kMaxWaitForPacket;
}
} // namespace webrtc