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/*
* 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 "modules/audio_coding/neteq/decision_logic.h"
#include <algorithm>
#include "modules/audio_coding/neteq/buffer_level_filter.h"
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:28 +00:00
#include "modules/audio_coding/neteq/decision_logic_fax.h"
#include "modules/audio_coding/neteq/decision_logic_normal.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
DecisionLogic* DecisionLogic::Create(int fs_hz,
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t output_size_samples,
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:28 +00:00
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter,
const TickTimer* tick_timer) {
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:28 +00:00
switch (playout_mode) {
case kPlayoutOn:
case kPlayoutStreaming:
return new DecisionLogicNormal(
fs_hz, output_size_samples, playout_mode, decoder_database,
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
case kPlayoutFax:
case kPlayoutOff:
return new DecisionLogicFax(
fs_hz, output_size_samples, playout_mode, decoder_database,
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
}
// This line cannot be reached, but must be here to avoid compiler errors.
assert(false);
return NULL;
}
DecisionLogic::DecisionLogic(int fs_hz,
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t output_size_samples,
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:28 +00:00
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter,
const TickTimer* tick_timer)
: decoder_database_(decoder_database),
packet_buffer_(packet_buffer),
delay_manager_(delay_manager),
buffer_level_filter_(buffer_level_filter),
tick_timer_(tick_timer),
cng_state_(kCngOff),
packet_length_samples_(0),
sample_memory_(0),
prev_time_scale_(false),
timescale_countdown_(
tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
num_consecutive_expands_(0),
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:28 +00:00
playout_mode_(playout_mode) {
delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
SetSampleRate(fs_hz, output_size_samples);
}
DecisionLogic::~DecisionLogic() = default;
void DecisionLogic::Reset() {
cng_state_ = kCngOff;
noise_fast_forward_ = 0;
packet_length_samples_ = 0;
sample_memory_ = 0;
prev_time_scale_ = false;
timescale_countdown_.reset();
num_consecutive_expands_ = 0;
}
void DecisionLogic::SoftReset() {
packet_length_samples_ = 0;
sample_memory_ = 0;
prev_time_scale_ = false;
timescale_countdown_ =
tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
fs_mult_ = fs_hz / 8000;
output_size_samples_ = output_size_samples;
}
Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
const Expand& expand,
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t decoder_frame_length,
const Packet* next_packet,
Modes prev_mode,
bool play_dtmf,
size_t generated_noise_samples,
bool* reset_decoder) {
// If last mode was CNG (or Expand, since this could be covering up for
// a lost CNG packet), remember that CNG is on. This is needed if comfort
// noise is interrupted by DTMF.
if (prev_mode == kModeRfc3389Cng) {
cng_state_ = kCngRfc3389On;
} else if (prev_mode == kModeCodecInternalCng) {
cng_state_ = kCngInternalOn;
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
const size_t samples_left =
sync_buffer.FutureLength() - expand.overlap_length();
const size_t cur_size_samples =
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
prev_time_scale_ =
prev_time_scale_ && (prev_mode == kModeAccelerateSuccess ||
prev_mode == kModeAccelerateLowEnergy ||
prev_mode == kModePreemptiveExpandSuccess ||
prev_mode == kModePreemptiveExpandLowEnergy);
FilterBufferLevel(cur_size_samples, prev_mode);
return GetDecisionSpecialized(
sync_buffer, expand, decoder_frame_length, next_packet, prev_mode,
play_dtmf, reset_decoder, generated_noise_samples, cur_size_samples);
}
void DecisionLogic::ExpandDecision(Operations operation) {
if (operation == kExpand) {
num_consecutive_expands_++;
} else {
num_consecutive_expands_ = 0;
}
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
Modes prev_mode) {
// Do not update buffer history if currently playing CNG since it will bias
// the filtered buffer level.
if ((prev_mode != kModeRfc3389Cng) && (prev_mode != kModeCodecInternalCng)) {
buffer_level_filter_->SetTargetBufferLevel(
delay_manager_->base_target_level());
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t buffer_size_packets = 0;
if (packet_length_samples_ > 0) {
// Calculate size in packets.
buffer_size_packets = buffer_size_samples / packet_length_samples_;
}
int sample_memory_local = 0;
if (prev_time_scale_) {
sample_memory_local = sample_memory_;
timescale_countdown_ =
tick_timer_->GetNewCountdown(kMinTimescaleInterval);
}
buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
packet_length_samples_);
prev_time_scale_ = false;
}
}
} // namespace webrtc