Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

443 lines
15 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_processing/agc/agc_manager_direct.h"
#include <cassert>
#include <cmath>
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <cstdio>
#endif
#include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
#include "webrtc/modules/audio_processing/gain_control_impl.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
namespace {
// Lowest the microphone level can be lowered due to clipping.
const int kClippedLevelMin = 170;
// Amount the microphone level is lowered with every clipping event.
const int kClippedLevelStep = 15;
// Proportion of clipped samples required to declare a clipping event.
const float kClippedRatioThreshold = 0.1f;
// Time in frames to wait after a clipping event before checking again.
const int kClippedWaitFrames = 300;
// Amount of error we tolerate in the microphone level (presumably due to OS
// quantization) before we assume the user has manually adjusted the microphone.
const int kLevelQuantizationSlack = 25;
const int kDefaultCompressionGain = 7;
const int kMaxCompressionGain = 12;
const int kMinCompressionGain = 2;
// Controls the rate of compression changes towards the target.
const float kCompressionGainStep = 0.05f;
const int kMaxMicLevel = 255;
static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
const int kMinMicLevel = 12;
// Prevent very large microphone level changes.
const int kMaxResidualGainChange = 15;
// Maximum additional gain allowed to compensate for microphone level
// restrictions from clipping events.
const int kSurplusCompressionGain = 6;
int ClampLevel(int mic_level) {
return std::min(std::max(kMinMicLevel, mic_level), kMaxMicLevel);
}
int LevelFromGainError(int gain_error, int level) {
assert(level >= 0 && level <= kMaxMicLevel);
if (gain_error == 0) {
return level;
}
// TODO(ajm): Could be made more efficient with a binary search.
int new_level = level;
if (gain_error > 0) {
while (kGainMap[new_level] - kGainMap[level] < gain_error &&
new_level < kMaxMicLevel) {
++new_level;
}
} else {
while (kGainMap[new_level] - kGainMap[level] > gain_error &&
new_level > kMinMicLevel) {
--new_level;
}
}
return new_level;
}
} // namespace
// Facility for dumping debug audio files. All methods are no-ops in the
// default case where WEBRTC_AGC_DEBUG_DUMP is undefined.
class DebugFile {
#ifdef WEBRTC_AGC_DEBUG_DUMP
public:
explicit DebugFile(const char* filename)
: file_(fopen(filename, "wb")) {
assert(file_);
}
~DebugFile() {
fclose(file_);
}
void Write(const int16_t* data, size_t length_samples) {
fwrite(data, 1, length_samples * sizeof(int16_t), file_);
}
private:
FILE* file_;
#else
public:
explicit DebugFile(const char* filename) {
}
~DebugFile() {
}
void Write(const int16_t* data, size_t length_samples) {
}
#endif // WEBRTC_AGC_DEBUG_DUMP
};
AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level)
: agc_(new Agc()),
gctrl_(gctrl),
volume_callbacks_(volume_callbacks),
frames_since_clipped_(kClippedWaitFrames),
level_(0),
max_level_(kMaxMicLevel),
max_compression_gain_(kMaxCompressionGain),
target_compression_(kDefaultCompressionGain),
compression_(target_compression_),
compression_accumulator_(compression_),
capture_muted_(false),
check_volume_on_next_process_(true), // Check at startup.
startup_(true),
startup_min_level_(ClampLevel(startup_min_level)),
file_preproc_(new DebugFile("agc_preproc.pcm")),
file_postproc_(new DebugFile("agc_postproc.pcm")) {
}
AgcManagerDirect::AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level)
: agc_(agc),
gctrl_(gctrl),
volume_callbacks_(volume_callbacks),
frames_since_clipped_(kClippedWaitFrames),
level_(0),
max_level_(kMaxMicLevel),
max_compression_gain_(kMaxCompressionGain),
target_compression_(kDefaultCompressionGain),
compression_(target_compression_),
compression_accumulator_(compression_),
capture_muted_(false),
check_volume_on_next_process_(true), // Check at startup.
startup_(true),
startup_min_level_(ClampLevel(startup_min_level)),
file_preproc_(new DebugFile("agc_preproc.pcm")),
file_postproc_(new DebugFile("agc_postproc.pcm")) {
}
AgcManagerDirect::~AgcManagerDirect() {}
int AgcManagerDirect::Initialize() {
max_level_ = kMaxMicLevel;
max_compression_gain_ = kMaxCompressionGain;
target_compression_ = kDefaultCompressionGain;
compression_ = target_compression_;
compression_accumulator_ = compression_;
capture_muted_ = false;
check_volume_on_next_process_ = true;
// TODO(bjornv): Investigate if we need to reset |startup_| as well. For
// example, what happens when we change devices.
if (gctrl_->set_mode(GainControl::kFixedDigital) != 0) {
LOG_FERR1(LS_ERROR, set_mode, GainControl::kFixedDigital);
return -1;
}
if (gctrl_->set_target_level_dbfs(2) != 0) {
LOG_FERR1(LS_ERROR, set_target_level_dbfs, 2);
return -1;
}
if (gctrl_->set_compression_gain_db(kDefaultCompressionGain) != 0) {
LOG_FERR1(LS_ERROR, set_compression_gain_db, kDefaultCompressionGain);
return -1;
}
if (gctrl_->enable_limiter(true) != 0) {
LOG_FERR1(LS_ERROR, enable_limiter, true);
return -1;
}
return 0;
}
void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
int num_channels,
size_t samples_per_channel) {
size_t length = num_channels * samples_per_channel;
if (capture_muted_) {
return;
}
file_preproc_->Write(audio, length);
if (frames_since_clipped_ < kClippedWaitFrames) {
++frames_since_clipped_;
return;
}
// Check for clipped samples, as the AGC has difficulty detecting pitch
// under clipping distortion. We do this in the preprocessing phase in order
// to catch clipped echo as well.
//
// If we find a sufficiently clipped frame, drop the current microphone level
// and enforce a new maximum level, dropped the same amount from the current
// maximum. This harsh treatment is an effort to avoid repeated clipped echo
// events. As compensation for this restriction, the maximum compression
// gain is increased, through SetMaxLevel().
float clipped_ratio = agc_->AnalyzePreproc(audio, length);
if (clipped_ratio > kClippedRatioThreshold) {
LOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
<< clipped_ratio;
// Always decrease the maximum level, even if the current level is below
// threshold.
SetMaxLevel(std::max(kClippedLevelMin, max_level_ - kClippedLevelStep));
if (level_ > kClippedLevelMin) {
// Don't try to adjust the level if we're already below the limit. As
// a consequence, if the user has brought the level above the limit, we
// will still not react until the postproc updates the level.
SetLevel(std::max(kClippedLevelMin, level_ - kClippedLevelStep));
// Reset the AGC since the level has changed.
agc_->Reset();
}
frames_since_clipped_ = 0;
}
}
void AgcManagerDirect::Process(const int16_t* audio,
size_t length,
int sample_rate_hz) {
if (capture_muted_) {
return;
}
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
// We have to wait until the first process call to check the volume,
// because Chromium doesn't guarantee it to be valid any earlier.
CheckVolumeAndReset();
}
if (agc_->Process(audio, length, sample_rate_hz) != 0) {
LOG_FERR0(LS_ERROR, Agc::Process);
assert(false);
}
UpdateGain();
UpdateCompressor();
file_postproc_->Write(audio, length);
}
void AgcManagerDirect::SetLevel(int new_level) {
int voe_level = volume_callbacks_->GetMicVolume();
if (voe_level < 0) {
return;
}
if (voe_level == 0) {
LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action.";
return;
}
if (voe_level > kMaxMicLevel) {
LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level=" << voe_level;
return;
}
if (voe_level > level_ + kLevelQuantizationSlack ||
voe_level < level_ - kLevelQuantizationSlack) {
LOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
<< "stored level from " << level_ << " to " << voe_level;
level_ = voe_level;
// Always allow the user to increase the volume.
if (level_ > max_level_) {
SetMaxLevel(level_);
}
// Take no action in this case, since we can't be sure when the volume
// was manually adjusted. The compressor will still provide some of the
// desired gain change.
agc_->Reset();
return;
}
new_level = std::min(new_level, max_level_);
if (new_level == level_) {
return;
}
volume_callbacks_->SetMicVolume(new_level);
LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
<< "level_=" << level_ << ", "
<< "new_level=" << new_level;
level_ = new_level;
}
void AgcManagerDirect::SetMaxLevel(int level) {
assert(level >= kClippedLevelMin);
max_level_ = level;
// Scale the |kSurplusCompressionGain| linearly across the restricted
// level range.
max_compression_gain_ = kMaxCompressionGain + std::floor(
(1.f * kMaxMicLevel - max_level_) / (kMaxMicLevel - kClippedLevelMin) *
kSurplusCompressionGain + 0.5f);
LOG(LS_INFO) << "[agc] max_level_=" << max_level_
<< ", max_compression_gain_=" << max_compression_gain_;
}
void AgcManagerDirect::SetCaptureMuted(bool muted) {
if (capture_muted_ == muted) {
return;
}
capture_muted_ = muted;
if (!muted) {
// When we unmute, we should reset things to be safe.
check_volume_on_next_process_ = true;
}
}
float AgcManagerDirect::voice_probability() {
return agc_->voice_probability();
}
int AgcManagerDirect::CheckVolumeAndReset() {
int level = volume_callbacks_->GetMicVolume();
if (level < 0) {
return -1;
}
// Reasons for taking action at startup:
// 1) A person starting a call is expected to be heard.
// 2) Independent of interpretation of |level| == 0 we should raise it so the
// AGC can do its job properly.
if (level == 0 && !startup_) {
LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action.";
return 0;
}
if (level > kMaxMicLevel) {
LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level=" << level;
return -1;
}
LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
int minLevel = startup_ ? startup_min_level_ : kMinMicLevel;
if (level < minLevel) {
level = minLevel;
LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
volume_callbacks_->SetMicVolume(level);
}
agc_->Reset();
level_ = level;
startup_ = false;
return 0;
}
// Requests the RMS error from AGC and distributes the required gain change
// between the digital compression stage and volume slider. We use the
// compressor first, providing a slack region around the current slider
// position to reduce movement.
//
// If the slider needs to be moved, we check first if the user has adjusted
// it, in which case we take no action and cache the updated level.
void AgcManagerDirect::UpdateGain() {
int rms_error = 0;
if (!agc_->GetRmsErrorDb(&rms_error)) {
// No error update ready.
return;
}
// The compressor will always add at least kMinCompressionGain. In effect,
// this adjusts our target gain upward by the same amount and rms_error
// needs to reflect that.
rms_error += kMinCompressionGain;
// Handle as much error as possible with the compressor first.
int raw_compression = std::max(std::min(rms_error, max_compression_gain_),
kMinCompressionGain);
// Deemphasize the compression gain error. Move halfway between the current
// target and the newly received target. This serves to soften perceptible
// intra-talkspurt adjustments, at the cost of some adaptation speed.
if ((raw_compression == max_compression_gain_ &&
target_compression_ == max_compression_gain_ - 1) ||
(raw_compression == kMinCompressionGain &&
target_compression_ == kMinCompressionGain + 1)) {
// Special case to allow the target to reach the endpoints of the
// compression range. The deemphasis would otherwise halt it at 1 dB shy.
target_compression_ = raw_compression;
} else {
target_compression_ = (raw_compression - target_compression_) / 2
+ target_compression_;
}
// Residual error will be handled by adjusting the volume slider. Use the
// raw rather than deemphasized compression here as we would otherwise
// shrink the amount of slack the compressor provides.
int residual_gain = rms_error - raw_compression;
residual_gain = std::min(std::max(residual_gain, -kMaxResidualGainChange),
kMaxResidualGainChange);
LOG(LS_INFO) << "[agc] rms_error=" << rms_error << ", "
<< "target_compression=" << target_compression_ << ", "
<< "residual_gain=" << residual_gain;
if (residual_gain == 0)
return;
SetLevel(LevelFromGainError(residual_gain, level_));
}
void AgcManagerDirect::UpdateCompressor() {
if (compression_ == target_compression_) {
return;
}
// Adapt the compression gain slowly towards the target, in order to avoid
// highly perceptible changes.
if (target_compression_ > compression_) {
compression_accumulator_ += kCompressionGainStep;
} else {
compression_accumulator_ -= kCompressionGainStep;
}
// The compressor accepts integer gains in dB. Adjust the gain when
// we've come within half a stepsize of the nearest integer. (We don't
// check for equality due to potential floating point imprecision).
int new_compression = compression_;
int nearest_neighbor = std::floor(compression_accumulator_ + 0.5);
if (std::fabs(compression_accumulator_ - nearest_neighbor) <
kCompressionGainStep / 2) {
new_compression = nearest_neighbor;
}
// Set the new compression gain.
if (new_compression != compression_) {
compression_ = new_compression;
compression_accumulator_ = new_compression;
if (gctrl_->set_compression_gain_db(compression_) != 0) {
LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
}
}
}
} // namespace webrtc