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WebRtc_Word32 -> int32_t in audio_processing/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-10 07:50:54 +00:00
parent 557e92515d
commit b7192b8247
37 changed files with 1189 additions and 1191 deletions
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24
webrtc/modules/audio_processing/gain_control_impl.cc
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@@ -23,7 +23,7 @@ namespace webrtc {
typedef void Handle;
namespace {
WebRtc_Word16 MapSetting(GainControl::Mode mode) {
int16_t MapSetting(GainControl::Mode mode) {
switch (mode) {
case GainControl::kAdaptiveAnalog:
return kAgcModeAdaptiveAnalog;
@@ -59,7 +59,7 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
assert(audio->samples_per_split_channel() <= 160);
WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
int16_t* mixed_data = audio->low_pass_split_data(0);
if (audio->num_channels() > 1) {
audio->CopyAndMixLowPass(1);
mixed_data = audio->mixed_low_pass_data(0);
@@ -70,7 +70,7 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
int err = WebRtcAgc_AddFarend(
my_handle,
mixed_data,
static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
static_cast<int16_t>(audio->samples_per_split_channel()));
if (err != apm_->kNoError) {
return GetHandleError(my_handle);
@@ -97,7 +97,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
my_handle,
audio->low_pass_split_data(i),
audio->high_pass_split_data(i),
static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
static_cast<int16_t>(audio->samples_per_split_channel()));
if (err != apm_->kNoError) {
return GetHandleError(my_handle);
@@ -107,13 +107,13 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
for (int i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
WebRtc_Word32 capture_level_out = 0;
int32_t capture_level_out = 0;
err = WebRtcAgc_VirtualMic(
my_handle,
audio->low_pass_split_data(i),
audio->high_pass_split_data(i),
static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
static_cast<int16_t>(audio->samples_per_split_channel()),
//capture_levels_[i],
analog_capture_level_,
&capture_level_out);
@@ -145,14 +145,14 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
stream_is_saturated_ = false;
for (int i = 0; i < num_handles(); i++) {
Handle* my_handle = static_cast<Handle*>(handle(i));
WebRtc_Word32 capture_level_out = 0;
WebRtc_UWord8 saturation_warning = 0;
int32_t capture_level_out = 0;
uint8_t saturation_warning = 0;
int err = WebRtcAgc_Process(
my_handle,
audio->low_pass_split_data(i),
audio->high_pass_split_data(i),
static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
static_cast<int16_t>(audio->samples_per_split_channel()),
audio->low_pass_split_data(i),
audio->high_pass_split_data(i),
capture_levels_[i],
@@ -345,10 +345,10 @@ int GainControlImpl::ConfigureHandle(void* handle) const {
// TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
// change the interface.
//assert(target_level_dbfs_ <= 0);
//config.targetLevelDbfs = static_cast<WebRtc_Word16>(-target_level_dbfs_);
config.targetLevelDbfs = static_cast<WebRtc_Word16>(target_level_dbfs_);
//config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
config.compressionGaindB =
static_cast<WebRtc_Word16>(compression_gain_db_);
static_cast<int16_t>(compression_gain_db_);
config.limiterEnable = limiter_enabled_;
return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);