Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(480)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase + compile Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/media/webrtc/webrtcvoiceengine.h ('k') | talk/media/webrtc/webrtcvoiceengine_unittest.cc » ('j') | webrtc/base/bitbuffer_unittest.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 40d8442405ad83df68862de2fb03dec13a0d729e..fb9d18a75ed6568a6ce298c7f3631bb9da3211f1 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -843,7 +843,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
audioproc->SetExtraOptions(config);
}
- uint32 recording_sample_rate;
+ uint32_t recording_sample_rate;
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) {
@@ -851,7 +851,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
}
}
- uint32 playout_sample_rate;
+ uint32_t playout_sample_rate;
if (options.playout_sample_rate.Get(&playout_sample_rate)) {
LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate;
if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) {
@@ -1189,7 +1189,7 @@ void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
rtc::CritScope lock(&channels_cs_);
WebRtcVoiceMediaChannel* channel = NULL;
- uint32 ssrc = 0;
+ uint32_t ssrc = 0;
LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
<< channel_num << ".";
if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
@@ -1201,8 +1201,9 @@ void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
}
}
-bool WebRtcVoiceEngine::FindChannelAndSsrc(
- int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
+bool WebRtcVoiceEngine::FindChannelAndSsrc(int channel_num,
+ WebRtcVoiceMediaChannel** channel,
+ uint32_t* ssrc) const {
RTC_DCHECK(channel != NULL && ssrc != NULL);
*channel = NULL;
@@ -2085,7 +2086,8 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
return true;
}
-bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool enable,
+bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
+ bool enable,
const AudioOptions* options,
AudioRenderer* renderer) {
// TODO(solenberg): The state change should be fully rolled back if any one of
@@ -2207,7 +2209,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
return ChangeSend(channel, desired_send_);
}
-bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
+bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
ChannelMap::iterator it = send_channels_.find(ssrc);
if (it == send_channels_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
@@ -2247,7 +2249,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
if (!VERIFY(sp.ssrcs.size() == 1))
return false;
- uint32 ssrc = sp.first_ssrc();
+ uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
@@ -2371,7 +2373,7 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
return SetPlayout(channel, playout_);
}
-bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
+bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&receive_channels_cs_);
ChannelMap::iterator it = receive_channels_.find(ssrc);
@@ -2429,7 +2431,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
return true;
}
-bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
+bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32_t ssrc,
AudioRenderer* renderer) {
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
@@ -2451,7 +2453,7 @@ bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
return true;
}
-bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
+bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
AudioRenderer* renderer) {
ChannelMap::iterator it = send_channels_.find(ssrc);
if (it == send_channels_.end()) {
@@ -2522,8 +2524,9 @@ void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
}
}
-bool WebRtcVoiceMediaChannel::SetOutputScaling(
- uint32 ssrc, double left, double right) {
+bool WebRtcVoiceMediaChannel::SetOutputScaling(uint32_t ssrc,
+ double left,
+ double right) {
rtc::CritScope lock(&receive_channels_cs_);
// Collect the channels to scale the output volume.
std::vector<int> channels;
@@ -2574,8 +2577,10 @@ bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
return dtmf_allowed_;
}
-bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
- int duration, int flags) {
+bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
+ int event,
+ int duration,
+ int flags) {
if (!dtmf_allowed_) {
return false;
}
@@ -2699,7 +2704,7 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
}
}
-bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
+bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
@@ -2970,7 +2975,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
return true;
}
-bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
+bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32_t* ssrc) {
rtc::CritScope lock(&receive_channels_cs_);
RTC_DCHECK(ssrc != NULL);
if (channel_num == -1 && send_ != SEND_NOTHING) {
@@ -2985,7 +2990,7 @@ bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
for (const auto& ch : send_channels_) {
if (ch.second->channel() == channel_num) {
// This is a sending channel.
- uint32 local_ssrc = 0;
+ uint32_t local_ssrc = 0;
if (engine()->voe()->rtp()->GetLocalSSRC(
channel_num, local_ssrc) != -1) {
*ssrc = local_ssrc;
@@ -3005,7 +3010,7 @@ bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
return false;
}
-void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
+void WebRtcVoiceMediaChannel::OnError(uint32_t ssrc, int error) {
if (error == VE_TYPING_NOISE_WARNING) {
typing_noise_detected_ = true;
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
@@ -3020,14 +3025,14 @@ int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
return (ret == 0) ? static_cast<int>(ulevel) : -1;
}
-int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) const {
+int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32_t ssrc) const {
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
if (it != receive_channels_.end())
return it->second->channel();
return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
}
-int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) const {
+int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32_t ssrc) const {
ChannelMap::const_iterator it = send_channels_.find(ssrc);
if (it != send_channels_.end())
return it->second->channel();
@@ -3121,10 +3126,11 @@ bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
return true;
}
-uint32 WebRtcVoiceMediaChannel::ParseSsrc(const void* data, size_t len,
- bool rtcp) {
+uint32_t WebRtcVoiceMediaChannel::ParseSsrc(const void* data,
+ size_t len,
+ bool rtcp) {
size_t ssrc_pos = (!rtcp) ? 8 : 4;
- uint32 ssrc = 0;
+ uint32_t ssrc = 0;
if (len >= (ssrc_pos + sizeof(ssrc))) {
ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
}
@@ -3191,7 +3197,7 @@ void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() {
}
}
-void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32 ssrc) {
+void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
RTC_DCHECK(channel != nullptr);
@@ -3208,7 +3214,7 @@ void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32 ssrc) {
receive_streams_.insert(std::make_pair(ssrc, s));
}
-void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32 ssrc) {
+void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
auto stream_it = receive_streams_.find(ssrc);
if (stream_it != receive_streams_.end()) {
« no previous file with comments | « talk/media/webrtc/webrtcvoiceengine.h ('k') | talk/media/webrtc/webrtcvoiceengine_unittest.cc » ('j') | webrtc/base/bitbuffer_unittest.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698