#include <string>
#include <vector>
-#include "talk/base/base64.h"
-#include "talk/base/byteorder.h"
-#include "talk/base/common.h"
-#include "talk/base/helpers.h"
-#include "talk/base/logging.h"
-#include "talk/base/stringencode.h"
-#include "talk/base/stringutils.h"
#include "talk/media/base/audiorenderer.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/voiceprocessor.h"
#include "talk/media/webrtc/webrtcvoe.h"
+#include "webrtc/base/base64.h"
+#include "webrtc/base/byteorder.h"
+#include "webrtc/base/common.h"
+#include "webrtc/base/helpers.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/stringencode.h"
+#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
-static const talk_base::DiffServCodePoint kAudioDscpValue = talk_base::DSCP_EF;
+static const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
// Ensure we open the file in a writeable path on ChromeOS and Android. This
// workaround can be removed when it's possible to specify a filename for audio
return ss.str();
}
-static void LogMultiline(talk_base::LoggingSeverity sev, char* text) {
+static void LogMultiline(rtc::LoggingSeverity sev, char* text) {
const char* delim = "\r\n";
for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
LOG_V(sev) << tok;
static int SeverityToFilter(int severity) {
int filter = webrtc::kTraceNone;
switch (severity) {
- case talk_base::LS_VERBOSE:
+ case rtc::LS_VERBOSE:
filter |= webrtc::kTraceAll;
- case talk_base::LS_INFO:
+ case rtc::LS_INFO:
filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
- case talk_base::LS_WARNING:
+ case rtc::LS_WARNING:
filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
- case talk_base::LS_ERROR:
+ case rtc::LS_ERROR:
filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
}
return filter;
options.experimental_aec.Set(false);
options.experimental_ns.Set(false);
options.aec_dump.Set(false);
+ options.opus_fec.Set(false);
return options;
}
private:
WebRtcVoiceEngine *engine_;
int webrtc_channel_;
- talk_base::scoped_ptr<WebRtcSoundclipStream> stream_;
+ rtc::scoped_ptr<WebRtcSoundclipStream> stream_;
};
WebRtcVoiceEngine::WebRtcVoiceEngine()
// True if params["stereo"] == "1"
static bool IsOpusStereoEnabled(const AudioCodec& codec) {
- CodecParameterMap::const_iterator param =
- codec.params.find(kCodecParamStereo);
- if (param == codec.params.end()) {
- return false;
- }
- return param->second == kParamValueTrue;
+ int value;
+ return codec.GetParam(kCodecParamStereo, &value) && value == 1;
}
static bool IsValidOpusBitrate(int bitrate) {
return bitrate;
}
+// Return true if params[kCodecParamUseInbandFec] == "1", false
+// otherwise.
+static bool IsOpusFecEnabled(const AudioCodec& codec) {
+ int value;
+ return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
+}
+
+// Set params[kCodecParamUseInbandFec]. Caller should make sure codec is Opus.
+static void SetOpusFec(AudioCodec* codec, bool opus_fec) {
+ if (opus_fec) {
+ codec->SetParam(kCodecParamUseInbandFec, 1);
+ } else {
+ codec->RemoveParam(kCodecParamUseInbandFec);
+ }
+}
+
void WebRtcVoiceEngine::ConstructCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
// Only add fmtp parameters that differ from the spec.
if (kPreferredMinPTime != kOpusDefaultMinPTime) {
codec.params[kCodecParamMinPTime] =
- talk_base::ToString(kPreferredMinPTime);
+ rtc::ToString(kPreferredMinPTime);
}
if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
codec.params[kCodecParamMaxPTime] =
- talk_base::ToString(kPreferredMaxPTime);
+ rtc::ToString(kPreferredMaxPTime);
}
// TODO(hellner): Add ptime, sprop-stereo, stereo and useinbandfec
// when they can be set to values other than the default.
+ SetOpusFec(&codec, false);
}
codecs_.push_back(codec);
} else {
tracing_->SetTraceCallback(NULL);
}
-bool WebRtcVoiceEngine::Init(talk_base::Thread* worker_thread) {
+bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
bool res = InitInternal();
if (res) {
bool WebRtcVoiceEngine::InitInternal() {
// Temporarily turn logging level up for the Init call
int old_filter = log_filter_;
- int extended_filter = log_filter_ | SeverityToFilter(talk_base::LS_INFO);
+ int extended_filter = log_filter_ | SeverityToFilter(rtc::LS_INFO);
SetTraceFilter(extended_filter);
SetTraceOptions("");
char buffer[1024] = "";
voe_wrapper_->base()->GetVersion(buffer);
LOG(LS_INFO) << "WebRtc VoiceEngine Version:";
- LogMultiline(talk_base::LS_INFO, buffer);
+ LogMultiline(rtc::LS_INFO, buffer);
// Save the default AGC configuration settings. This must happen before
// calling SetOptions or the default will be overwritten.
if (options.experimental_ns.Get(&experimental_ns)) {
webrtc::AudioProcessing* audioproc =
voe_wrapper_->base()->audio_processing();
+#ifdef USE_WEBRTC_DEV_BRANCH
+ webrtc::Config config;
+ config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(
+ experimental_ns));
+ audioproc->SetExtraOptions(config);
+#else
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
if (audioproc) {
LOG(LS_VERBOSE) << "Experimental noise suppression set to "
<< experimental_ns;
}
+#endif
}
bool highpass_filter;
}
}
+ bool opus_fec;
+ if (options.opus_fec.Get(&opus_fec)) {
+ LOG(LS_INFO) << "Opus FEC is enabled? " << opus_fec;
+ for (std::vector<AudioCodec>::iterator it = codecs_.begin();
+ it != codecs_.end(); ++it) {
+ if (IsOpus(*it))
+ SetOpusFec(&(*it), opus_fec);
+ }
+ }
+
return true;
}
bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
const Device* out_device) {
#if !defined(IOS)
- int in_id = in_device ? talk_base::FromString<int>(in_device->id) :
+ int in_id = in_device ? rtc::FromString<int>(in_device->id) :
kDefaultAudioDeviceId;
- int out_id = out_device ? talk_base::FromString<int>(out_device->id) :
+ int out_id = out_device ? rtc::FromString<int>(out_device->id) :
kDefaultAudioDeviceId;
// The device manager uses -1 as the default device, which was the case for
// VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
ret = false;
}
+ webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
+ if (ap)
+ ap->Initialize();
}
// Find the playout device id in VoiceEngine and set playout device.
void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) {
// Set encrypted trace file.
std::vector<std::string> opts;
- talk_base::tokenize(options, ' ', '"', '"', &opts);
+ rtc::tokenize(options, ' ', '"', '"', &opts);
std::vector<std::string>::iterator tracefile =
std::find(opts.begin(), opts.end(), "tracefile");
if (tracefile != opts.end() && ++tracefile != opts.end()) {
std::vector<std::string>::iterator tracefilter =
std::find(opts.begin(), opts.end(), "tracefilter");
if (tracefilter != opts.end() && ++tracefilter != opts.end()) {
- if (!tracing_->SetTraceFilter(talk_base::FromString<int>(*tracefilter))) {
+ if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) {
LOG_RTCERR1(SetTraceFilter, *tracefilter);
}
}
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
- talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;
+ rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
- sev = talk_base::LS_ERROR;
+ sev = rtc::LS_ERROR;
else if (level == webrtc::kTraceWarning)
- sev = talk_base::LS_WARNING;
+ sev = rtc::LS_WARNING;
else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
- sev = talk_base::LS_INFO;
+ sev = rtc::LS_INFO;
else if (level == webrtc::kTraceTerseInfo)
- sev = talk_base::LS_INFO;
+ sev = rtc::LS_INFO;
// Skip past boilerplate prefix text
if (length < 72) {
}
void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
- talk_base::CritScope lock(&channels_cs_);
+ rtc::CritScope lock(&channels_cs_);
WebRtcVoiceMediaChannel* channel = NULL;
uint32 ssrc = 0;
LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
}
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel *channel) {
- talk_base::CritScope lock(&channels_cs_);
+ rtc::CritScope lock(&channels_cs_);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
- talk_base::CritScope lock(&channels_cs_);
+ rtc::CritScope lock(&channels_cs_);
ChannelList::iterator i = std::find(channels_.begin(),
channels_.end(),
channel);
return true;
}
-bool WebRtcVoiceEngine::StartAecDump(talk_base::PlatformFile file) {
- FILE* aec_dump_file_stream = talk_base::FdopenPlatformFileForWriting(file);
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
+ FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
if (!aec_dump_file_stream) {
LOG(LS_ERROR) << "Could not open AEC dump file stream.";
- if (!talk_base::ClosePlatformFile(file))
+ if (!rtc::ClosePlatformFile(file))
LOG(LS_WARNING) << "Could not close file.";
return false;
}
webrtc::ProcessingTypes processing_type;
{
- talk_base::CritScope cs(&signal_media_critical_);
+ rtc::CritScope cs(&signal_media_critical_);
if (direction == MPD_RX) {
processing_type = webrtc::kPlaybackAllChannelsMixed;
if (SignalRxMediaFrame.is_empty()) {
int deregister_id = -1;
{
- talk_base::CritScope cs(&signal_media_critical_);
+ rtc::CritScope cs(&signal_media_critical_);
if ((processor_direction & channel_direction) != 0 && !signal->is_empty()) {
signal->disconnect(voice_processor);
int channel_id = -1;
int length,
int sampling_freq,
bool is_stereo) {
- talk_base::CritScope cs(&signal_media_critical_);
+ rtc::CritScope cs(&signal_media_critical_);
AudioFrame frame(audio10ms, length, sampling_freq, is_stereo);
if (type == webrtc::kPlaybackAllChannelsMixed) {
SignalRxMediaFrame(rx_processor_ssrc_, MPD_RX, &frame);
// This method is called on the libjingle worker thread.
// TODO(xians): Make sure Start() is called only once.
void Start(AudioRenderer* renderer) {
- talk_base::CritScope lock(&lock_);
+ rtc::CritScope lock(&lock_);
ASSERT(renderer != NULL);
if (renderer_ != NULL) {
ASSERT(renderer_ == renderer);
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void Stop() {
- talk_base::CritScope lock(&lock_);
+ rtc::CritScope lock(&lock_);
if (renderer_ == NULL)
return;
// Callback from the |renderer_| when it is going away. In case Start() has
// never been called, this callback won't be triggered.
virtual void OnClose() OVERRIDE {
- talk_base::CritScope lock(&lock_);
+ rtc::CritScope lock(&lock_);
// Set |renderer_| to NULL to make sure no more callback will get into
// the renderer.
renderer_ = NULL;
AudioRenderer* renderer_;
// Protects |renderer_| in Start(), Stop() and OnClose().
- talk_base::CriticalSection lock_;
+ rtc::CriticalSection lock_;
};
// WebRtcVoiceMediaChannel
}
}
if (dscp_option_changed) {
- talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
+ rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
dscp = kAudioDscpValue;
if (MediaChannel::SetDscp(dscp) != 0) {
bool WebRtcVoiceMediaChannel::SetSendCodecs(
int channel, const std::vector<AudioCodec>& codecs) {
- // Disable VAD, and FEC unless we know the other side wants them.
+ // Disable VAD, FEC, and RED unless we know the other side wants them.
engine()->voe()->codec()->SetVADStatus(channel, false);
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
+#ifdef USE_WEBRTC_DEV_BRANCH
+ engine()->voe()->rtp()->SetREDStatus(channel, false);
+ engine()->voe()->codec()->SetFECStatus(channel, false);
+#else
+ // TODO(minyue): Remove code under #else case after new WebRTC roll.
engine()->voe()->rtp()->SetFECStatus(channel, false);
+#endif // USE_WEBRTC_DEV_BRANCH
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD and DTMF.
memset(&send_codec, 0, sizeof(send_codec));
bool nack_enabled = nack_enabled_;
+ bool enable_codec_fec = false;
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
// We'll use the first codec in the list to actually send audio data.
// Be sure to use the payload type requested by the remote side.
- // "red", for FEC audio, is a special case where the actual codec to be
+ // "red", for RED audio, is a special case where the actual codec to be
// used is specified in params.
if (IsRedCodec(it->name)) {
// Parse out the RED parameters. If we fail, just ignore RED;
// Enable redundant encoding of the specified codec. Treat any
// failure as a fatal internal error.
+#ifdef USE_WEBRTC_DEV_BRANCH
+ LOG(LS_INFO) << "Enabling RED on channel " << channel;
+ if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) {
+ LOG_RTCERR3(SetREDStatus, channel, true, it->id);
+#else
+ // TODO(minyue): Remove code under #else case after new WebRTC roll.
LOG(LS_INFO) << "Enabling FEC";
if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) {
LOG_RTCERR3(SetFECStatus, channel, true, it->id);
+#endif // USE_WEBRTC_DEV_BRANCH
return false;
}
} else {
send_codec = voe_codec;
nack_enabled = IsNackEnabled(*it);
+ // For Opus as the send codec, we enable inband FEC if requested.
+ enable_codec_fec = IsOpus(*it) && IsOpusFecEnabled(*it);
}
found_send_codec = true;
break;
if (!SetSendCodec(channel, send_codec))
return false;
+ // FEC should be enabled after SetSendCodec.
+ if (enable_codec_fec) {
+ LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
+ << channel;
+#ifdef USE_WEBRTC_DEV_BRANCH
+ if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
+ // Enable codec internal FEC. Treat any failure as fatal internal error.
+ LOG_RTCERR2(SetFECStatus, channel, true);
+ return false;
+ }
+#endif // USE_WEBRTC_DEV_BRANCH
+ }
+
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
-#ifdef USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (!SetHeaderExtension(
audio_level_extension)) {
return false;
}
-#endif // USE_WEBRTC_DEV_BRANCH
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
- talk_base::CritScope lock(&receive_channels_cs_);
+ rtc::CritScope lock(&receive_channels_cs_);
if (!VERIFY(sp.ssrcs.size() == 1))
return false;
}
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
- talk_base::CritScope lock(&receive_channels_cs_);
+ rtc::CritScope lock(&receive_channels_cs_);
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
for (ChannelMap::iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
int level = GetOutputLevel(it->second->channel());
- highest = talk_base::_max(level, highest);
+ highest = rtc::_max(level, highest);
}
return highest;
}
bool WebRtcVoiceMediaChannel::SetOutputScaling(
uint32 ssrc, double left, double right) {
- talk_base::CritScope lock(&receive_channels_cs_);
+ rtc::CritScope lock(&receive_channels_cs_);
// Collect the channels to scale the output volume.
std::vector<int> channels;
if (0 == ssrc) { // Collect all channels, including the default one.
// Scale the output volume for the collected channels. We first normalize to
// scale the volume and then set the left and right pan.
- float scale = static_cast<float>(talk_base::_max(left, right));
+ float scale = static_cast<float>(rtc::_max(left, right));
if (scale > 0.0001f) {
left /= scale;
right /= scale;
uint32 ssrc, double* left, double* right) {
if (!left || !right) return false;
- talk_base::CritScope lock(&receive_channels_cs_);
+ rtc::CritScope lock(&receive_channels_cs_);
// Determine which channel based on ssrc.
int channel = (0 == ssrc) ? voe_channel() : GetReceiveChannelNum(ssrc);
if (channel == -1) {
}
void WebRtcVoiceMediaChannel::OnPacketReceived(
- talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
+ rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
// Pick which channel to send this packet to. If this packet doesn't match
// any multiplexed streams, just send it to the default channel. Otherwise,
// send it to the specific decoder instance for that stream.
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
- talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
+ rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
// Sending channels need all RTCP packets with feedback information.
// Even sender reports can contain attached report blocks.
// Receiving channels need sender reports in order to create
LOG_RTCERR2(SetInputMute, channel, muted);
return false;
}
+ // We set the AGC to mute state only when all the channels are muted.
+ // This implementation is not ideal, instead we should signal the AGC when
+ // the mic channel is muted/unmuted. We can't do it today because there
+ // is no good way to know which stream is mapping to the mic channel.
+ bool all_muted = muted;
+ for (ChannelMap::const_iterator iter = send_channels_.begin();
+ iter != send_channels_.end() && all_muted; ++iter) {
+ if (engine()->voe()->volume()->GetInputMute(iter->second->channel(),
+ all_muted)) {
+ LOG_RTCERR1(GetInputMute, iter->second->channel());
+ return false;
+ }
+ }
+
+ webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
+ if (ap)
+ ap->set_output_will_be_muted(all_muted);
return true;
}
rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
rinfo.packets_lost = cs.cumulativeLost;
rinfo.ext_seqnum = cs.extendedMax;
+#ifdef USE_WEBRTC_DEV_BRANCH
+ rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
+#endif
+ if (codec.pltype != -1) {
+ rinfo.codec_name = codec.plname;
+ }
// Convert samples to milliseconds.
if (codec.plfreq / 1000 > 0) {
rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
}
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
- talk_base::CritScope lock(&receive_channels_cs_);
+ rtc::CritScope lock(&receive_channels_cs_);
ASSERT(ssrc != NULL);
if (channel_num == -1 && send_ != SEND_NOTHING) {
// Sometimes the VoiceEngine core will throw error with channel_num = -1.
if (it != red_codec.params.end()) {
red_params = it->second;
std::vector<std::string> red_pts;
- if (talk_base::split(red_params, '/', &red_pts) != 2 ||
+ if (rtc::split(red_params, '/', &red_pts) != 2 ||
red_pts[0] != red_pts[1] ||
- !talk_base::FromString(red_pts[0], &red_pt)) {
+ !rtc::FromString(red_pts[0], &red_pt)) {
LOG(LS_WARNING) << "RED params " << red_params << " not supported.";
return false;
}
size_t ssrc_pos = (!rtcp) ? 8 : 4;
uint32 ssrc = 0;
if (len >= (ssrc_pos + sizeof(ssrc))) {
- ssrc = talk_base::GetBE32(static_cast<const char*>(data) + ssrc_pos);
+ ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
}
return ssrc;
}
projects / platform / framework / web / crosswalk.git / blobdiff