Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1386653002
Cr-Commit-Position: refs/heads/master@{#10194}
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 2f06340..c53dac8 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -51,10 +51,6 @@
#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
-#define UNIMPLEMENTED \
- LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
- RTC_NOTREACHED()
-
namespace cricket {
namespace {
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 3c187b3..63130b0 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -908,10 +908,6 @@
}
protected:
- virtual std::vector<cricket::VideoCodec> GetCodecs() {
- return engine_.codecs();
- }
-
FakeVideoSendStream* AddSendStream() {
return AddSendStream(StreamParams::CreateLegacy(++last_ssrc_));
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 40d8442..05b98ec 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -54,8 +54,9 @@
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace cricket {
+namespace {
-static const int kMaxNumPacketSize = 6;
+const int kMaxNumPacketSize = 6;
struct CodecPref {
const char* name;
int clockrate;
@@ -65,7 +66,7 @@
int packet_sizes_ms[kMaxNumPacketSize];
};
// Note: keep the supported packet sizes in ascending order.
-static const CodecPref kCodecPrefs[] = {
+const CodecPref kCodecPrefs[] = {
{ kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
{ kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
{ kIsacCodecName, 32000, 1, 104, true, { 30 } },
@@ -97,14 +98,14 @@
// It's not clear yet whether the -2 index is handled properly on other OSes.
#ifdef WIN32
-static const int kDefaultAudioDeviceId = -1;
+const int kDefaultAudioDeviceId = -1;
#else
-static const int kDefaultAudioDeviceId = 0;
+const int kDefaultAudioDeviceId = 0;
#endif
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
-static const int kNackMaxPackets = 250;
+const int kNackMaxPackets = 250;
// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
@@ -117,18 +118,18 @@
// 64-128 kb/s for FB stereo music.
// The current implementation applies the following values to mono signals,
// and multiplies them by 2 for stereo.
-static const int kOpusBitrateNb = 12000;
-static const int kOpusBitrateWb = 20000;
-static const int kOpusBitrateFb = 32000;
+const int kOpusBitrateNb = 12000;
+const int kOpusBitrateWb = 20000;
+const int kOpusBitrateFb = 32000;
// Opus bitrate should be in the range between 6000 and 510000.
-static const int kOpusMinBitrate = 6000;
-static const int kOpusMaxBitrate = 510000;
+const int kOpusMinBitrate = 6000;
+const int kOpusMaxBitrate = 510000;
// 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 rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
+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
@@ -140,29 +141,29 @@
// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
// below.
#if defined(CHROMEOS)
-static const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
+const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
#elif defined(ANDROID)
-static const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
+const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
#else
-static const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
+const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
#endif
// Dumps an AudioCodec in RFC 2327-ish format.
-static std::string ToString(const AudioCodec& codec) {
+std::string ToString(const AudioCodec& codec) {
std::stringstream ss;
ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
<< " (" << codec.id << ")";
return ss.str();
}
-static std::string ToString(const webrtc::CodecInst& codec) {
+std::string ToString(const webrtc::CodecInst& codec) {
std::stringstream ss;
ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
<< " (" << codec.pltype << ")";
return ss.str();
}
-static void LogMultiline(rtc::LoggingSeverity sev, char* text) {
+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;
@@ -170,7 +171,7 @@
}
// Severity is an integer because it comes is assumed to be from command line.
-static int SeverityToFilter(int severity) {
+int SeverityToFilter(int severity) {
int filter = webrtc::kTraceNone;
switch (severity) {
case rtc::LS_VERBOSE:
@@ -188,15 +189,15 @@
return filter;
}
-static bool IsCodec(const AudioCodec& codec, const char* ref_name) {
+bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
-static bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
+bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
return (_stricmp(codec.plname, ref_name) == 0);
}
-static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
+bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
if (IsCodec(codec, kCodecPrefs[i].name) &&
kCodecPrefs[i].clockrate == codec.plfreq) {
@@ -206,7 +207,7 @@
return false;
}
-static bool FindCodec(const std::vector<AudioCodec>& codecs,
+bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec) {
for (const AudioCodec& c : codecs) {
@@ -220,12 +221,12 @@
return false;
}
-static bool IsNackEnabled(const AudioCodec& codec) {
+bool IsNackEnabled(const AudioCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
}
-static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
+int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
for (int packet_size_ms : codec_pref.packet_sizes_ms) {
if (packet_size_ms && packet_size_ms <= ptime_ms) {
@@ -238,7 +239,7 @@
// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec
// pacsize if it's valid, or we will pick the next smallest value we support.
// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
-static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
+bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
for (const CodecPref& codec_pref : kCodecPrefs) {
if ((IsCodec(*codec, codec_pref.name) &&
codec_pref.clockrate == codec->plfreq) ||
@@ -255,7 +256,7 @@
}
// Return true if codec.params[feature] == "1", false otherwise.
-static bool IsCodecFeatureEnabled(const AudioCodec& codec,
+bool IsCodecFeatureEnabled(const AudioCodec& codec,
const char* feature) {
int value;
return codec.GetParam(feature, &value) && value == 1;
@@ -265,7 +266,7 @@
// otherwise. If the value (either from params or codec.bitrate) <=0, use the
// default configuration. If the value is beyond feasible bit rate of Opus,
// clamp it. Returns the Opus bit rate for operation.
-static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
+int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
int bitrate = 0;
bool use_param = true;
if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
@@ -298,7 +299,7 @@
// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
-static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
+int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
int value;
if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
return value;
@@ -306,7 +307,7 @@
return kOpusDefaultMaxPlaybackRate;
}
-static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
+void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
bool* enable_codec_fec, int* max_playback_rate,
bool* enable_codec_dtx) {
*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
@@ -326,7 +327,7 @@
// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
// codec.
-static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
+void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
if (IsCodec(*voe_codec, kG722CodecName)) {
// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
// has changed, and this special case is no longer needed.
@@ -338,7 +339,7 @@
// Gets the default set of options applied to the engine. Historically, these
// were supplied as a combination of flags from the channel manager (ec, agc,
// ns, and highpass) and the rest hardcoded in InitInternal.
-static AudioOptions GetDefaultEngineOptions() {
+AudioOptions GetDefaultEngineOptions() {
AudioOptions options;
options.echo_cancellation.Set(true);
options.auto_gain_control.Set(true);
@@ -358,9 +359,10 @@
return options;
}
-static std::string GetEnableString(bool enable) {
+std::string GetEnableString(bool enable) {
return enable ? "enable" : "disable";
}
+} // namespace {
WebRtcVoiceEngine::WebRtcVoiceEngine()
: voe_wrapper_(new VoEWrapper()),
@@ -862,18 +864,6 @@
return true;
}
-struct ResumeEntry {
- ResumeEntry(WebRtcVoiceMediaChannel *c, bool p, SendFlags s)
- : channel(c),
- playout(p),
- send(s) {
- }
-
- WebRtcVoiceMediaChannel *channel;
- bool playout;
- SendFlags send;
-};
-
// TODO(juberti): Refactor this so that the core logic can be used to set the
// soundclip device. At that time, reinstate the soundclip pause/resume code.
bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
@@ -1186,40 +1176,18 @@
}
}
-void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
- rtc::CritScope lock(&channels_cs_);
- WebRtcVoiceMediaChannel* channel = NULL;
- uint32 ssrc = 0;
+void WebRtcVoiceEngine::CallbackOnError(int channel_id, int err_code) {
+ RTC_DCHECK(channel_id == -1);
LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
- << channel_num << ".";
- if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
- RTC_DCHECK(channel != NULL);
- channel->OnError(ssrc, err_code);
- } else {
- LOG(LS_ERROR) << "VoiceEngine channel " << channel_num
- << " could not be found in channel list when error reported.";
+ << channel_id << ".";
+ rtc::CritScope lock(&channels_cs_);
+ for (WebRtcVoiceMediaChannel* channel : channels_) {
+ channel->OnError(err_code);
}
}
-bool WebRtcVoiceEngine::FindChannelAndSsrc(
- int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
- RTC_DCHECK(channel != NULL && ssrc != NULL);
-
- *channel = NULL;
- *ssrc = 0;
- // Find corresponding channel and ssrc
- for (WebRtcVoiceMediaChannel* ch : channels_) {
- RTC_DCHECK(ch != NULL);
- if (ch->FindSsrc(channel_num, ssrc)) {
- *channel = ch;
- return true;
- }
- }
-
- return false;
-}
-
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
+ RTC_DCHECK(channel != NULL);
rtc::CritScope lock(&channels_cs_);
channels_.push_back(channel);
}
@@ -1416,6 +1384,7 @@
send_(SEND_NOTHING),
call_(call),
default_receive_ssrc_(0) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
<< voe_channel();
@@ -1425,16 +1394,19 @@
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
// Remove any remaining send streams, the default channel will be deleted
// later.
- while (!send_channels_.empty())
+ while (!send_channels_.empty()) {
RemoveSendStream(send_channels_.begin()->first);
+ }
// Unregister ourselves from the engine.
engine()->UnregisterChannel(this);
+
// Remove any remaining streams.
while (!receive_channels_.empty()) {
RemoveRecvStream(receive_channels_.begin()->first);
@@ -1447,6 +1419,7 @@
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
return (SetSendCodecs(params.codecs) &&
@@ -1457,6 +1430,7 @@
bool WebRtcVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
return (SetRecvCodecs(params.codecs) &&
@@ -1464,6 +1438,7 @@
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
@@ -1553,6 +1528,7 @@
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Set the payload types to be used for incoming media.
LOG(LS_INFO) << "Setting receive voice codecs:";
@@ -1804,6 +1780,8 @@
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
dtmf_allowed_ = false;
for (const AudioCodec& codec : codecs) {
// Find the DTMF telephone event "codec".
@@ -1875,6 +1853,7 @@
bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (receive_extensions_ == extensions) {
return true;
}
@@ -1942,6 +1921,7 @@
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (send_extensions_ == extensions) {
return true;
}
@@ -2000,6 +1980,7 @@
}
bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (playout_ == playout) {
return true;
}
@@ -2088,6 +2069,7 @@
bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool enable,
const AudioOptions* options,
AudioRenderer* renderer) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(solenberg): The state change should be fully rolled back if any one of
// these calls fail.
if (!SetLocalRenderer(ssrc, renderer)) {
@@ -2133,9 +2115,10 @@
}
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// If the default channel is already used for sending create a new channel
// otherwise use the default channel for sending.
- int channel = GetSendChannelNum(sp.first_ssrc());
+ int channel = GetSendChannelId(sp.first_ssrc());
if (channel != -1) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc();
return false;
@@ -2243,6 +2226,8 @@
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
+
rtc::CritScope lock(&receive_channels_cs_);
if (!VERIFY(sp.ssrcs.size() == 1))
@@ -2301,6 +2286,7 @@
}
bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Configure to use external transport, like our default channel.
if (engine()->voe()->network()->RegisterExternalTransport(
channel, *this) == -1) {
@@ -2373,6 +2359,8 @@
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
+
rtc::CritScope lock(&receive_channels_cs_);
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
@@ -2431,6 +2419,7 @@
bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
AudioRenderer* renderer) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
if (renderer) {
@@ -2475,6 +2464,7 @@
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// In conference mode, the default channel should not be in
// |receive_channels_|.
actives->clear();
@@ -2488,6 +2478,7 @@
}
int WebRtcVoiceMediaChannel::GetOutputLevel() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
// return the highest output level of all streams
int highest = GetOutputLevel(voe_channel());
for (const auto& ch : receive_channels_) {
@@ -2524,6 +2515,7 @@
bool WebRtcVoiceMediaChannel::SetOutputScaling(
uint32 ssrc, double left, double right) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&receive_channels_cs_);
// Collect the channels to scale the output volume.
std::vector<int> channels;
@@ -2536,7 +2528,7 @@
channels.push_back(ch.second->channel());
}
} else { // Collect only the channel of the specified ssrc.
- int channel = GetReceiveChannelNum(ssrc);
+ int channel = GetReceiveChannelId(ssrc);
if (-1 == channel) {
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
return false;
@@ -2597,7 +2589,7 @@
channel = send_channels_.begin()->second->channel();
}
} else {
- channel = GetSendChannelNum(ssrc);
+ channel = GetSendChannelId(ssrc);
}
if (channel == -1) {
LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
@@ -2639,7 +2631,7 @@
// any multiplexed streams, just send it to the default channel. Otherwise,
// send it to the specific decoder instance for that stream.
int which_channel =
- GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), false));
+ GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), false));
if (which_channel == -1) {
which_channel = voe_channel();
}
@@ -2675,7 +2667,7 @@
bool has_sent_to_default_channel = false;
if (type == kRtcpTypeSR) {
int which_channel =
- GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), true));
+ GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), true));
if (which_channel != -1) {
engine()->voe()->network()->ReceivedRTCPPacket(
which_channel, packet->data(), packet->size());
@@ -2700,7 +2692,7 @@
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
- int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
+ int channel = (ssrc == 0) ? voe_channel() : GetSendChannelId(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
@@ -2784,6 +2776,8 @@
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
bool echo_metrics_on = false;
// These can take on valid negative values, so use the lowest possible level
// as default rather than -1.
@@ -2970,42 +2964,10 @@
return true;
}
-bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
- rtc::CritScope lock(&receive_channels_cs_);
- RTC_DCHECK(ssrc != NULL);
- if (channel_num == -1 && send_ != SEND_NOTHING) {
- // Sometimes the VoiceEngine core will throw error with channel_num = -1.
- // This means the error is not limited to a specific channel. Signal the
- // message using ssrc=0. If the current channel is sending, use this
- // channel for sending the message.
- *ssrc = 0;
- return true;
- } else {
- // Check whether this is a sending channel.
- for (const auto& ch : send_channels_) {
- if (ch.second->channel() == channel_num) {
- // This is a sending channel.
- uint32 local_ssrc = 0;
- if (engine()->voe()->rtp()->GetLocalSSRC(
- channel_num, local_ssrc) != -1) {
- *ssrc = local_ssrc;
- }
- return true;
- }
- }
-
- // Check whether this is a receiving channel.
- for (const auto& ch : receive_channels_) {
- if (ch.second->channel() == channel_num) {
- *ssrc = ch.first;
- return true;
- }
- }
+void WebRtcVoiceMediaChannel::OnError(int error) {
+ if (send_ == SEND_NOTHING) {
+ return;
}
- return false;
-}
-
-void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
if (error == VE_TYPING_NOISE_WARNING) {
typing_noise_detected_ = true;
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
@@ -3014,20 +2976,21 @@
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
- unsigned int ulevel;
- int ret =
- engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
+ unsigned int ulevel = 0;
+ int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
return (ret == 0) ? static_cast<int>(ulevel) : -1;
}
-int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) const {
+int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32 ssrc) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
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::GetSendChannelId(uint32 ssrc) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
ChannelMap::const_iterator it = send_channels_.find(ssrc);
if (it != send_channels_.end())
return it->second->channel();
@@ -3219,6 +3182,7 @@
bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
const std::vector<AudioCodec>& new_codecs) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
for (const AudioCodec& codec : new_codecs) {
webrtc::CodecInst voe_codec;
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 69e88e4..61dc7b1 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -127,15 +127,12 @@
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
// webrtc::VoiceEngineObserver:
- void CallbackOnError(int channel, int errCode) override;
+ void CallbackOnError(int channel_id, int errCode) override;
// Given the device type, name, and id, find device id. Return true and
// set the output parameter rtc_id if successful.
bool FindWebRtcAudioDeviceId(
bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
- bool FindChannelAndSsrc(int channel_num,
- WebRtcVoiceMediaChannel** channel,
- uint32* ssrc) const;
void StartAecDump(const std::string& filename);
void StopAecDump();
@@ -237,11 +234,10 @@
return VoiceMediaChannel::SendRtcp(&packet);
}
- bool FindSsrc(int channel_num, uint32* ssrc);
- void OnError(uint32 ssrc, int error);
+ void OnError(int error);
- int GetReceiveChannelNum(uint32 ssrc) const;
- int GetSendChannelNum(uint32 ssrc) const;
+ int GetReceiveChannelId(uint32 ssrc) const;
+ int GetSendChannelId(uint32 ssrc) const;
private:
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 0a72783..8a03b24 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -2866,23 +2866,23 @@
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
-// Test that GetReceiveChannelNum returns the default channel for the first
+// Test that GetReceiveChannelId returns the default channel for the first
// recv stream in 1-1 calls.
-TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelNumIn1To1Calls) {
+TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelIdIn1To1Calls) {
EXPECT_TRUE(SetupEngine());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
// Test that GetChannelNum returns the default channel if the SSRC is unknown.
EXPECT_EQ(media_channel->voe_channel(),
- media_channel->GetReceiveChannelNum(0));
+ media_channel->GetReceiveChannelId(0));
cricket::StreamParams stream;
stream.ssrcs.push_back(kSsrc2);
EXPECT_TRUE(channel_->AddRecvStream(stream));
EXPECT_EQ(media_channel->voe_channel(),
- media_channel->GetReceiveChannelNum(kSsrc2));
+ media_channel->GetReceiveChannelId(kSsrc2));
}
-// Test that GetReceiveChannelNum doesn't return the default channel for the
+// Test that GetReceiveChannelId doesn't return the default channel for the
// first recv stream in conference calls.
TEST_F(WebRtcVoiceEngineTestFake, TestGetChannelNumInConferenceCalls) {
EXPECT_TRUE(SetupEngine());
@@ -2894,7 +2894,7 @@
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
EXPECT_LT(media_channel->voe_channel(),
- media_channel->GetReceiveChannelNum(kSsrc2));
+ media_channel->GetReceiveChannelId(kSsrc2));
}
TEST_F(WebRtcVoiceEngineTestFake, SetOutputScaling) {
