Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 Specifically, usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0. BUG=1014 R=juberti@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48499004 Cr-Commit-Position: refs/heads/master@{#8872}
This commit is contained in:
Minyue Li
parent
bef8d2d020
commit
7100dcd317
@ -69,6 +69,7 @@ using cricket::kCodecParamSPropStereo;
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using cricket::kCodecParamStartBitrate;
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using cricket::kCodecParamStereo;
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using cricket::kCodecParamUseInbandFec;
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using cricket::kCodecParamUseDtx;
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using cricket::kCodecParamSctpProtocol;
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using cricket::kCodecParamSctpStreams;
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using cricket::kCodecParamMaxAverageBitrate;
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@ -1560,9 +1561,9 @@ void WriteFmtpParameters(const cricket::CodecParameterMap& parameters,
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bool IsFmtpParam(const std::string& name) {
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const char* kFmtpParams[] = {
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kCodecParamMinPTime, kCodecParamSPropStereo,
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kCodecParamStereo, kCodecParamUseInbandFec, kCodecParamStartBitrate,
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kCodecParamMaxBitrate, kCodecParamMinBitrate, kCodecParamMaxQuantization,
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kCodecParamSctpProtocol, kCodecParamSctpStreams,
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kCodecParamStereo, kCodecParamUseInbandFec, kCodecParamUseDtx,
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kCodecParamStartBitrate, kCodecParamMaxBitrate, kCodecParamMinBitrate,
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kCodecParamMaxQuantization, kCodecParamSctpProtocol, kCodecParamSctpStreams,
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kCodecParamMaxAverageBitrate, kCodecParamMaxPlaybackRate,
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kCodecParamAssociatedPayloadType
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};
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@ -62,6 +62,7 @@ const char kCodecParamMinPTime[] = "minptime";
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const char kCodecParamSPropStereo[] = "sprop-stereo";
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const char kCodecParamStereo[] = "stereo";
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const char kCodecParamUseInbandFec[] = "useinbandfec";
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const char kCodecParamUseDtx[] = "usedtx";
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const char kCodecParamMaxAverageBitrate[] = "maxaveragebitrate";
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const char kCodecParamMaxPlaybackRate[] = "maxplaybackrate";
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@ -77,6 +78,7 @@ const int kOpusDefaultMinPTime = 3;
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const int kOpusDefaultSPropStereo = 0;
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const int kOpusDefaultStereo = 0;
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const int kOpusDefaultUseInbandFec = 0;
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const int kOpusDefaultUseDtx = 0;
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const int kOpusDefaultMaxPlaybackRate = 48000;
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const int kPreferredMaxPTime = 60;
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@ -68,6 +68,7 @@ extern const char kCodecParamMinPTime[];
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extern const char kCodecParamSPropStereo[];
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extern const char kCodecParamStereo[];
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extern const char kCodecParamUseInbandFec[];
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extern const char kCodecParamUseDtx[];
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extern const char kCodecParamMaxAverageBitrate[];
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extern const char kCodecParamMaxPlaybackRate[];
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extern const char kCodecParamSctpProtocol[];
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@ -87,6 +88,7 @@ extern const int kOpusDefaultMinPTime;
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extern const int kOpusDefaultSPropStereo;
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extern const int kOpusDefaultStereo;
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extern const int kOpusDefaultUseInbandFec;
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extern const int kOpusDefaultUseDtx;
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extern const int kOpusDefaultMaxPlaybackRate;
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// Prefered values in this code base. Note that they may differ from the default
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@ -314,6 +314,9 @@ class FakeWebRtcVoiceEngine
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bool GetVAD(int channel) {
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return channels_[channel]->vad;
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}
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bool GetOpusDtx(int channel) {
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return channels_[channel]->opus_dtx;
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}
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bool GetRED(int channel) {
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return channels_[channel]->red;
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}
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@ -79,7 +79,6 @@ static const CodecPref kCodecPrefs[] = {
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{ kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
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{ kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
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{ kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
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{ kCnCodecName, 48000, 1, 107, false, { } },
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{ kCnCodecName, 32000, 1, 106, false, { } },
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{ kCnCodecName, 16000, 1, 105, false, { } },
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{ kCnCodecName, 8000, 1, 13, false, { } },
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@ -161,6 +160,7 @@ static std::string ToString(const AudioCodec& codec) {
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<< " (" << codec.id << ")";
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return ss.str();
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}
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static std::string ToString(const webrtc::CodecInst& codec) {
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std::stringstream ss;
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ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
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@ -191,9 +191,17 @@ static int SeverityToFilter(int severity) {
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return filter;
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}
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static bool IsCodec(const AudioCodec& codec, const char* ref_name) {
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return (_stricmp(codec.name.c_str(), ref_name) == 0);
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}
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static bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
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return (_stricmp(codec.plname, ref_name) == 0);
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}
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static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
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if (_stricmp(kCodecPrefs[i].name, codec.plname) == 0 &&
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if (IsCodec(codec, kCodecPrefs[i].name) &&
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kCodecPrefs[i].clockrate == codec.plfreq) {
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return kCodecPrefs[i].is_multi_rate;
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}
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@ -201,18 +209,6 @@ static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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return false;
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}
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static bool IsTelephoneEventCodec(const std::string& name) {
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return _stricmp(name.c_str(), "telephone-event") == 0;
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}
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static bool IsCNCodec(const std::string& name) {
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return _stricmp(name.c_str(), "CN") == 0;
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}
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static bool IsRedCodec(const std::string& name) {
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return _stricmp(name.c_str(), "red") == 0;
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}
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static bool FindCodec(const std::vector<AudioCodec>& codecs,
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const AudioCodec& codec,
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AudioCodec* found_codec) {
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@ -248,9 +244,9 @@ static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
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static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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for (const CodecPref& codec_pref : kCodecPrefs) {
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if ((_stricmp(codec_pref.name, codec->plname) == 0 &&
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if ((IsCodec(*codec, codec_pref.name) &&
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codec_pref.clockrate == codec->plfreq) ||
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_stricmp(codec_pref.name, kG722CodecName) == 0) {
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IsCodec(*codec, kG722CodecName)) {
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int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
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if (packet_size_ms) {
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// Convert unit from milli-seconds to samples.
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@ -262,6 +258,87 @@ static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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return false;
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}
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// Return true if codec.params[feature] == "1", false otherwise.
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static bool IsCodecFeatureEnabled(const AudioCodec& codec,
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const char* feature) {
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int value;
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return codec.GetParam(feature, &value) && value == 1;
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}
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// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
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// otherwise. If the value (either from params or codec.bitrate) <=0, use the
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// default configuration. If the value is beyond feasible bit rate of Opus,
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// clamp it. Returns the Opus bit rate for operation.
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static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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int bitrate = 0;
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bool use_param = true;
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if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
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bitrate = codec.bitrate;
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use_param = false;
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}
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if (bitrate <= 0) {
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if (max_playback_rate <= 8000) {
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bitrate = kOpusBitrateNb;
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} else if (max_playback_rate <= 16000) {
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bitrate = kOpusBitrateWb;
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} else {
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bitrate = kOpusBitrateFb;
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}
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if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
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bitrate *= 2;
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}
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} else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
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bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
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std::string rate_source =
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use_param ? "Codec parameter \"maxaveragebitrate\"" :
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"Supplied Opus bitrate";
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LOG(LS_WARNING) << rate_source
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<< " is invalid and is replaced by: "
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<< bitrate;
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}
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return bitrate;
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}
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// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
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// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
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static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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int value;
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if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
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return value;
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}
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return kOpusDefaultMaxPlaybackRate;
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}
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static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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bool* enable_codec_fec, int* max_playback_rate,
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bool* enable_codec_dtx) {
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*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
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*enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
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*max_playback_rate = GetOpusMaxPlaybackRate(codec);
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// If OPUS, change what we send according to the "stereo" codec
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// parameter, and not the "channels" parameter. We set
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// voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
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// the bitrate is not specified, i.e. is <= zero, we set it to the
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// appropriate default value for mono or stereo Opus.
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voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
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voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
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}
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// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
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// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
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// codec.
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static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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if (IsCodec(*voe_codec, kG722CodecName)) {
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// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
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// has changed, and this special case is no longer needed.
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ASSERT(voe_codec->plfreq != new_plfreq);
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voe_codec->plfreq = new_plfreq;
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}
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}
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// Gets the default set of options applied to the engine. Historically, these
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// were supplied as a combination of flags from the channel manager (ec, agc,
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// ns, and highpass) and the rest hardcoded in InitInternal.
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@ -283,100 +360,8 @@ static AudioOptions GetDefaultEngineOptions() {
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return options;
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}
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static bool IsOpus(const AudioCodec& codec) {
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return (_stricmp(codec.name.c_str(), kOpusCodecName) == 0);
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}
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static bool IsIsac(const AudioCodec& codec) {
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return (_stricmp(codec.name.c_str(), kIsacCodecName) == 0);
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}
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// True if params["stereo"] == "1"
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static bool IsOpusStereoEnabled(const AudioCodec& codec) {
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int value;
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return codec.GetParam(kCodecParamStereo, &value) && value == 1;
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}
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// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
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// otherwise. If the value (either from params or codec.bitrate) <=0, use the
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// default configuration. If the value is beyond feasible bit rate of Opus,
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// clamp it. Returns the Opus bit rate for operation.
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static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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int bitrate = 0;
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bool use_param = true;
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if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
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bitrate = codec.bitrate;
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use_param = false;
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}
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if (bitrate <= 0) {
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if (max_playback_rate <= 8000) {
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bitrate = kOpusBitrateNb;
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}
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else if (max_playback_rate <= 16000) {
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bitrate = kOpusBitrateWb;
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}
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else {
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bitrate = kOpusBitrateFb;
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}
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if (IsOpusStereoEnabled(codec)) {
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bitrate *= 2;
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}
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}
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else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
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bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
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std::string rate_source =
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use_param ? "Codec parameter \"maxaveragebitrate\"" :
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"Supplied Opus bitrate";
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LOG(LS_WARNING) << rate_source
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<< " is invalid and is replaced by: "
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<< bitrate;
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}
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return bitrate;
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}
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// Return true if params[kCodecParamUseInbandFec] == "1", false
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// otherwise.
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static bool IsOpusFecEnabled(const AudioCodec& codec) {
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int value;
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return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
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}
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// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
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// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
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static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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int value;
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if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
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return value;
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}
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return kOpusDefaultMaxPlaybackRate;
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}
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static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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bool* enable_codec_fec, int* max_playback_rate) {
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*enable_codec_fec = IsOpusFecEnabled(codec);
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*max_playback_rate = GetOpusMaxPlaybackRate(codec);
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// If OPUS, change what we send according to the "stereo" codec
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// parameter, and not the "channels" parameter. We set
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// voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
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// the bitrate is not specified, i.e. is <= zero, we set it to the
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// appropriate default value for mono or stereo Opus.
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voe_codec->channels = IsOpusStereoEnabled(codec) ? 2 : 1;
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voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
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}
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// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
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// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
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// codec.
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static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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if (_stricmp(voe_codec->plname, kG722CodecName) == 0) {
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// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
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// has changed, and this special case is no longer needed.
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ASSERT(voe_codec->plfreq != new_plfreq);
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voe_codec->plfreq = new_plfreq;
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}
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static std::string GetEnableString(bool enable) {
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return enable ? "enable" : "disable";
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}
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class WebRtcSoundclipMedia : public SoundclipMedia {
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@ -535,13 +520,13 @@ void WebRtcVoiceEngine::ConstructCodecs() {
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webrtc::CodecInst voe_codec;
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if (GetVoeCodec(i, &voe_codec)) {
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// Skip uncompressed formats.
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if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
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if (IsCodec(voe_codec, kL16CodecName)) {
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continue;
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}
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const CodecPref* pref = NULL;
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for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
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if (_stricmp(kCodecPrefs[j].name, voe_codec.plname) == 0 &&
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if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
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kCodecPrefs[j].clockrate == voe_codec.plfreq &&
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kCodecPrefs[j].channels == voe_codec.channels) {
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pref = &kCodecPrefs[j];
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@ -556,11 +541,11 @@ void WebRtcVoiceEngine::ConstructCodecs() {
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voe_codec.rate, voe_codec.channels,
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ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
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LOG(LS_INFO) << ToString(codec);
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if (IsIsac(codec)) {
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if (IsCodec(codec, kIsacCodecName)) {
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// Indicate auto-bitrate in signaling.
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codec.bitrate = 0;
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}
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if (IsOpus(codec)) {
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if (IsCodec(codec, kOpusCodecName)) {
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// Only add fmtp parameters that differ from the spec.
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if (kPreferredMinPTime != kOpusDefaultMinPTime) {
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codec.params[kCodecParamMinPTime] =
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@ -1323,7 +1308,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
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MaybeFixupG722(&voe_codec, 16000);
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// Apply codec-specific settings.
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if (IsIsac(codec)) {
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if (IsCodec(codec, kIsacCodecName)) {
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// If ISAC and an explicit bitrate is not specified,
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// enable auto bitrate adjustment.
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voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
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@ -2120,7 +2105,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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bool nack_enabled = nack_enabled_;
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bool enable_codec_fec = false;
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bool enable_opus_dtx = false;
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int opus_max_playback_rate = 0;
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// Set send codec (the first non-telephone-event/CN codec)
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@ -2134,7 +2119,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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continue;
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}
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if (IsTelephoneEventCodec(it->name) || IsCNCodec(it->name)) {
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if (IsCodec(*it, kDtmfCodecName) || IsCodec(*it, kCnCodecName)) {
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// Skip telephone-event/CN codec, which will be handled later.
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continue;
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}
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@ -2143,7 +2128,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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// Be sure to use the payload type requested by the remote side.
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// "red", for RED audio, is a special case where the actual codec to be
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// used is specified in params.
|
||||
if (IsRedCodec(it->name)) {
|
||||
if (IsCodec(*it, kRedCodecName)) {
|
||||
// Parse out the RED parameters. If we fail, just ignore RED;
|
||||
// we don't support all possible params/usage scenarios.
|
||||
if (!GetRedSendCodec(*it, codecs, &send_codec)) {
|
||||
@ -2160,11 +2145,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
} else {
|
||||
send_codec = voe_codec;
|
||||
nack_enabled = IsNackEnabled(*it);
|
||||
// For Opus as the send codec, we are to enable inband FEC if requested
|
||||
// and set maximum playback rate.
|
||||
if (IsOpus(*it)) {
|
||||
// For Opus as the send codec, we are to determine inband FEC, maximum
|
||||
// playback rate, and opus internal dtx.
|
||||
if (IsCodec(*it, kOpusCodecName)) {
|
||||
GetOpusConfig(*it, &send_codec, &enable_codec_fec,
|
||||
&opus_max_playback_rate);
|
||||
&opus_max_playback_rate, &enable_opus_dtx);
|
||||
}
|
||||
|
||||
// Set packet size if the AudioCodec param kCodecParamPTime is set.
|
||||
@ -2207,17 +2192,32 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
}
|
||||
}
|
||||
|
||||
// maxplaybackrate should be set after SetSendCodec.
|
||||
// If opus_max_playback_rate <= 0, the default maximum playback rate of 48 kHz
|
||||
// will be used.
|
||||
if (opus_max_playback_rate > 0) {
|
||||
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
|
||||
<< opus_max_playback_rate
|
||||
<< " Hz on channel "
|
||||
if (IsCodec(send_codec, kOpusCodecName)) {
|
||||
// DTX and maxplaybackrate should be set after SetSendCodec. Because current
|
||||
// send codec has to be Opus.
|
||||
|
||||
// Set Opus internal DTX.
|
||||
LOG(LS_INFO) << "Attempt to "
|
||||
<< GetEnableString(enable_opus_dtx)
|
||||
<< " Opus DTX on channel "
|
||||
<< channel;
|
||||
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
|
||||
channel, opus_max_playback_rate) == -1) {
|
||||
LOG(LS_WARNING) << "Could not set maximum playback rate.";
|
||||
if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) {
|
||||
LOG_RTCERR2(SetOpusDtx, channel, enable_opus_dtx);
|
||||
return false;
|
||||
}
|
||||
|
||||
// If opus_max_playback_rate <= 0, the default maximum playback rate
|
||||
// (48 kHz) will be used.
|
||||
if (opus_max_playback_rate > 0) {
|
||||
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
|
||||
<< opus_max_playback_rate
|
||||
<< " Hz on channel "
|
||||
<< channel;
|
||||
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
|
||||
channel, opus_max_playback_rate) == -1) {
|
||||
LOG_RTCERR2(SetOpusMaxPlaybackRate, channel, opus_max_playback_rate);
|
||||
return false;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@ -2241,13 +2241,13 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
|
||||
// Find the DTMF telephone event "codec" and tell VoiceEngine channels
|
||||
// about it.
|
||||
if (IsTelephoneEventCodec(it->name)) {
|
||||
if (IsCodec(*it, kDtmfCodecName)) {
|
||||
if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
|
||||
channel, it->id) == -1) {
|
||||
LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, it->id);
|
||||
return false;
|
||||
}
|
||||
} else if (IsCNCodec(it->name)) {
|
||||
} else if (IsCodec(*it, kCnCodecName)) {
|
||||
// Turn voice activity detection/comfort noise on if supported.
|
||||
// Set the wideband CN payload type appropriately.
|
||||
// (narrowband always uses the static payload type 13).
|
||||
@ -2285,7 +2285,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
}
|
||||
// Only turn on VAD if we have a CN payload type that matches the
|
||||
// clockrate for the codec we are going to use.
|
||||
if (it->clockrate == send_codec.plfreq) {
|
||||
if (it->clockrate == send_codec.plfreq && send_codec.channels != 2) {
|
||||
// TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
|
||||
// interaction between VAD and Opus FEC.
|
||||
LOG(LS_INFO) << "Enabling VAD";
|
||||
if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
|
||||
LOG_RTCERR2(SetVADStatus, channel, true);
|
||||
@ -2303,8 +2305,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
|
||||
it != codecs.end(); ++it) {
|
||||
// Find the DTMF telephone event "codec".
|
||||
if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
|
||||
_stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
|
||||
if (IsCodec(*it, kDtmfCodecName)) {
|
||||
dtmf_allowed_ = true;
|
||||
}
|
||||
}
|
||||
|
||||
@ -1402,6 +1402,53 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
|
||||
voe_.GetMaxEncodingBandwidth(channel_num));
|
||||
}
|
||||
|
||||
// Test that with usedtx=0, Opus DTX is off.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, DisableOpusDtxOnOpus) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
std::vector<cricket::AudioCodec> codecs;
|
||||
codecs.push_back(kOpusCodec);
|
||||
codecs[0].params["usedtx"] = "0";
|
||||
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
|
||||
EXPECT_FALSE(voe_.GetOpusDtx(channel_num));
|
||||
}
|
||||
|
||||
// Test that with usedtx=1, Opus DTX is on.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpus) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
std::vector<cricket::AudioCodec> codecs;
|
||||
codecs.push_back(kOpusCodec);
|
||||
codecs[0].params["usedtx"] = "1";
|
||||
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
|
||||
EXPECT_TRUE(voe_.GetOpusDtx(channel_num));
|
||||
EXPECT_FALSE(voe_.GetVAD(channel_num)); // Opus DTX should not affect VAD.
|
||||
}
|
||||
|
||||
// Test that usedtx=1 works with stereo Opus.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpusStereo) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
std::vector<cricket::AudioCodec> codecs;
|
||||
codecs.push_back(kOpusCodec);
|
||||
codecs[0].params["usedtx"] = "1";
|
||||
codecs[0].params["stereo"] = "1";
|
||||
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
|
||||
EXPECT_TRUE(voe_.GetOpusDtx(channel_num));
|
||||
EXPECT_FALSE(voe_.GetVAD(channel_num)); // Opus DTX should not affect VAD.
|
||||
}
|
||||
|
||||
// Test that usedtx=1 does not work with non Opus.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, CannotEnableOpusDtxOnNonOpus) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
std::vector<cricket::AudioCodec> codecs;
|
||||
codecs.push_back(kIsacCodec);
|
||||
codecs[0].params["usedtx"] = "1";
|
||||
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
|
||||
EXPECT_FALSE(voe_.GetOpusDtx(channel_num));
|
||||
}
|
||||
|
||||
// Test that we can switch back and forth between Opus and ISAC with CN.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
|
||||
Loading…
Reference in New Issue
Block a user