Remove RED support from WebRtcVoiceEngine/MediaChannel
This CL was originally written by solenberg@webrtc.org: https://codereview.webrtc.org/1928233003/ BUG=webrtc:4690, webrtc:5922 Review-Url: https://codereview.webrtc.org/2051073002 Cr-Commit-Position: refs/heads/master@{#13133}
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@ -50,7 +50,6 @@ std::string AudioSendStream::Config::ToString() const {
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ss << ", voe_channel_id: " << voe_channel_id;
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// TODO(solenberg): Encoder config.
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ss << ", cng_payload_type: " << cng_payload_type;
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ss << ", red_payload_type: " << red_payload_type;
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ss << '}';
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return ss.str();
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}
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@ -175,12 +175,10 @@ TEST(AudioSendStreamTest, ConfigToString) {
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config.rtp.c_name = kCName;
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config.voe_channel_id = kChannelId;
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config.cng_payload_type = 42;
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config.red_payload_type = 17;
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EXPECT_EQ(
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"{rtp: {ssrc: 1234, extensions: [{uri: "
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
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"c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
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"red_payload_type: 17}",
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"c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42}",
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config.ToString());
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}
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@ -85,7 +85,6 @@ class AudioSendStream {
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// TODO(solenberg): Implement, once we configure codecs through the new API.
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// std::unique_ptr<AudioEncoder> encoder;
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int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
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int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
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};
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// Starts stream activity.
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@ -134,11 +134,9 @@ class FakeWebRtcVoiceEngine
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bool codec_fec = false;
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int max_encoding_bandwidth = 0;
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bool opus_dtx = false;
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bool red = false;
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bool nack = false;
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int cn8_type = 13;
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int cn16_type = 105;
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int red_type = 117;
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int nack_max_packets = 0;
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uint32_t send_ssrc = 0;
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int associate_send_channel = -1;
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@ -172,9 +170,6 @@ class FakeWebRtcVoiceEngine
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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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bool GetCodecFEC(int channel) {
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return channels_[channel]->codec_fec;
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}
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@ -192,9 +187,6 @@ class FakeWebRtcVoiceEngine
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channels_[channel]->cn16_type :
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channels_[channel]->cn8_type;
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}
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int GetSendREDPayloadType(int channel) {
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return channels_[channel]->red_type;
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}
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void set_playout_fail_channel(int channel) {
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playout_fail_channel_ = channel;
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}
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@ -304,7 +296,7 @@ class FakeWebRtcVoiceEngine
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if (_stricmp(codec.plname, "telephone-event") == 0 ||
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_stricmp(codec.plname, "audio/telephone-event") == 0 ||
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_stricmp(codec.plname, "CN") == 0 ||
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_stricmp(codec.plname, "red") == 0 ) {
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_stricmp(codec.plname, "red") == 0) {
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return -1;
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}
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channels_[channel]->send_codec = codec;
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@ -483,18 +475,8 @@ class FakeWebRtcVoiceEngine
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unsigned int& maxJitterMs,
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unsigned int& discardedPackets));
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WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
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WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
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WEBRTC_CHECK_CHANNEL(channel);
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channels_[channel]->red = enable;
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channels_[channel]->red_type = redPayloadtype;
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return 0;
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}
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WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
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WEBRTC_CHECK_CHANNEL(channel);
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enable = channels_[channel]->red;
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redPayloadtype = channels_[channel]->red_type;
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return 0;
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}
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WEBRTC_STUB(SetREDStatus, (int channel, bool enable, int redPayloadtype));
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WEBRTC_STUB(GetREDStatus, (int channel, bool& enable, int& redPayloadtype));
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WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
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WEBRTC_CHECK_CHANNEL(channel);
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channels_[channel]->nack = enable;
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@ -384,13 +384,10 @@ class WebRtcVoiceCodecs final {
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static const AudioCodec* GetPreferredCodec(
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const std::vector<AudioCodec>& codecs,
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webrtc::CodecInst* out,
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int* red_payload_type) {
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webrtc::CodecInst* out) {
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RTC_DCHECK(out);
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RTC_DCHECK(red_payload_type);
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// Select the preferred send codec (the first non-telephone-event/CN codec).
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for (const AudioCodec& codec : codecs) {
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*red_payload_type = -1;
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if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
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// Skip telephone-event/CN codec, which will be handled later.
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continue;
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@ -398,27 +395,15 @@ class WebRtcVoiceCodecs final {
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// We'll use the first codec in the list to actually send audio data.
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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.
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const AudioCodec* found_codec = &codec;
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if (IsCodec(*found_codec, kRedCodecName)) {
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// Parse out the RED parameters. If we fail, just ignore RED;
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// we don't support all possible params/usage scenarios.
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*red_payload_type = codec.id;
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found_codec = GetRedSendCodec(*found_codec, codecs);
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if (!found_codec) {
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continue;
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}
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}
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// Ignore codecs we don't know about. The negotiation step should prevent
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// this, but double-check to be sure.
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webrtc::CodecInst voe_codec = {0};
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if (!ToCodecInst(*found_codec, &voe_codec)) {
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LOG(LS_WARNING) << "Unknown codec " << ToString(*found_codec);
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if (!ToCodecInst(codec, &voe_codec)) {
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LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
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continue;
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}
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*out = voe_codec;
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return found_codec;
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return &codec;
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}
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return nullptr;
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}
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@ -435,7 +420,7 @@ class WebRtcVoiceCodecs final {
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int max_bitrate_bps;
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};
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// Note: keep the supported packet sizes in ascending order.
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static const CodecPref kCodecPrefs[12];
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static const CodecPref kCodecPrefs[11];
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static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
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@ -458,47 +443,9 @@ class WebRtcVoiceCodecs final {
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voe_codec->plfreq = new_plfreq;
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}
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}
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static const AudioCodec* GetRedSendCodec(
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const AudioCodec& red_codec,
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const std::vector<AudioCodec>& all_codecs) {
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// Get the RED encodings from the parameter with no name. This may
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// change based on what is discussed on the Jingle list.
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// The encoding parameter is of the form "a/b"; we only support where
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// a == b. Verify this and parse out the value into red_pt.
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// If the parameter value is absent (as it will be until we wire up the
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// signaling of this message), use the second codec specified (i.e. the
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// one after "red") as the encoding parameter.
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int red_pt = -1;
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std::string red_params;
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CodecParameterMap::const_iterator it = red_codec.params.find("");
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if (it != red_codec.params.end()) {
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red_params = it->second;
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std::vector<std::string> red_pts;
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if (rtc::split(red_params, '/', &red_pts) != 2 ||
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red_pts[0] != red_pts[1] || !rtc::FromString(red_pts[0], &red_pt)) {
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LOG(LS_WARNING) << "RED params " << red_params << " not supported.";
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return nullptr;
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}
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} else if (red_codec.params.empty()) {
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LOG(LS_WARNING) << "RED params not present, using defaults";
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if (all_codecs.size() > 1) {
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red_pt = all_codecs[1].id;
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}
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}
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// Try to find red_pt in |codecs|.
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for (const AudioCodec& codec : all_codecs) {
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if (codec.id == red_pt) {
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return &codec;
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}
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}
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LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
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return nullptr;
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}
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};
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const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
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const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[11] = {
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{kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrate},
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{kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrate},
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{kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrate},
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@ -510,8 +457,7 @@ const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
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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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{kRedCodecName, 8000, 1, 127, false, {}},
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{kDtmfCodecName, 8000, 1, 126, false, {}},
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{kDtmfCodecName, 8000, 1, 126, false, {}}
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};
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} // namespace {
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@ -1670,46 +1616,41 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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}
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// Scan through the list to figure out the codec to use for sending, along
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// with the proper configuration for VAD, CNG, RED, NACK and Opus-specific
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// with the proper configuration for VAD, CNG, NACK and Opus-specific
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// parameters.
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// TODO(solenberg): Refactor this logic once we create AudioEncoders here.
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{
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SendCodecSpec send_codec_spec;
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send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
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// Find send codec (the first non-telephone-event/CN codec).
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const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
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codecs, &send_codec_spec.codec_inst, &send_codec_spec.red_payload_type);
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codecs, &send_codec_spec.codec_inst);
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if (!codec) {
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LOG(LS_WARNING) << "Received empty list of codecs.";
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return false;
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}
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send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
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send_codec_spec.nack_enabled = HasNack(*codec);
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// This condition is apparently here because Opus does not support RED and
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// FEC simultaneously. However, DTX and max playback rate shouldn't have
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// such limitations.
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// TODO(solenberg): Refactor this logic once we create AudioEncoders here.
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if (send_codec_spec.red_payload_type == -1) {
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send_codec_spec.nack_enabled = HasNack(*codec);
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// For Opus as the send codec, we are to determine inband FEC, maximum
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// playback rate, and opus internal dtx.
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if (IsCodec(*codec, kOpusCodecName)) {
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GetOpusConfig(*codec, &send_codec_spec.codec_inst,
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&send_codec_spec.enable_codec_fec,
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&send_codec_spec.opus_max_playback_rate,
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&send_codec_spec.enable_opus_dtx);
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}
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// For Opus as the send codec, we are to determine inband FEC, maximum
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// playback rate, and opus internal dtx.
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if (IsCodec(*codec, kOpusCodecName)) {
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GetOpusConfig(*codec, &send_codec_spec.codec_inst,
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&send_codec_spec.enable_codec_fec,
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&send_codec_spec.opus_max_playback_rate,
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&send_codec_spec.enable_opus_dtx);
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}
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// Set packet size if the AudioCodec param kCodecParamPTime is set.
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int ptime_ms = 0;
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if (codec->GetParam(kCodecParamPTime, &ptime_ms)) {
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if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(
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&send_codec_spec.codec_inst, ptime_ms)) {
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LOG(LS_WARNING) << "Failed to set packet size for codec "
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<< send_codec_spec.codec_inst.plname;
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return false;
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}
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// Set packet size if the AudioCodec param kCodecParamPTime is set.
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int ptime_ms = 0;
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if (codec->GetParam(kCodecParamPTime, &ptime_ms)) {
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if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(
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&send_codec_spec.codec_inst, ptime_ms)) {
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LOG(LS_WARNING) << "Failed to set packet size for codec "
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<< send_codec_spec.codec_inst.plname;
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return false;
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}
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}
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@ -1781,24 +1722,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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bool WebRtcVoiceMediaChannel::SetSendCodecs(
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int channel,
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const webrtc::RtpParameters& rtp_parameters) {
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// Disable VAD, FEC, and RED unless we know the other side wants them.
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// Disable VAD, NACK and FEC unless we know the other side wants them.
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engine()->voe()->codec()->SetVADStatus(channel, false);
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engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
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engine()->voe()->rtp()->SetREDStatus(channel, false);
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engine()->voe()->codec()->SetFECStatus(channel, false);
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if (send_codec_spec_.red_payload_type != -1) {
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// Enable redundant encoding of the specified codec. Treat any
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// failure as a fatal internal error.
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LOG(LS_INFO) << "Enabling RED on channel " << channel;
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if (engine()->voe()->rtp()->SetREDStatus(channel, true,
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send_codec_spec_.red_payload_type) == -1) {
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LOG_RTCERR3(SetREDStatus, channel, true,
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send_codec_spec_.red_payload_type);
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return false;
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}
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}
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SetNack(channel, send_codec_spec_.nack_enabled);
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// Set the codec immediately, since SetVADStatus() depends on whether
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@ -295,7 +295,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
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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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int red_payload_type = -1;
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int cng_payload_type = -1;
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int cng_plfreq = -1;
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webrtc::CodecInst codec_inst;
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@ -38,7 +38,6 @@ const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
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const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2);
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const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
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const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
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const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1);
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const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
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const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
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const cricket::AudioCodec kTelephoneEventCodec(106,
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@ -1045,7 +1044,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
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cricket::AudioSendParameters parameters;
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parameters.codecs.push_back(kIsacCodec);
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parameters.codecs.push_back(kPcmuCodec);
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parameters.codecs.push_back(kRedCodec);
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parameters.codecs.push_back(kCn8000Codec);
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parameters.codecs[0].id = 96;
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parameters.codecs[0].bitrate = 48000;
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EXPECT_TRUE(channel_->SetSendParameters(parameters));
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@ -1057,7 +1056,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
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EXPECT_EQ(48000, gcodec.rate);
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EXPECT_STREQ("ISAC", gcodec.plname);
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EXPECT_FALSE(voe_.GetVAD(channel_num));
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EXPECT_FALSE(voe_.GetRED(channel_num));
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EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
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EXPECT_EQ(105, voe_.GetSendCNPayloadType(channel_num, true));
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EXPECT_FALSE(channel_->CanInsertDtmf());
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@ -1070,7 +1068,7 @@ TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) {
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cricket::AudioSendParameters parameters;
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parameters.codecs.push_back(kIsacCodec);
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parameters.codecs.push_back(kPcmuCodec);
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parameters.codecs.push_back(kRedCodec);
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parameters.codecs.push_back(kCn8000Codec);
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parameters.codecs[0].id = 96;
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parameters.codecs[0].bitrate = 48000;
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EXPECT_TRUE(channel_->SetSendParameters(parameters));
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@ -1979,7 +1977,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
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parameters.codecs.push_back(kCn16000Codec);
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parameters.codecs.push_back(kCn8000Codec);
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parameters.codecs.push_back(kTelephoneEventCodec);
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parameters.codecs.push_back(kRedCodec);
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parameters.codecs[0].id = 96;
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parameters.codecs[2].id = 97; // wideband CN
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parameters.codecs[4].id = 98; // DTMF
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@ -1989,7 +1986,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
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EXPECT_EQ(96, gcodec.pltype);
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EXPECT_STREQ("ISAC", gcodec.plname);
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EXPECT_TRUE(voe_.GetVAD(channel_num));
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EXPECT_FALSE(voe_.GetRED(channel_num));
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EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
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EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
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EXPECT_TRUE(channel_->CanInsertDtmf());
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@ -2005,7 +2001,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
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parameters.codecs.push_back(kCn16000Codec);
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parameters.codecs.push_back(kCn8000Codec);
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parameters.codecs.push_back(kTelephoneEventCodec);
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parameters.codecs.push_back(kRedCodec);
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parameters.codecs[0].id = 96;
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parameters.codecs[2].id = 97; // wideband CN
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parameters.codecs[4].id = 98; // DTMF
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@ -2019,7 +2014,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
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EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetVAD(channel_num));
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
|
||||
EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
|
||||
EXPECT_TRUE(channel_->CanInsertDtmf());
|
||||
@ -2072,7 +2066,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
|
||||
parameters.codecs.push_back(kCn16000Codec);
|
||||
parameters.codecs.push_back(kCn8000Codec);
|
||||
parameters.codecs.push_back(kTelephoneEventCodec);
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs[0].name = "iSaC";
|
||||
parameters.codecs[0].id = 96;
|
||||
parameters.codecs[2].id = 97; // wideband CN
|
||||
@ -2083,168 +2076,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetVAD(channel_num));
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
|
||||
EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
|
||||
EXPECT_TRUE(channel_->CanInsertDtmf());
|
||||
}
|
||||
|
||||
// Test that we set up RED correctly as caller.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "96/96";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetRED(channel_num));
|
||||
EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
|
||||
}
|
||||
|
||||
// Test that we set up RED correctly as callee.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCallee) {
|
||||
EXPECT_TRUE(SetupChannel());
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "96/96";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
EXPECT_TRUE(channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrc1)));
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetRED(channel_num));
|
||||
EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
|
||||
}
|
||||
|
||||
// Test that we set up RED correctly if params are omitted.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDNoParams) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetRED(channel_num));
|
||||
EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
|
||||
}
|
||||
|
||||
// Test that we ignore RED if the parameters aren't named the way we expect.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED1) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params["ABC"] = "96/96";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
}
|
||||
|
||||
// Test that we ignore RED if it uses different primary/secondary encoding.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED2) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "96/0";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
}
|
||||
|
||||
// Test that we ignore RED if it uses more than 2 encodings.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED3) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "96/96/96";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
}
|
||||
|
||||
// Test that we ignore RED if it has bogus codec ids.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED4) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "ABC/ABC";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
}
|
||||
|
||||
// Test that we ignore RED if it refers to a codec that is not present.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
cricket::AudioSendParameters parameters;
|
||||
parameters.codecs.push_back(kRedCodec);
|
||||
parameters.codecs.push_back(kIsacCodec);
|
||||
parameters.codecs.push_back(kPcmuCodec);
|
||||
parameters.codecs[0].id = 127;
|
||||
parameters.codecs[0].params[""] = "97/97";
|
||||
parameters.codecs[1].id = 96;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
webrtc::CodecInst gcodec;
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_EQ(96, gcodec.pltype);
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetRED(channel_num));
|
||||
}
|
||||
|
||||
class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake {
|
||||
public:
|
||||
WebRtcVoiceEngineWithSendSideBweTest()
|
||||
@ -3602,8 +3438,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
|
||||
cricket::AudioCodec(96, "PCMA", 8000, 0, 1), nullptr));
|
||||
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
|
||||
cricket::AudioCodec(96, "G722", 8000, 0, 1), nullptr));
|
||||
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
|
||||
cricket::AudioCodec(96, "red", 8000, 0, 1), nullptr));
|
||||
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
|
||||
cricket::AudioCodec(96, "CN", 32000, 0, 1), nullptr));
|
||||
EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
|
||||
@ -3653,8 +3487,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
|
||||
EXPECT_EQ(9, it->id);
|
||||
} else if (it->name == "telephone-event") {
|
||||
EXPECT_EQ(126, it->id);
|
||||
} else if (it->name == "red") {
|
||||
EXPECT_EQ(127, it->id);
|
||||
} else if (it->name == "opus") {
|
||||
EXPECT_EQ(111, it->id);
|
||||
ASSERT_TRUE(it->params.find("minptime") != it->params.end());
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user