Reland "red: generate and parse the red fmtp format"
This is a reland of 9d0730942677a520ce7e184d081b4c5a2469fc48
Original change's description:
> red: generate and parse the red fmtp format
>
> generates a fmtp line like
> a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
> and matches the incoming redundant payload types against the
> send codec one. Offers without an FMTP line will not use RED.
> Redundancy levels of 1 (plus main packet ) to 32 are accepted but
> this is not wired up to the encoder since the O/A semantic of
> RFC 2198 is not clear.
>
> This decreases the chance of a collision with the SATIN codec
> which also runs on 48khz (but so far does not specify a channelCount of 2)
>
> BUG=webrtc:11640
>
> Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34848}
Bug: webrtc:11640
Change-Id: I9465e489897a8ded9845592477fe14678af7ab61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230545
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34965}
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 75d1a5f..da5f594 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -777,7 +777,9 @@
out.push_back(codec);
if (codec.name == kOpusCodecName && audio_red_for_opus_enabled_) {
- map_format({kRedCodecName, 48000, 2}, &out);
+ std::string redFmtp =
+ rtc::ToString(codec.id) + "/" + rtc::ToString(codec.id);
+ map_format({kRedCodecName, 48000, 2, {{"", redFmtp}}}, &out);
}
}
}
@@ -1661,6 +1663,37 @@
return true;
}
+// Utility function to check if RED codec and its parameters match a codec spec.
+bool CheckRedParameters(
+ const AudioCodec& red_codec,
+ const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
+ if (red_codec.clockrate != send_codec_spec.format.clockrate_hz ||
+ red_codec.channels != send_codec_spec.format.num_channels) {
+ return false;
+ }
+
+ // Check the FMTP line for the empty parameter which should match
+ // <primary codec>/<primary codec>[/...]
+ auto red_parameters = red_codec.params.find("");
+ if (red_parameters == red_codec.params.end()) {
+ RTC_LOG(LS_WARNING) << "audio/RED missing fmtp parameters.";
+ return false;
+ }
+ std::vector<std::string> redundant_payloads;
+ rtc::split(red_parameters->second, '/', &redundant_payloads);
+ // 32 is chosen as a maximum upper bound for consistency with the
+ // red payload splitter.
+ if (redundant_payloads.size() < 2 || redundant_payloads.size() > 32) {
+ return false;
+ }
+ for (auto pt : redundant_payloads) {
+ if (pt != rtc::ToString(send_codec_spec.payload_type)) {
+ return false;
+ }
+ }
+ return true;
+}
+
// Utility function called from SetSendParameters() to extract current send
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.
@@ -1772,8 +1805,7 @@
for (const AudioCodec& red_codec : codecs) {
if (red_codec_position < send_codec_position &&
IsCodec(red_codec, kRedCodecName) &&
- red_codec.clockrate == send_codec_spec->format.clockrate_hz &&
- red_codec.channels == send_codec_spec->format.num_channels) {
+ CheckRedParameters(red_codec, *send_codec_spec)) {
send_codec_spec->red_payload_type = red_codec.id;
break;
}
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index f1c0c1e..8007210 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1028,10 +1028,12 @@
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs[1].params[""] = "111/111";
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{111, {"opus", 48000, 2}}, {112, {"red", 48000, 2}}})));
+ {{111, {"opus", 48000, 2}},
+ {112, {"red", 48000, 2, {{"", "111/111"}}}}})));
}
// Test that we disable Opus/Red with the kill switch.
@@ -1495,15 +1497,13 @@
EXPECT_FALSE(channel_->CanInsertDtmf());
}
-// Test that we use Opus/Red under the field trial when it is
-// listed as the first codec.
+// Test that we use Opus/Red by default when it is
+// listed as the first codec and there is an fmtp line.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) {
- webrtc::test::ScopedFieldTrials override_field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled/");
-
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs[0].params[""] = "111/111";
parameters.codecs.push_back(kOpusCodec);
SetSendParameters(parameters);
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
@@ -1512,15 +1512,13 @@
EXPECT_EQ(112, send_codec_spec.red_payload_type);
}
-// Test that we do not use Opus/Red under the field trial by default.
-TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) {
- webrtc::test::ScopedFieldTrials override_field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled/");
-
+// Test that we do not use Opus/Red by default when it is
+// listed as the first codec but there is no fmtp line.
+TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
- parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs.push_back(kOpusCodec);
SetSendParameters(parameters);
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec.payload_type);
@@ -1528,6 +1526,62 @@
EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
}
+// Test that we do not use Opus/Red by default.
+TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) {
+ EXPECT_TRUE(SetupSendStream());
+ cricket::AudioSendParameters parameters;
+ parameters.codecs.push_back(kOpusCodec);
+ parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs[1].params[""] = "111/111";
+ SetSendParameters(parameters);
+ const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+ EXPECT_EQ(111, send_codec_spec.payload_type);
+ EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
+ EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+}
+
+// Test that the RED fmtp line must match the payload type.
+TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) {
+ EXPECT_TRUE(SetupSendStream());
+ cricket::AudioSendParameters parameters;
+ parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs[0].params[""] = "8/8";
+ parameters.codecs.push_back(kOpusCodec);
+ SetSendParameters(parameters);
+ const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+ EXPECT_EQ(111, send_codec_spec.payload_type);
+ EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
+ EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+}
+
+// Test that the RED fmtp line must show 2..32 payloads.
+TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) {
+ EXPECT_TRUE(SetupSendStream());
+ cricket::AudioSendParameters parameters;
+ parameters.codecs.push_back(kRed48000Codec);
+ parameters.codecs[0].params[""] = "111";
+ parameters.codecs.push_back(kOpusCodec);
+ SetSendParameters(parameters);
+ const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+ EXPECT_EQ(111, send_codec_spec.payload_type);
+ EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
+ EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+ for (int i = 1; i < 32; i++) {
+ parameters.codecs[0].params[""] += "/111";
+ SetSendParameters(parameters);
+ const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+ EXPECT_EQ(111, send_codec_spec.payload_type);
+ EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
+ EXPECT_EQ(112, send_codec_spec.red_payload_type);
+ }
+ parameters.codecs[0].params[""] += "/111";
+ SetSendParameters(parameters);
+ const auto& send_codec_spec2 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+ EXPECT_EQ(111, send_codec_spec2.payload_type);
+ EXPECT_STRCASEEQ("opus", send_codec_spec2.format.name.c_str());
+ EXPECT_EQ(absl::nullopt, send_codec_spec2.red_payload_type);
+}
+
// Test that WebRtcVoiceEngine reconfigures, rather than recreates its
// AudioSendStream.
TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
