Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: https://chromium.googlesource.com/external/webrtc/+/1724cfbdbafef4736fedda37312b0286f1eb03d0
TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index f64d305..8c073d9 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -94,7 +94,10 @@
channel_proxy_->GetAudioDecoderFactory());
channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
- channel_proxy_->SetReceiveCodecs(config.decoder_map);
+
+ for (const auto& kv : config.decoder_map) {
+ channel_proxy_->SetRecPayloadType(kv.first, kv.second);
+ }
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 9025550..e6b81a8 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <map>
#include <string>
#include <vector>
@@ -112,11 +111,6 @@
.Times(1)
.After(expect_set);
EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
- EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
- .WillRepeatedly(
- Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
- EXPECT_THAT(codecs, testing::IsEmpty());
- }));
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index a45b91c..99d6812 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -9,7 +9,6 @@
*/
#include <list>
-#include <map>
#include <memory>
#include "webrtc/call/audio_state.h"
@@ -142,11 +141,6 @@
new testing::NiceMock<test::MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
.WillRepeatedly(testing::ReturnRef(decoder_factory));
- EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
- .WillRepeatedly(testing::Invoke(
- [](const std::map<int, SdpAudioFormat>& codecs) {
- EXPECT_THAT(codecs, testing::IsEmpty());
- }));
// If being called for the send channel, save a pointer to the channel
// proxy for later.
if (channel_id == kRecvChannelId) {
@@ -194,11 +188,6 @@
new testing::NiceMock<test::MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
.WillRepeatedly(testing::ReturnRef(decoder_factory));
- EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
- .WillRepeatedly(testing::Invoke(
- [](const std::map<int, SdpAudioFormat>& codecs) {
- EXPECT_THAT(codecs, testing::IsEmpty());
- }));
// If being called for the send channel, save a pointer to the channel
// proxy for later.
if (channel_id == kRecvChannelId) {
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 180d06b..e7d95d0 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -117,6 +117,8 @@
return -1;
}
Channel* ch = new Channel();
+ auto db = webrtc::acm2::RentACodec::Database();
+ ch->recv_codecs.assign(db.begin(), db.end());
ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
ch->neteq_fast_accelerate =
config.acm_config.neteq_config.enable_fast_accelerate;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 31a9d27..7c6d794 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -132,6 +132,13 @@
return ss.str();
}
+std::string ToString(const webrtc::CodecInst& codec) {
+ std::stringstream ss;
+ ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
+ << " (" << codec.pltype << ")";
+ return ss.str();
+}
+
bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
@@ -1459,8 +1466,7 @@
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call,
webrtc::Transport* rtcp_send_transport,
- const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
- const std::map<int, webrtc::SdpAudioFormat>& decoder_map)
+ const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
: call_(call), config_() {
RTC_DCHECK_GE(ch, 0);
RTC_DCHECK(call);
@@ -1473,7 +1479,6 @@
config_.voe_channel_id = ch;
config_.sync_group = sync_group;
config_.decoder_factory = decoder_factory;
- config_.decoder_map = decoder_map;
RecreateAudioReceiveStream();
}
@@ -1862,9 +1867,8 @@
ChangePlayout(false);
}
- decoder_map_ = std::move(decoder_map);
for (auto& kv : recv_streams_) {
- kv.second->RecreateAudioReceiveStream(decoder_map_);
+ kv.second->RecreateAudioReceiveStream(decoder_map);
}
recv_codecs_ = codecs;
@@ -2221,12 +2225,38 @@
return false;
}
+ // Turn off all supported codecs.
+ // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
+ for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+ voe_codec.pltype = -1;
+ if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ DeleteVoEChannel(channel);
+ return false;
+ }
+ }
+
+ // Only enable those configured for this channel.
+ for (const auto& codec : recv_codecs_) {
+ webrtc::CodecInst voe_codec = {0};
+ if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
+ voe_codec.pltype = codec.id;
+ if (engine()->voe()->codec()->SetRecPayloadType(
+ channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ DeleteVoEChannel(channel);
+ return false;
+ }
+ }
+ }
+
recv_streams_.insert(std::make_pair(
- ssrc,
- new WebRtcAudioReceiveStream(
- channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
- recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this,
- engine()->decoder_factory_, decoder_map_)));
+ ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
+ recv_transport_cc_enabled_,
+ recv_nack_enabled_,
+ sp.sync_label, recv_rtp_extensions_,
+ call_, this,
+ engine()->decoder_factory_)));
recv_streams_[ssrc]->SetPlayout(playout_);
return true;
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 64f7afa..c777c6c 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -251,12 +251,7 @@
WebRtcVoiceEngine* const engine_ = nullptr;
std::vector<AudioCodec> send_codecs_;
-
- // TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
- // information, in slightly different formats. Eliminate recv_codecs_.
- std::map<int, webrtc::SdpAudioFormat> decoder_map_;
std::vector<AudioCodec> recv_codecs_;
-
int max_send_bitrate_bps_ = 0;
AudioOptions options_;
rtc::Optional<int> dtmf_payload_type_;
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 3777b3d..2d41ecd 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -31,7 +31,6 @@
#include "webrtc/test/gtest.h"
#include "webrtc/voice_engine/transmit_mixer.h"
-using testing::ContainerEq;
using testing::Return;
using testing::StrictMock;
@@ -796,12 +795,26 @@
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {106, {"ISAC", 16000, 1}},
- {126, {"telephone-event", 8000, 1}},
- {107, {"telephone-event", 32000, 1}}})));
+ int channel_num = voe_.GetLastChannel();
+
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
+ gcodec.plfreq = 16000;
+ gcodec.channels = 1;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+ EXPECT_EQ(106, gcodec.pltype);
+ EXPECT_STREQ("ISAC", gcodec.plname);
+
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
+ gcodec.plfreq = 8000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+ EXPECT_EQ(126, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
+
+ gcodec.plfreq = 32000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+ EXPECT_EQ(107, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
}
// Test that we fail to set an unknown inbound codec.
@@ -832,11 +845,16 @@
parameters.codecs.push_back(kOpusCodec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2}}})));
+ int channel_num = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ // Even without stereo parameters, recv codecs still specify channels = 2.
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that we can decode OPUS with stereo = 0.
@@ -849,11 +867,16 @@
parameters.codecs[2].params["stereo"] = "0";
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ // Even when stereo is off, recv codecs still specify channels = 2.
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that we can decode OPUS with stereo = 1.
@@ -866,11 +889,15 @@
parameters.codecs[2].params["stereo"] = "1";
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {103, {"ISAC", 16000, 1}},
- {111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst opus;
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
+ EXPECT_EQ(2, opus.channels);
+ EXPECT_EQ(111, opus.pltype);
+ EXPECT_STREQ("opus", opus.plname);
+ opus.pltype = 0;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+ EXPECT_EQ(111, opus.pltype);
}
// Test that changes to recv codecs are applied to all streams.
@@ -884,15 +911,28 @@
parameters.codecs[0].id = 106; // collide with existing CN 32k
parameters.codecs[2].id = 126;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- for (const auto& ssrc : {kSsrcX, kSsrcY}) {
- EXPECT_TRUE(AddRecvStream(ssrc));
- EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}},
- {106, {"ISAC", 16000, 1}},
- {126, {"telephone-event", 8000, 1}},
- {107, {"telephone-event", 32000, 1}}})));
- }
+ EXPECT_TRUE(AddRecvStream(kSsrcX));
+ int channel_num2 = voe_.GetLastChannel();
+
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
+ gcodec.plfreq = 16000;
+ gcodec.channels = 1;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(106, gcodec.pltype);
+ EXPECT_STREQ("ISAC", gcodec.plname);
+
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
+ gcodec.plfreq = 8000;
+ gcodec.channels = 1;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(126, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
+
+ gcodec.plfreq = 32000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(107, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
}
TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
@@ -2921,9 +2961,12 @@
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_TRUE(AddRecvStream(kSsrcX));
- EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
- (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
- {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst gcodec;
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus");
+ gcodec.plfreq = 48000;
+ gcodec.channels = 2;
+ EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
}
// Test that we properly clean up any streams that were added, even if
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 4980d27..21dbc74 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -179,10 +179,6 @@
return 0;
}
-void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
- neteq_->SetCodecs(codecs);
-}
-
int32_t AcmReceiver::AddCodec(int acm_codec_id,
uint8_t payload_type,
size_t channels,
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 8f0963f..63ed43d 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -79,9 +79,6 @@
//
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
- // Replace the current set of decoders with the specified set.
- void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
-
//
// Adds a new codec to the NetEq codec database.
//
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index bc814f6..daeea35 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -121,8 +121,6 @@
// Get current playout frequency.
int PlayoutFrequency() const override;
- void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
-
bool RegisterReceiveCodec(int rtp_payload_type,
const SdpAudioFormat& audio_format) override;
@@ -320,6 +318,16 @@
webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
}
+// TODO(turajs): the same functionality is used in NetEq. If both classes
+// need them, make it a static function in ACMCodecDB.
+bool IsCodecRED(const CodecInst& codec) {
+ return (STR_CASE_CMP(codec.plname, "RED") == 0);
+}
+
+bool IsCodecCN(const CodecInst& codec) {
+ return (STR_CASE_CMP(codec.plname, "CN") == 0);
+}
+
// Stereo-to-mono can be used as in-place.
int DownMix(const AudioFrame& frame,
size_t length_out_buff,
@@ -948,6 +956,19 @@
receiver_.SetMaximumDelay(0);
receiver_.FlushBuffers();
+ // Register RED and CN.
+ auto db = acm2::RentACodec::Database();
+ for (size_t i = 0; i < db.size(); i++) {
+ if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
+ if (receiver_.AddCodec(static_cast<int>(i),
+ static_cast<uint8_t>(db[i].pltype), 1,
+ db[i].plfreq, nullptr, db[i].plname) < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+ "Cannot register master codec.");
+ return -1;
+ }
+ }
+ }
receiver_initialized_ = true;
return 0;
}
@@ -966,12 +987,6 @@
return receiver_.last_output_sample_rate_hz();
}
-void AudioCodingModuleImpl::SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) {
- rtc::CritScope lock(&acm_crit_sect_);
- receiver_.SetCodecs(codecs);
-}
-
bool AudioCodingModuleImpl::RegisterReceiveCodec(
int rtp_payload_type,
const SdpAudioFormat& audio_format) {
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index b5cbff2..1893b62 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -485,10 +485,6 @@
//
virtual int32_t PlayoutFrequency() const = 0;
- // Replace any existing decoders with the given payload type -> decoder map.
- virtual void SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) = 0;
-
// Registers a decoder for the given payload type. Returns true iff
// successful.
virtual bool RegisterReceiveCodec(int rtp_payload_type,
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.cc b/webrtc/modules/audio_coding/neteq/decoder_database.cc
index 8cd4386..d147d67 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -123,38 +123,6 @@
active_cng_decoder_type_ = -1;
}
-std::vector<int> DecoderDatabase::SetCodecs(
- const std::map<int, SdpAudioFormat>& codecs) {
- // First collect all payload types that we'll remove or reassign, then remove
- // them from the database.
- std::vector<int> changed_payload_types;
- for (const std::pair<uint8_t, const DecoderInfo&> kv : decoders_) {
- auto i = codecs.find(kv.first);
- if (i == codecs.end() || i->second != kv.second.GetFormat()) {
- changed_payload_types.push_back(kv.first);
- }
- }
- for (int pl_type : changed_payload_types) {
- Remove(pl_type);
- }
-
- // Enter the new and changed payload type mappings into the database.
- for (const auto& kv : codecs) {
- const int& rtp_payload_type = kv.first;
- const SdpAudioFormat& audio_format = kv.second;
- RTC_DCHECK_GE(rtp_payload_type, 0);
- RTC_DCHECK_LE(rtp_payload_type, 0x7f);
- if (decoders_.count(rtp_payload_type) == 0) {
- decoders_.insert(std::make_pair(
- rtp_payload_type, DecoderInfo(audio_format, decoder_factory_.get())));
- } else {
- // The mapping for this payload type hasn't changed.
- }
- }
-
- return changed_payload_types;
-}
-
int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
NetEqDecoder codec_type,
const std::string& name) {
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h
index 35d1d0d..ec470f8 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -149,11 +149,6 @@
// using InsertExternal().
virtual void Reset();
- // Replaces the existing set of decoders with the given set. Returns the
- // payload types that were reassigned or removed while doing so.
- virtual std::vector<int> SetCodecs(
- const std::map<int, SdpAudioFormat>& codecs);
-
// Registers |rtp_payload_type| as a decoder of type |codec_type|. The |name|
// is only used to populate the name field in the DecoderInfo struct in the
// database, and can be arbitrary (including empty). Returns kOK on success;
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 450318e..ea05940 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -157,9 +157,6 @@
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
- // Replaces the current set of decoders with the given one.
- virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
-
// Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
// information in the codec database. Returns 0 on success, -1 on failure.
// The name is only used to provide information back to the caller about the
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 501e567..5015b7e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -212,15 +212,6 @@
return kOK;
}
-void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
- rtc::CritScope lock(&crit_sect_);
- const std::vector<int> changed_payload_types =
- decoder_database_->SetCodecs(codecs);
- for (const int pt : changed_payload_types) {
- packet_buffer_->DiscardPacketsWithPayloadType(pt);
- }
-}
-
int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
const std::string& name,
uint8_t rtp_payload_type) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 88c0308..a8c3462 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -111,8 +111,6 @@
int GetAudio(AudioFrame* audio_frame, bool* muted) override;
- void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
-
int RegisterPayloadType(NetEqDecoder codec,
const std::string& codec_name,
uint8_t rtp_payload_type) override;
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index c1d69e0..541edd9 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -168,7 +168,6 @@
deps = [
"../..:webrtc_common",
"../../api:transport_api",
- "../../api/audio_codecs:audio_codecs_api",
"../../base:gtest_prod",
"../../base:rtc_base_approved",
"../../base:rtc_task_queue",
@@ -176,7 +175,6 @@
"../../common_video",
"../../logging:rtc_event_log_api",
"../../system_wrappers",
- "../audio_coding:audio_format_conversion",
"../remote_bitrate_estimator",
]
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 029de5d..767bd07 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -15,7 +15,6 @@
#include <memory>
#include <set>
-#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/deprecation.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -44,10 +43,6 @@
// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
// and simplify the code. http://crbug/webrtc/6743.
-
- // Replace all audio receive payload types with the given map.
- void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
-
int32_t RegisterReceivePayload(const CodecInst& audio_codec,
bool* created_new_payload_type);
int32_t RegisterReceivePayload(const VideoCodec& video_codec);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 234fd7f..1023bf8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
@@ -120,31 +119,6 @@
RTPPayloadRegistry::~RTPPayloadRegistry() = default;
-void RTPPayloadRegistry::SetAudioReceivePayloads(
- std::map<int, SdpAudioFormat> codecs) {
- rtc::CritScope cs(&crit_sect_);
-
-#if RTC_DCHECK_IS_ON
- RTC_DCHECK(!used_for_video_);
- used_for_audio_ = true;
-#endif
-
- payload_type_map_.clear();
- for (const auto& kv : codecs) {
- const int& rtp_payload_type = kv.first;
- const SdpAudioFormat& audio_format = kv.second;
- const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format);
- RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type));
- payload_type_map_.insert(
- std::make_pair(rtp_payload_type, CreatePayloadType(ci)));
- }
-
- // Clear the value of last received payload type since it might mean
- // something else now.
- last_received_payload_type_ = -1;
- last_received_media_payload_type_ = -1;
-}
-
int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
bool* created_new_payload) {
rtc::CritScope cs(&crit_sect_);
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index 450dd7b..d704e19 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -87,8 +87,6 @@
MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
MOCK_METHOD2(SetSendCNPayloadType,
bool(int type, PayloadFrequencies frequency));
- MOCK_METHOD1(SetReceiveCodecs,
- void(const std::map<int, SdpAudioFormat>& codecs));
MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
void(float recoverable_packet_loss_rate));
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index 6b1f9cf..ca3fe4f 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -52,11 +52,6 @@
new testing::NiceMock<webrtc::test::MockVoEChannelProxy>();
EXPECT_CALL(*proxy, GetAudioDecoderFactory())
.WillRepeatedly(testing::ReturnRef(decoder_factory_));
- EXPECT_CALL(*proxy, SetReceiveCodecs(testing::_))
- .WillRepeatedly(testing::Invoke(
- [](const std::map<int, SdpAudioFormat>& codecs) {
- EXPECT_THAT(codecs, testing::IsEmpty());
- }));
return proxy;
}));
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 3b562f9..d2b30ca 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -987,10 +987,9 @@
return -1;
}
- return 0;
-}
+ // --- Register all supported codecs to the receiving side of the
+ // RTP/RTCP module
-void Channel::RegisterLegacyCodecs() {
CodecInst codec;
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
@@ -1042,6 +1041,8 @@
}
}
}
+
+ return 0;
}
void Channel::Terminate() {
@@ -1359,11 +1360,6 @@
return 0;
}
-void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
- rtp_payload_registry_->SetAudioReceivePayloads(codecs);
- audio_coding_->SetReceiveCodecs(codecs);
-}
-
int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
return SetRecPayloadType(codec.pltype, CodecInstToSdp(codec));
}
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 1365ae8..0a12f21 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -151,7 +151,6 @@
uint32_t instanceId,
const VoEBase::ChannelConfig& config);
int32_t Init();
- void RegisterLegacyCodecs();
void Terminate();
int32_t SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
@@ -169,8 +168,6 @@
// go.
const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
- void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
-
// API methods
// VoEBase
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index d67011e..45cbf10 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -173,12 +173,6 @@
RTC_DCHECK_EQ(0, result);
}
-void ChannelProxy::SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- channel()->SetReceiveCodecs(codecs);
-}
-
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
@@ -385,11 +379,6 @@
channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
}
-void ChannelProxy::RegisterLegacyCodecs() {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- channel()->RegisterLegacyCodecs();
-}
-
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index b808096..685a168 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -82,7 +82,6 @@
virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
virtual void SetRecPayloadType(int payload_type,
const SdpAudioFormat& format);
- virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
virtual void SetInputMute(bool muted);
virtual void RegisterExternalTransport(Transport* transport);
@@ -120,7 +119,6 @@
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
virtual void OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate);
- virtual void RegisterLegacyCodecs();
private:
Channel* channel() const;
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index f1eee9c..b15d72f 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -15,8 +15,6 @@
#include "webrtc/base/byteorder.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/voice_engine/channel_proxy.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
namespace {
static const unsigned int kReflectorSsrc = 0x0000;
@@ -64,9 +62,6 @@
EXPECT_EQ(0, local_base_->Init());
local_sender_ = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(local_sender_)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
EXPECT_EQ(0, local_rtp_rtcp_->
@@ -77,9 +72,6 @@
EXPECT_EQ(0, remote_base_->Init());
reflector_ = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(reflector_)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
@@ -230,9 +222,6 @@
unsigned int ConferenceTransport::AddStream(std::string file_name,
webrtc::FileFormats format) {
const int new_sender = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(new_sender)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
@@ -246,9 +235,6 @@
new_sender, file_name.c_str(), true, false, format, 1.0));
const int new_receiver = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(new_receiver)
- ->RegisterLegacyCodecs();
EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
index 5582124..0fbe635 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
@@ -16,6 +16,5 @@
webrtc::VoiceEngineImpl* voe_impl =
static_cast<webrtc::VoiceEngineImpl*>(voice_engine_);
channel_proxy_ = voe_impl->GetChannelProxy(channel_);
- channel_proxy_->RegisterLegacyCodecs();
ResumePlaying();
}
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
index d9a5e2b..a733b12 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
@@ -8,9 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
class CodecBeforeStreamingTest : public AfterInitializationFixture {
protected:
@@ -21,9 +19,6 @@
codec_instance_.pacsize = 480;
channel_ = voe_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(voice_engine_)
- ->GetChannelProxy(channel_)
- ->RegisterLegacyCodecs();
}
void TearDown() {
