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Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )

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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: 1724cfbdba

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}
This commit is contained in:
kwiberg
2017-03-24 05:56:21 -07:00
committed by Commit bot
parent 6d7900de66
commit 670a7f3611
28 changed files with 150 additions and 221 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/media/engine/fakewebrtcvoiceengine.h
View File

@ -117,6 +117,8 @@ class FakeWebRtcVoiceEngine
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;

50
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -132,6 +132,13 @@ std::string ToString(const AudioCodec& codec) {
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 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
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 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
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 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
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 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
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;

5
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -251,12 +251,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
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_;

111
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
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 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
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