Rename cricket::RtpParameters and derived classes

Renames
  cricket::RtpParameters
to
  cricket::MediaChannelParameters
in order to distinguish it better from webrtc::RtpParameters.
This involves renaming
  RtpSendParameters -> SenderParameters
  AudioSendParameters -> AudioSenderParameters
  AudioRecvParameters -> AudioReceiverParameters
  VideoSendParameters -> VideoSenderParameters
  VideoRecvParameters -> VideoReceiverParameters

BUG=webrtc:13931

Change-Id: I664595ee3863418c0c6ca092ca77127be0f9498f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40497}
This commit is contained in:
Philipp Hancke 2023-07-31 13:27:37 +02:00 committed by WebRTC LUCI CQ
parent 787e326c18
commit a9d5141367
14 changed files with 278 additions and 284 deletions

View File

@ -77,7 +77,7 @@ int FakeVoiceMediaReceiveChannel::max_bps() const {
return max_bps_;
}
bool FakeVoiceMediaReceiveChannel::SetRecvParameters(
const AudioRecvParameters& params) {
const AudioReceiverParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
@ -236,7 +236,7 @@ int FakeVoiceMediaSendChannel::max_bps() const {
return max_bps_;
}
bool FakeVoiceMediaSendChannel::SetSendParameters(
const AudioSendParameters& params) {
const AudioSenderParameter& params) {
set_send_rtcp_parameters(params.rtcp);
SetExtmapAllowMixed(params.extmap_allow_mixed);
return (SetSendCodecs(params.codecs) &&
@ -356,7 +356,7 @@ int FakeVideoMediaSendChannel::max_bps() const {
return max_bps_;
}
bool FakeVideoMediaSendChannel::SetSendParameters(
const VideoSendParameters& params) {
const VideoSenderParameters& params) {
set_send_rtcp_parameters(params.rtcp);
SetExtmapAllowMixed(params.extmap_allow_mixed);
return (SetSendCodecs(params.codecs) &&
@ -442,7 +442,7 @@ int FakeVideoMediaReceiveChannel::max_bps() const {
return max_bps_;
}
bool FakeVideoMediaReceiveChannel::SetRecvParameters(
const VideoRecvParameters& params) {
const VideoReceiverParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));

View File

@ -469,7 +469,7 @@ class FakeVoiceMediaReceiveChannel
return cricket::MEDIA_TYPE_AUDIO;
}
bool SetRecvParameters(const AudioRecvParameters& params) override;
bool SetRecvParameters(const AudioReceiverParameters& params) override;
void SetPlayout(bool playout) override;
bool AddRecvStream(const StreamParams& sp) override;
@ -559,7 +559,7 @@ class FakeVoiceMediaSendChannel
return cricket::MEDIA_TYPE_AUDIO;
}
bool SetSendParameters(const AudioSendParameters& params) override;
bool SetSendParameters(const AudioSenderParameter& params) override;
void SetSend(bool send) override;
bool SetAudioSend(uint32_t ssrc,
bool enable,
@ -644,7 +644,7 @@ class FakeVideoMediaReceiveChannel
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
sinks() const;
int max_bps() const;
bool SetRecvParameters(const VideoRecvParameters& params) override;
bool SetRecvParameters(const VideoReceiverParameters& params) override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
@ -717,7 +717,7 @@ class FakeVideoMediaSendChannel
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
sinks() const;
int max_bps() const;
bool SetSendParameters(const VideoSendParameters& params) override;
bool SetSendParameters(const VideoSenderParameters& params) override;
absl::optional<Codec> GetSendCodec() const override;

View File

@ -792,7 +792,7 @@ struct VideoMediaInfo {
send_codecs.clear();
receive_codecs.clear();
}
// Each sender info represents one "outbound-rtp" stream.In non - simulcast,
// Each sender info represents one "outbound-rtp" stream. In non-simulcast,
// this means one info per RtpSender but if simulcast is used this means
// one info per simulcast layer.
std::vector<VideoSenderInfo> senders;
@ -811,8 +811,8 @@ struct RtcpParameters {
bool remote_estimate = false;
};
struct RtpParameters {
virtual ~RtpParameters() = default;
struct MediaChannelParameters {
virtual ~MediaChannelParameters() = default;
std::vector<Codec> codecs;
std::vector<webrtc::RtpExtension> extensions;
@ -842,9 +842,7 @@ struct RtpParameters {
}
};
// TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
// encapsulate all the parameters needed for an RtpSender.
struct RtpSendParameters : RtpParameters {
struct SenderParameters : MediaChannelParameters {
int max_bandwidth_bps = -1;
// This is the value to be sent in the MID RTP header extension (if the header
// extension in included in the list of extensions).
@ -853,7 +851,7 @@ struct RtpSendParameters : RtpParameters {
protected:
std::map<std::string, std::string> ToStringMap() const override {
auto params = RtpParameters::ToStringMap();
auto params = MediaChannelParameters::ToStringMap();
params["max_bandwidth_bps"] = rtc::ToString(max_bandwidth_bps);
params["mid"] = (mid.empty() ? "<not set>" : mid);
params["extmap-allow-mixed"] = extmap_allow_mixed ? "true" : "false";
@ -861,20 +859,20 @@ struct RtpSendParameters : RtpParameters {
}
};
struct AudioSendParameters : RtpSendParameters {
AudioSendParameters();
~AudioSendParameters() override;
struct AudioSenderParameter : SenderParameters {
AudioSenderParameter();
~AudioSenderParameter() override;
AudioOptions options;
protected:
std::map<std::string, std::string> ToStringMap() const override;
};
struct AudioRecvParameters : RtpParameters {};
struct AudioReceiverParameters : MediaChannelParameters {};
class VoiceMediaSendChannelInterface : public MediaSendChannelInterface {
public:
virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
virtual bool SetSendParameters(const AudioSenderParameter& params) = 0;
// Starts or stops sending (and potentially capture) of local audio.
virtual void SetSend(bool send) = 0;
// Configure stream for sending.
@ -896,7 +894,7 @@ class VoiceMediaSendChannelInterface : public MediaSendChannelInterface {
class VoiceMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
public:
virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
virtual bool SetRecvParameters(const AudioReceiverParameters& params) = 0;
// Get the receive parameters for the incoming stream identified by `ssrc`.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;
@ -920,11 +918,9 @@ class VoiceMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
virtual void SetReceiveNonSenderRttEnabled(bool enabled) = 0;
};
// TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpSender.
struct VideoSendParameters : RtpSendParameters {
VideoSendParameters();
~VideoSendParameters() override;
struct VideoSenderParameters : SenderParameters {
VideoSenderParameters();
~VideoSenderParameters() override;
// Use conference mode? This flag comes from the remote
// description's SDP line 'a=x-google-flag:conference', copied over
// by VideoChannel::SetRemoteContent_w, and ultimately used by
@ -937,13 +933,11 @@ struct VideoSendParameters : RtpSendParameters {
std::map<std::string, std::string> ToStringMap() const override;
};
// TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpReceiver.
struct VideoRecvParameters : RtpParameters {};
struct VideoReceiverParameters : MediaChannelParameters {};
class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
public:
virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
virtual bool SetSendParameters(const VideoSenderParameters& params) = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
// Configure stream for sending and register a source.
@ -975,7 +969,7 @@ class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
public:
virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
virtual bool SetRecvParameters(const VideoReceiverParameters& params) = 0;
// Get the receive parameters for the incoming stream identified by `ssrc`.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;

View File

@ -167,20 +167,20 @@ VideoMediaReceiveInfo::~VideoMediaReceiveInfo() = default;
VoiceMediaReceiveInfo::VoiceMediaReceiveInfo() = default;
VoiceMediaReceiveInfo::~VoiceMediaReceiveInfo() = default;
AudioSendParameters::AudioSendParameters() = default;
AudioSendParameters::~AudioSendParameters() = default;
AudioSenderParameter::AudioSenderParameter() = default;
AudioSenderParameter::~AudioSenderParameter() = default;
std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
auto params = RtpSendParameters::ToStringMap();
std::map<std::string, std::string> AudioSenderParameter::ToStringMap() const {
auto params = SenderParameters::ToStringMap();
params["options"] = options.ToString();
return params;
}
VideoSendParameters::VideoSendParameters() = default;
VideoSendParameters::~VideoSendParameters() = default;
VideoSenderParameters::VideoSenderParameters() = default;
VideoSenderParameters::~VideoSenderParameters() = default;
std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
auto params = RtpSendParameters::ToStringMap();
std::map<std::string, std::string> VideoSenderParameters::ToStringMap() const {
auto params = SenderParameters::ToStringMap();
params["conference_mode"] = (conference_mode ? "yes" : "no");
return params;
}

View File

@ -972,7 +972,7 @@ std::vector<VideoCodecSettings> WebRtcVideoSendChannel::SelectSendVideoCodecs(
}
bool WebRtcVideoSendChannel::GetChangedSendParameters(
const VideoSendParameters& params,
const VideoSenderParameters& params,
ChangedSendParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions, send_rtp_extensions_)) {
@ -1045,7 +1045,7 @@ bool WebRtcVideoSendChannel::GetChangedSendParameters(
}
bool WebRtcVideoSendChannel::SetSendParameters(
const VideoSendParameters& params) {
const VideoSenderParameters& params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoSendChannel::SetSendParameters");
RTC_LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
@ -1594,7 +1594,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
// TODO(deadbeef): Don't duplicate information between send_params,
// rtp_extensions, options, etc.
const VideoSendParameters& send_params)
const VideoSenderParameters& send_params)
: worker_thread_(call->worker_thread()),
ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
@ -2554,7 +2554,7 @@ WebRtcVideoReceiveChannel::GetDefaultRtpReceiveParameters() const {
}
bool WebRtcVideoReceiveChannel::GetChangedRecvParameters(
const VideoRecvParameters& params,
const VideoReceiverParameters& params,
ChangedRecvParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions, recv_rtp_extensions_)) {
@ -2609,7 +2609,7 @@ bool WebRtcVideoReceiveChannel::GetChangedRecvParameters(
}
bool WebRtcVideoReceiveChannel::SetRecvParameters(
const VideoRecvParameters& params) {
const VideoReceiverParameters& params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoReceiveChannel::SetRecvParameters");
RTC_LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();

View File

@ -190,7 +190,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
// Common functions between sender and receiver
void SetInterface(MediaChannelNetworkInterface* iface) override;
// VideoMediaSendChannelInterface implementation
bool SetSendParameters(const VideoSendParameters& params) override;
bool SetSendParameters(const VideoSenderParameters& params) override;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
@ -308,7 +308,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
absl::optional<webrtc::RtcpMode> rtcp_mode;
};
bool GetChangedSendParameters(const VideoSendParameters& params,
bool GetChangedSendParameters(const VideoSenderParameters& params,
ChangedSendParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
bool ApplyChangedParams(const ChangedSendParameters& changed_params);
@ -336,7 +336,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
int max_bitrate_bps,
const absl::optional<VideoCodecSettings>& codec_settings,
const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
const VideoSendParameters& send_params);
const VideoSenderParameters& send_params);
~WebRtcVideoSendStream();
void SetSendParameters(const ChangedSendParameters& send_params);
@ -530,9 +530,9 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
webrtc::BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
// TODO(deadbeef): Don't duplicate information between
// send_params/recv_params, rtp_extensions, options, etc.
VideoSendParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
VideoRecvParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
int64_t last_send_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
int64_t last_receive_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
const bool discard_unknown_ssrc_packets_ RTC_GUARDED_BY(thread_checker_);
@ -590,7 +590,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
// Common functions between sender and receiver
void SetInterface(MediaChannelNetworkInterface* iface) override;
// VideoMediaReceiveChannelInterface implementation
bool SetRecvParameters(const VideoRecvParameters& params) override;
bool SetRecvParameters(const VideoReceiverParameters& params) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
void SetReceive(bool receive) override;
@ -786,7 +786,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
bool receiving_ RTC_GUARDED_BY(&thread_checker_);
};
bool GetChangedRecvParameters(const VideoRecvParameters& params,
bool GetChangedRecvParameters(const VideoReceiverParameters& params,
ChangedRecvParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
@ -862,9 +862,9 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
webrtc::BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
// TODO(deadbeef): Don't duplicate information between
// send_params/recv_params, rtp_extensions, options, etc.
VideoSendParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
VideoRecvParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
int64_t last_receive_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
const bool discard_unknown_ssrc_packets_ RTC_GUARDED_BY(thread_checker_);
// This is a stream param that comes from the remote description, but wasn't

File diff suppressed because it is too large Load Diff

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@ -1263,7 +1263,7 @@ bool WebRtcVoiceSendChannel::SetOptions(const AudioOptions& options) {
}
bool WebRtcVoiceSendChannel::SetSendParameters(
const AudioSendParameters& params) {
const AudioSenderParameter& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters");
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
@ -1998,7 +1998,7 @@ WebRtcVoiceReceiveChannel::~WebRtcVoiceReceiveChannel() {
}
bool WebRtcVoiceReceiveChannel::SetRecvParameters(
const AudioRecvParameters& params) {
const AudioReceiverParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "

View File

@ -227,7 +227,7 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
const AudioOptions& options() const { return options_; }
bool SetSendParameters(const AudioSendParameters& params) override;
bool SetSendParameters(const AudioSenderParameter& params) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
@ -377,7 +377,7 @@ class WebRtcVoiceReceiveChannel final
void SetInterface(MediaChannelNetworkInterface* iface) override {
MediaChannelUtil::SetInterface(iface);
}
bool SetRecvParameters(const AudioRecvParameters& params) override;
bool SetRecvParameters(const AudioReceiverParameters& params) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;

View File

@ -338,7 +338,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
send_channel_->SetSend(enable);
}
void SetSendParameters(const cricket::AudioSendParameters& params) {
void SetSendParameters(const cricket::AudioSenderParameter& params) {
ASSERT_TRUE(send_channel_);
EXPECT_TRUE(send_channel_->SetSendParameters(params));
}
@ -429,7 +429,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
int max_bitrate,
bool expected_result,
int expected_bitrate) {
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(codec);
parameters.max_bandwidth_bps = max_bitrate;
if (expected_result) {
@ -451,7 +451,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
}
void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
cricket::AudioSendParameters send_parameters;
cricket::AudioSenderParameter send_parameters;
send_parameters.codecs.push_back(codec);
send_parameters.max_bandwidth_bps = bitrate;
SetSendParameters(send_parameters);
@ -847,8 +847,8 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel_;
std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> receive_channel_;
cricket::AudioSendParameters send_parameters_;
cricket::AudioRecvParameters recv_parameters_;
cricket::AudioSenderParameter send_parameters_;
cricket::AudioReceiverParameters recv_parameters_;
FakeAudioSource fake_source_;
webrtc::AudioProcessing::Config apm_config_;
};
@ -901,7 +901,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
// Test that we set our inbound codecs properly, including changing PT.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kTelephoneEventCodec1);
@ -921,7 +921,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
// Test that we fail to set an unknown inbound codec.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(cricket::CreateAudioCodec(127, "XYZ", 32000, 1));
EXPECT_FALSE(receive_channel_->SetRecvParameters(parameters));
@ -930,7 +930,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
// Test that we fail if we have duplicate types in the inbound list.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs[1].id = kOpusCodec.id;
@ -940,7 +940,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) {
// Test that we can decode OPUS without stereo parameters.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kOpusCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -953,7 +953,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
// Test that we can decode OPUS with stereo = 0.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[1].params["stereo"] = "0";
@ -968,7 +968,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
// Test that we can decode OPUS with stereo = 1.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[1].params["stereo"] = "1";
@ -983,7 +983,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
// Test that changes to recv codecs are applied to all streams.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
EXPECT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kTelephoneEventCodec1);
@ -1004,7 +1004,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].id = 106; // collide with existing CN 32k
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1017,7 +1017,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
// Test that we can apply the same set of codecs again while playing.
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1034,7 +1034,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
// Test that we can add a codec while playing.
TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1049,7 +1049,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847
TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1060,7 +1060,7 @@ TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
// Test that we do allow setting Opus/Red by default.
TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs[1].params[""] = "111/111";
@ -1114,7 +1114,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
EXPECT_TRUE(SetupChannel());
const int kDesiredBitrate = 128000;
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs = engine_->send_codecs();
parameters.max_bandwidth_bps = kDesiredBitrate;
SetSendParameters(parameters);
@ -1303,7 +1303,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
// Test that GetRtpSendParameters returns the currently configured codecs.
TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
SetSendParameters(parameters);
@ -1353,7 +1353,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
// Test that if we set/get parameters multiple times, we get the same results.
TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
SetSendParameters(parameters);
@ -1374,7 +1374,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
// SetRtpSendParameters is called.
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters send_parameters;
cricket::AudioSenderParameter send_parameters;
send_parameters.codecs.push_back(kOpusCodec);
SetSendParameters(send_parameters);
@ -1434,7 +1434,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) {
// Test that GetRtpReceiveParameters returns the currently configured codecs.
TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1458,7 +1458,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) {
// Test that if we set/get parameters multiple times, we get the same results.
TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1481,7 +1481,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) {
ASSERT_TRUE(SetupChannel());
// Call necessary methods to configure receiving a default stream as
// soon as it arrives.
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1512,7 +1512,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) {
TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
ASSERT_TRUE(SetupChannel());
cricket::AudioRecvParameters parameters = recv_parameters_;
cricket::AudioReceiverParameters parameters = recv_parameters_;
parameters.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, /*id=*/1));
ASSERT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -1538,7 +1538,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
// Test that we apply codecs properly.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn8000Codec);
@ -1558,7 +1558,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) {
// listed as the first codec and there is an fmtp line.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs[0].params[""] = "111/111";
parameters.codecs.push_back(kOpusCodec);
@ -1573,7 +1573,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) {
// listed as the first codec but there is no fmtp line.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs.push_back(kOpusCodec);
SetSendParameters(parameters);
@ -1586,7 +1586,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) {
// Test that we do not use Opus/Red by default.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs[1].params[""] = "111/111";
@ -1600,7 +1600,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) {
// Test that the RED fmtp line must match the payload type.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs[0].params[""] = "8/8";
parameters.codecs.push_back(kOpusCodec);
@ -1614,7 +1614,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) {
// Test that the RED fmtp line must show 2..32 payloads.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kRed48000Codec);
parameters.codecs[0].params[""] = "111";
parameters.codecs.push_back(kOpusCodec);
@ -1643,7 +1643,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) {
// AudioSendStream.
TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn8000Codec);
@ -1664,7 +1664,7 @@ TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
// Test that if clockrate is not 48000 for opus, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].clockrate = 50000;
@ -1674,7 +1674,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
// Test that if channels=0 for opus, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 0;
@ -1684,7 +1684,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) {
// Test that if channels=0 for opus, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 0;
@ -1695,7 +1695,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) {
// Test that if channel is 1 for opus and there's no stereo, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 1;
@ -1705,7 +1705,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) {
// Test that if channel is 1 for opus and stereo=0, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 1;
@ -1716,7 +1716,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) {
// Test that if channel is 1 for opus and stereo=1, we fail.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 1;
@ -1727,7 +1727,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) {
// Test that with bitrate=0 and no stereo, bitrate is 32000.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
SetSendParameters(parameters);
@ -1737,7 +1737,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) {
// Test that with bitrate=0 and stereo=0, bitrate is 32000.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].params["stereo"] = "0";
@ -1748,7 +1748,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) {
// Test that with bitrate=invalid and stereo=0, bitrate is 32000.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].params["stereo"] = "0";
// bitrate that's out of the range between 6000 and 510000 will be clamped.
@ -1764,7 +1764,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) {
// Test that with bitrate=0 and stereo=1, bitrate is 64000.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].params["stereo"] = "1";
@ -1775,7 +1775,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) {
// Test that with bitrate=invalid and stereo=1, bitrate is 64000.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].params["stereo"] = "1";
// bitrate that's out of the range between 6000 and 510000 will be clamped.
@ -1791,7 +1791,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) {
// Test that with bitrate=N and stereo unset, bitrate is N.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 96000;
SetSendParameters(parameters);
@ -1806,7 +1806,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
// Test that with bitrate=N and stereo=0, bitrate is N.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 30000;
parameters.codecs[0].params["stereo"] = "0";
@ -1817,7 +1817,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) {
// Test that with bitrate=N and without any parameters, bitrate is N.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 30000;
SetSendParameters(parameters);
@ -1827,7 +1827,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) {
// Test that with bitrate=N and stereo=1, bitrate is N.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].bitrate = 30000;
parameters.codecs[0].params["stereo"] = "1";
@ -1870,7 +1870,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
// Test that we can enable NACK with opus as callee.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@ -1887,7 +1887,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
EXPECT_TRUE(SetupSendStream());
EXPECT_TRUE(AddRecvStream(kSsrcY));
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@ -1900,7 +1900,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
EXPECT_TRUE(SetupSendStream());
EXPECT_TRUE(AddRecvStream(kSsrcY));
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@ -1916,7 +1916,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
// Test that NACK is enabled on a new receive stream.
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
@ -1933,7 +1933,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters opus_parameters;
cricket::AudioSenderParameter opus_parameters;
opus_parameters.codecs.push_back(kOpusCodec);
SetSendParameters(opus_parameters);
{
@ -1942,7 +1942,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) {
EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
}
cricket::AudioSendParameters pcmu_parameters;
cricket::AudioSenderParameter pcmu_parameters;
pcmu_parameters.codecs.push_back(kPcmuCodec);
pcmu_parameters.codecs.push_back(kCn16000Codec);
pcmu_parameters.codecs.push_back(kOpusCodec);
@ -1964,7 +1964,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) {
// Test that we handle various ways of specifying bitrate.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kPcmuCodec);
SetSendParameters(parameters);
{
@ -1997,7 +1997,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
// Test that we fail if no codecs are specified.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
EXPECT_FALSE(send_channel_->SetSendParameters(parameters));
}
@ -2005,7 +2005,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
// one on the list.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
@ -2022,7 +2022,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
// Test that CanInsertDtmf() is governed by the send flag
TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kTelephoneEventCodec1);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs[0].id = 98; // DTMF
@ -2038,7 +2038,7 @@ TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) {
// Test that payload type range is limited for telephone-event codec.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kTelephoneEventCodec2);
parameters.codecs.push_back(kOpusCodec);
parameters.codecs[0].id = 0; // DTMF
@ -2061,7 +2061,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
// one on the list.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kCn8000Codec);
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs[0].id = 98; // narrowband CN
@ -2075,7 +2075,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
// Test that we set VAD and DTMF types correctly as caller.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
@ -2096,7 +2096,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
// Test that we set VAD and DTMF types correctly as callee.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
EXPECT_TRUE(SetupChannel());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
@ -2121,7 +2121,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
// send codec clockrate.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
// Set PCMU(8K) and CN(16K). VAD should not be activated.
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
@ -2154,7 +2154,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
// Test that we perform case-insensitive matching of codec names.
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn16000Codec);
parameters.codecs.push_back(kCn8000Codec);
@ -2306,7 +2306,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
cricket::StreamParams::CreateLegacy(ssrc)));
}
cricket::AudioSendParameters parameters;
cricket::AudioSenderParameter parameters;
// Set PCMU and CN(8K). VAD should be activated.
parameters.codecs.push_back(kPcmuCodec);
parameters.codecs.push_back(kCn8000Codec);
@ -2893,7 +2893,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStream) {
// those previously passed into SetRecvCodecs.
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs.push_back(kOpusCodec);
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(receive_channel_->SetRecvParameters(parameters));
@ -3120,7 +3120,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
send_channel2->AddSendStream(stream2);
// AEC and AGC and NS
cricket::AudioSendParameters parameters_options_all = send_parameters_;
cricket::AudioSenderParameter parameters_options_all = send_parameters_;
parameters_options_all.options.echo_cancellation = true;
parameters_options_all.options.auto_gain_control = true;
parameters_options_all.options.noise_suppression = true;
@ -3140,7 +3140,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
}
// unset NS
cricket::AudioSendParameters parameters_options_no_ns = send_parameters_;
cricket::AudioSenderParameter parameters_options_no_ns = send_parameters_;
parameters_options_no_ns.options.noise_suppression = false;
EXPECT_TRUE(send_channel1->SetSendParameters(parameters_options_no_ns));
cricket::AudioOptions expected_options = parameters_options_all.options;
@ -3157,7 +3157,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
}
// unset AGC
cricket::AudioSendParameters parameters_options_no_agc = send_parameters_;
cricket::AudioSenderParameter parameters_options_no_agc = send_parameters_;
parameters_options_no_agc.options.auto_gain_control = false;
EXPECT_TRUE(send_channel2->SetSendParameters(parameters_options_no_agc));
if (!use_null_apm_) {
@ -3197,7 +3197,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
}
// Make sure settings take effect while we are sending.
cricket::AudioSendParameters parameters_options_no_agc_nor_ns =
cricket::AudioSenderParameter parameters_options_no_agc_nor_ns =
send_parameters_;
parameters_options_no_agc_nor_ns.options.auto_gain_control = false;
parameters_options_no_agc_nor_ns.options.noise_suppression = false;
@ -3441,7 +3441,7 @@ TEST_P(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
// Set up receive extensions.
const std::vector<webrtc::RtpExtension> header_extensions =
GetDefaultEnabledRtpHeaderExtensions(*engine_);
cricket::AudioRecvParameters recv_parameters;
cricket::AudioReceiverParameters recv_parameters;
recv_parameters.extensions = header_extensions;
receive_channel_->SetRecvParameters(recv_parameters);
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
@ -3452,7 +3452,7 @@ TEST_P(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
}
// Disable receive extensions.
receive_channel_->SetRecvParameters(cricket::AudioRecvParameters());
receive_channel_->SetRecvParameters(cricket::AudioReceiverParameters());
for (uint32_t ssrc : ssrcs) {
EXPECT_THAT(
receive_channel_->GetRtpReceiveParameters(ssrc).header_extensions,
@ -3630,7 +3630,7 @@ TEST_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
// Changing RTP header extensions will recreate the
// AudioReceiveStreamInterface.
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
receive_channel_->SetRecvParameters(parameters);
@ -3849,7 +3849,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
&engine, cricket::MediaConfig(), cricket::AudioOptions(),
webrtc::CryptoOptions(), call.get(),
webrtc::AudioCodecPairId::Create());
cricket::AudioRecvParameters parameters;
cricket::AudioReceiverParameters parameters;
parameters.codecs = engine.recv_codecs();
EXPECT_TRUE(channel.SetRecvParameters(parameters));
}
@ -3885,7 +3885,7 @@ TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {
&engine, cricket::MediaConfig(), cricket::AudioOptions(),
webrtc::CryptoOptions(), call.get(), webrtc::AudioCodecPairId::Create());
{
cricket::AudioSendParameters params;
cricket::AudioSenderParameter params;
params.codecs.push_back(cricket::CreateAudioCodec(1, "opus", 48000, 2));
params.extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri, 1));

View File

@ -78,11 +78,11 @@ struct StreamFinder {
} // namespace
template <class Codec>
void RtpParametersFromMediaDescription(
void MediaChannelParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
bool is_stream_active,
RtpParameters* params) {
MediaChannelParameters* params) {
params->is_stream_active = is_stream_active;
params->codecs = desc->codecs();
// TODO(bugs.webrtc.org/11513): See if we really need
@ -98,14 +98,14 @@ template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
webrtc::RtpExtension::Filter extensions_filter,
RtpSendParameters* send_params) {
SenderParameters* send_params) {
RtpHeaderExtensions extensions =
webrtc::RtpExtension::DeduplicateHeaderExtensions(
desc->rtp_header_extensions(), extensions_filter);
const bool is_stream_active =
webrtc::RtpTransceiverDirectionHasRecv(desc->direction());
RtpParametersFromMediaDescription(desc, extensions, is_stream_active,
send_params);
MediaChannelParametersFromMediaDescription(desc, extensions, is_stream_active,
send_params);
send_params->max_bandwidth_bps = desc->bandwidth();
send_params->extmap_allow_mixed = desc->extmap_allow_mixed();
}
@ -882,8 +882,8 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
bool update_header_extensions = true;
media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
AudioReceiverParameters recv_params = last_recv_params_;
MediaChannelParametersFromMediaDescription(
content->as_audio(), header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&recv_params);
@ -935,7 +935,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
AudioSendParameters send_params = last_send_params_;
AudioSenderParameter send_params = last_send_params_;
RtpSendParametersFromMediaDescription(content->as_audio(),
extensions_filter(), &send_params);
send_params.mid = mid();
@ -1022,14 +1022,14 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
bool update_header_extensions = true;
media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
VideoRecvParameters recv_params = last_recv_params_;
VideoReceiverParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
MediaChannelParametersFromMediaDescription(
content->as_video(), header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&recv_params);
VideoSendParameters send_params = last_send_params_;
VideoSenderParameters send_params = last_send_params_;
bool needs_send_params_update = false;
if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
@ -1108,13 +1108,13 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
const VideoContentDescription* video = content->as_video();
VideoSendParameters send_params = last_send_params_;
VideoSenderParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(video, extensions_filter(),
&send_params);
send_params.mid = mid();
send_params.conference_mode = video->conference_mode();
VideoRecvParameters recv_params = last_recv_params_;
VideoReceiverParameters recv_params = last_recv_params_;
bool needs_recv_params_update = false;
if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {

View File

@ -429,12 +429,12 @@ class VoiceChannel : public BaseChannel {
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
// Last AudioSendParameters sent down to the media_channel() via
// Last AudioSenderParameter sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last AudioRecvParameters sent down to the media_channel() via
AudioSenderParameter last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last AudioReceiverParameters sent down to the media_channel() via
// SetRecvParameters.
AudioRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
AudioReceiverParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
};
// VideoChannel is a specialization for video.
@ -498,12 +498,12 @@ class VideoChannel : public BaseChannel {
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
// Last VideoSendParameters sent down to the media_channel() via
// Last VideoSenderParameters sent down to the media_channel() via
// SetSendParameters.
VideoSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last VideoRecvParameters sent down to the media_channel() via
VideoSenderParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last VideoReceiverParameters sent down to the media_channel() via
// SetRecvParameters.
VideoRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
VideoReceiverParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
};
} // namespace cricket

View File

@ -172,7 +172,7 @@ class RtpSenderReceiverTest
// Needed to use DTMF sender.
void AddDtmfCodec() {
cricket::AudioSendParameters params;
cricket::AudioSenderParameter params;
const cricket::AudioCodec kTelephoneEventCodec =
cricket::CreateAudioCodec(106, "telephone-event", 8000, 1);
params.codecs.push_back(kTelephoneEventCodec);

View File

@ -35,7 +35,7 @@ class MockVoiceMediaReceiveChannelInterface
// VoiceMediaReceiveChannelInterface
MOCK_METHOD(bool,
SetRecvParameters,
(const AudioRecvParameters& params),
(const AudioReceiverParameters& params),
(override));
MOCK_METHOD(webrtc::RtpParameters,
GetRtpReceiveParameters,