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Clean up logging in AudioSendStream::SetupSendCodec().

Browse Source 
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.

BUG=webrtc:4690

Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Original-Commit-Position: refs/heads/master@{#14771}
Cr-Commit-Position: refs/heads/master@{#14780}
This commit is contained in:
solenberg
2016-10-25 11:19:07 -07:00
committed by Commit bot
parent da389e3518
commit 940b6d648f
10 changed files with 185 additions and 128 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/api/BUILD.gn
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@ -21,6 +21,7 @@ group("api") {
rtc_source_set("call_api") {
sources = [
"call/audio_receive_stream.h",
"call/audio_send_stream.cc",
"call/audio_send_stream.h",
"call/audio_sink.h",
"call/audio_state.h",

1
webrtc/api/api.gyp
View File

@ -105,6 +105,7 @@
],
'sources': [
'call/audio_receive_stream.h',
'call/audio_send_stream.cc',
'call/audio_send_stream.h',
'call/audio_sink.h',
'call/audio_state.h',

118
webrtc/api/call/audio_send_stream.cc Normal file
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@ -0,0 +1,118 @@
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/call/audio_send_stream.h"
#include <string>
namespace {
std::string ToString(const webrtc::CodecInst& codec_inst) {
std::stringstream ss;
ss << "{pltype: " << codec_inst.pltype;
ss << ", plname: \"" << codec_inst.plname << "\"";
ss << ", plfreq: " << codec_inst.plfreq;
ss << ", pacsize: " << codec_inst.pacsize;
ss << ", channels: " << codec_inst.channels;
ss << ", rate: " << codec_inst.rate;
ss << '}';
return ss.str();
}
} // namespace
namespace webrtc {
AudioSendStream::Stats::Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
ss << ", voe_channel_id: " << voe_channel_id;
ss << ", min_bitrate_kbps: " << min_bitrate_kbps;
ss << ", max_bitrate_kbps: " << max_bitrate_kbps;
ss << ", send_codec_spec: " << send_codec_spec.ToString();
ss << '}';
return ss.str();
}
AudioSendStream::Config::Rtp::Rtp() = default;
AudioSendStream::Config::Rtp::~Rtp() = default;
std::string AudioSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrc: " << ssrc;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << ", nack: " << nack.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
AudioSendStream::Config::SendCodecSpec::SendCodecSpec() {
webrtc::CodecInst empty_inst = {0};
codec_inst = empty_inst;
codec_inst.pltype = -1;
}
std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
std::stringstream ss;
ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
ss << ", cng_payload_type: " << cng_payload_type;
ss << ", cng_plfreq: " << cng_plfreq;
ss << ", codec_inst: " << ::ToString(codec_inst);
ss << '}';
return ss.str();
}
bool AudioSendStream::Config::SendCodecSpec::operator==(
const AudioSendStream::Config::SendCodecSpec& rhs) const {
if (nack_enabled != rhs.nack_enabled) {
return false;
}
if (transport_cc_enabled != rhs.transport_cc_enabled) {
return false;
}
if (enable_codec_fec != rhs.enable_codec_fec) {
return false;
}
if (enable_opus_dtx != rhs.enable_opus_dtx) {
return false;
}
if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
return false;
}
if (cng_payload_type != rhs.cng_payload_type) {
return false;
}
if (cng_plfreq != rhs.cng_plfreq) {
return false;
}
if (codec_inst != rhs.codec_inst) {
return false;
}
return true;
}
} // namespace webrtc

46
webrtc/api/call/audio_send_stream.h
View File

@ -30,6 +30,8 @@ namespace webrtc {
class AudioSendStream {
public:
struct Stats {
Stats();
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
int64_t bytes_sent = 0;
@ -52,13 +54,13 @@ class AudioSendStream {
struct Config {
Config() = delete;
explicit Config(Transport* send_transport)
: send_transport(send_transport) {}
explicit Config(Transport* send_transport);
std::string ToString() const;
// Send-stream specific RTP settings.
struct Rtp {
Rtp();
~Rtp();
std::string ToString() const;
// Sender SSRC.
@ -91,40 +93,10 @@ class AudioSendStream {
int max_bitrate_kbps = -1;
struct SendCodecSpec {
SendCodecSpec() {
webrtc::CodecInst empty_inst = {0};
codec_inst = empty_inst;
codec_inst.pltype = -1;
}
bool operator==(const SendCodecSpec& rhs) const {
{
if (nack_enabled != rhs.nack_enabled) {
return false;
}
if (transport_cc_enabled != rhs.transport_cc_enabled) {
return false;
}
if (enable_codec_fec != rhs.enable_codec_fec) {
return false;
}
if (enable_opus_dtx != rhs.enable_opus_dtx) {
return false;
}
if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
return false;
}
if (cng_payload_type != rhs.cng_payload_type) {
return false;
}
if (cng_plfreq != rhs.cng_plfreq) {
return false;
}
if (codec_inst != rhs.codec_inst) {
return false;
}
return true;
}
}
SendCodecSpec();
std::string ToString() const;
bool operator==(const SendCodecSpec& rhs) const;
bool operator!=(const SendCodecSpec& rhs) const {
return !(*this == rhs);
}

99
webrtc/audio/audio_send_stream.cc
View File

@ -35,56 +35,11 @@ namespace {
constexpr char kOpusCodecName[] = "opus";
// TODO(minyue): Remove |LOG_RTCERR2|.
#define LOG_RTCERR2(func, a1, a2, err) \
LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 \
<< ") failed, err=" << err
// TODO(minyue): Remove |LOG_RTCERR3|.
#define LOG_RTCERR3(func, a1, a2, a3, err) \
LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
<< ") failed, err=" << err
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 webrtc::CodecInst& codec, const char* ref_name) {
return (_stricmp(codec.plname, ref_name) == 0);
}
} // namespace
std::string AudioSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrc: " << ssrc;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << ", nack: " << nack.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", voe_channel_id: " << voe_channel_id;
// TODO(solenberg): Encoder config.
ss << ", cng_payload_type: " << send_codec_spec.cng_payload_type;
ss << '}';
return ss.str();
}
namespace internal {
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
@ -333,11 +288,8 @@ bool AudioSendStream::SetupSendCodec() {
const auto& send_codec_spec = config_.send_codec_spec;
// Set the codec immediately, since SetVADStatus() depends on whether
// the current codec is mono or stereo.
LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
<< ToString(send_codec_spec.codec_inst)
<< ", bitrate=" << send_codec_spec.codec_inst.rate;
// We set the codec first, since the below extra configuration is only applied
// to the "current" codec.
// If codec is already configured, we do not it again.
// TODO(minyue): check if this check is really needed, or can we move it into
@ -346,47 +298,33 @@ bool AudioSendStream::SetupSendCodec() {
if (codec->GetSendCodec(channel, current_codec) != 0 ||
(send_codec_spec.codec_inst != current_codec)) {
if (codec->SetSendCodec(channel, send_codec_spec.codec_inst) == -1) {
LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec_spec.codec_inst),
base->LastError());
LOG(LS_WARNING) << "SetSendCodec() failed: " << base->LastError();
return false;
}
}
// FEC should be enabled after SetSendCodec.
// Codec internal FEC. Treat any failure as fatal internal error.
if (send_codec_spec.enable_codec_fec) {
LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
<< channel;
if (codec->SetFECStatus(channel, true) == -1) {
// Enable codec internal FEC. Treat any failure as fatal internal error.
LOG_RTCERR2(SetFECStatus, channel, true, base->LastError());
if (codec->SetFECStatus(channel, true) != 0) {
LOG(LS_WARNING) << "SetFECStatus() failed: " << base->LastError();
return false;
}
}
// DTX and maxplaybackrate are only set if current codec is Opus.
if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) {
// DTX and maxplaybackrate should be set after SetSendCodec. Because current
// send codec has to be Opus.
// Set Opus internal DTX.
LOG(LS_INFO) << "Attempt to "
<< (send_codec_spec.enable_opus_dtx ? "enable" : "disable")
<< " Opus DTX on channel " << channel;
if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx)) {
LOG_RTCERR2(SetOpusDtx, channel, send_codec_spec.enable_opus_dtx,
base->LastError());
if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx) != 0) {
LOG(LS_WARNING) << "SetOpusDtx() failed: " << base->LastError();
return false;
}
// If opus_max_playback_rate <= 0, the default maximum playback rate
// (48 kHz) will be used.
if (send_codec_spec.opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
<< send_codec_spec.opus_max_playback_rate
<< " Hz on channel " << channel;
if (codec->SetOpusMaxPlaybackRate(
channel, send_codec_spec.opus_max_playback_rate) == -1) {
LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
send_codec_spec.opus_max_playback_rate, base->LastError());
channel, send_codec_spec.opus_max_playback_rate) != 0) {
LOG(LS_WARNING) << "SetOpusMaxPlaybackRate() failed: "
<< base->LastError();
return false;
}
}
@ -409,11 +347,9 @@ bool AudioSendStream::SetupSendCodec() {
return false;
}
if (codec->SetSendCNPayloadType(channel, send_codec_spec.cng_payload_type,
cn_freq) == -1) {
LOG_RTCERR3(SetSendCNPayloadType, channel,
send_codec_spec.cng_payload_type, cn_freq,
base->LastError());
cn_freq) != 0) {
LOG(LS_WARNING) << "SetSendCNPayloadType() failed: "
<< base->LastError();
// TODO(ajm): This failure condition will be removed from VoE.
// Restore the return here when we update to a new enough webrtc.
//
@ -431,9 +367,8 @@ bool AudioSendStream::SetupSendCodec() {
send_codec_spec.codec_inst.channels == 1) {
// TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
// interaction between VAD and Opus FEC.
LOG(LS_INFO) << "Enabling VAD";
if (codec->SetVADStatus(channel, true) == -1) {
LOG_RTCERR2(SetVADStatus, channel, true, base->LastError());
if (codec->SetVADStatus(channel, true) != 0) {
LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
return false;
}
}

18
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -219,12 +219,26 @@ TEST(AudioSendStreamTest, ConfigToString) {
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
config.rtp.c_name = kCName;
config.voe_channel_id = kChannelId;
config.min_bitrate_kbps = 12;
config.max_bitrate_kbps = 34;
config.send_codec_spec.nack_enabled = true;
config.send_codec_spec.transport_cc_enabled = false;
config.send_codec_spec.enable_codec_fec = true;
config.send_codec_spec.enable_opus_dtx = false;
config.send_codec_spec.opus_max_playback_rate = 32000;
config.send_codec_spec.cng_payload_type = 42;
config.send_codec_spec.cng_plfreq = 56;
config.send_codec_spec.codec_inst = kIsacCodec;
EXPECT_EQ(
"{rtp: {ssrc: 1234, extensions: [{uri: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
"nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, "
"cng_payload_type: 42}",
"nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
"voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
"32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: "
"103, plname: \"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: "
"32000}}}",
config.ToString());
}

13
webrtc/config.cc
View File

@ -31,6 +31,11 @@ std::string UlpfecConfig::ToString() const {
return ss.str();
}
FlexfecConfig::FlexfecConfig()
: flexfec_payload_type(-1), flexfec_ssrc(0), protected_media_ssrcs() {}
FlexfecConfig::~FlexfecConfig() = default;
std::string FlexfecConfig::ToString() const {
std::stringstream ss;
ss << "{flexfec_payload_type: " << flexfec_payload_type;
@ -134,6 +139,8 @@ VideoEncoderConfig::VideoEncoderConfig()
max_bitrate_bps(0),
number_of_streams(0) {}
VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
VideoEncoderConfig::~VideoEncoderConfig() = default;
std::string VideoEncoderConfig::ToString() const {
@ -155,6 +162,8 @@ std::string VideoEncoderConfig::ToString() const {
return ss.str();
}
VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
VideoCodec* codec) const {
if (codec->codecType == kVideoCodecH264) {
@ -210,4 +219,8 @@ void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
*vp9_settings = specifics_;
}
DecoderSpecificSettings::DecoderSpecificSettings() = default;
DecoderSpecificSettings::~DecoderSpecificSettings() = default;
} // namespace webrtc

15
webrtc/config.h
View File

@ -57,8 +57,8 @@ struct UlpfecConfig {
// Settings for FlexFEC forward error correction.
// Set the payload type to '-1' to disable.
struct FlexfecConfig {
FlexfecConfig()
: flexfec_payload_type(-1), flexfec_ssrc(0), protected_media_ssrcs() {}
FlexfecConfig();
~FlexfecConfig();
std::string ToString() const;
// Payload type of FlexFEC.
@ -163,7 +163,7 @@ class VideoEncoderConfig {
virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
private:
virtual ~EncoderSpecificSettings() {}
~EncoderSpecificSettings() override {}
friend class VideoEncoderConfig;
};
@ -211,7 +211,7 @@ class VideoEncoderConfig {
const VideoEncoderConfig& encoder_config) = 0;
protected:
virtual ~VideoStreamFactoryInterface() {}
~VideoStreamFactoryInterface() override {}
};
VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
@ -221,7 +221,7 @@ class VideoEncoderConfig {
VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
VideoEncoderConfig();
VideoEncoderConfig(VideoEncoderConfig&&) = default;
VideoEncoderConfig(VideoEncoderConfig&&);
~VideoEncoderConfig();
std::string ToString() const;
@ -243,7 +243,7 @@ class VideoEncoderConfig {
private:
// Access to the copy constructor is private to force use of the Copy()
// method for those exceptional cases where we do use it.
VideoEncoderConfig(const VideoEncoderConfig&) = default;
VideoEncoderConfig(const VideoEncoderConfig&);
};
struct VideoDecoderH264Settings {
@ -252,7 +252,8 @@ struct VideoDecoderH264Settings {
class DecoderSpecificSettings {
public:
virtual ~DecoderSpecificSettings() {}
DecoderSpecificSettings();
virtual ~DecoderSpecificSettings();
rtc::Optional<VideoDecoderH264Settings> h264_extra_settings;
};

1
webrtc/logging/BUILD.gn
View File

@ -32,6 +32,7 @@ rtc_static_library("rtc_event_log_impl") {
deps = [
":rtc_event_log_api",
"..:webrtc_common",
"../api:call_api",
"../modules/rtp_rtcp",
]

1
webrtc/webrtc.gyp
View File

@ -67,6 +67,7 @@
'dependencies': [
'rtc_event_log_api',
'rtc_event_log_proto',
'<(webrtc_root)/api/api.gyp:call_api',
],
'defines': [
'ENABLE_RTC_EVENT_LOG',