Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1386653002 Cr-Commit-Position: refs/heads/master@{#10194}
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@ -51,10 +51,6 @@
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#include "webrtc/video_decoder.h"
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#include "webrtc/video_encoder.h"
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#define UNIMPLEMENTED \
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LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
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RTC_NOTREACHED()
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namespace cricket {
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namespace {
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@ -908,10 +908,6 @@ class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test {
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}
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protected:
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virtual std::vector<cricket::VideoCodec> GetCodecs() {
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return engine_.codecs();
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}
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FakeVideoSendStream* AddSendStream() {
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return AddSendStream(StreamParams::CreateLegacy(++last_ssrc_));
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}
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@ -54,8 +54,9 @@
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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namespace cricket {
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namespace {
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static const int kMaxNumPacketSize = 6;
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const int kMaxNumPacketSize = 6;
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struct CodecPref {
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const char* name;
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int clockrate;
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@ -65,7 +66,7 @@ struct CodecPref {
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int packet_sizes_ms[kMaxNumPacketSize];
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};
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// Note: keep the supported packet sizes in ascending order.
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static const CodecPref kCodecPrefs[] = {
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const CodecPref kCodecPrefs[] = {
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{ kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
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{ kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
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{ kIsacCodecName, 32000, 1, 104, true, { 30 } },
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@ -97,14 +98,14 @@ static const CodecPref kCodecPrefs[] = {
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// It's not clear yet whether the -2 index is handled properly on other OSes.
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#ifdef WIN32
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static const int kDefaultAudioDeviceId = -1;
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const int kDefaultAudioDeviceId = -1;
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#else
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static const int kDefaultAudioDeviceId = 0;
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const int kDefaultAudioDeviceId = 0;
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#endif
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// Parameter used for NACK.
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// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
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static const int kNackMaxPackets = 250;
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const int kNackMaxPackets = 250;
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// Codec parameters for Opus.
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// draft-spittka-payload-rtp-opus-03
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@ -117,18 +118,18 @@ static const int kNackMaxPackets = 250;
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// 64-128 kb/s for FB stereo music.
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// The current implementation applies the following values to mono signals,
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// and multiplies them by 2 for stereo.
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static const int kOpusBitrateNb = 12000;
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static const int kOpusBitrateWb = 20000;
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static const int kOpusBitrateFb = 32000;
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const int kOpusBitrateNb = 12000;
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const int kOpusBitrateWb = 20000;
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const int kOpusBitrateFb = 32000;
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// Opus bitrate should be in the range between 6000 and 510000.
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static const int kOpusMinBitrate = 6000;
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static const int kOpusMaxBitrate = 510000;
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const int kOpusMinBitrate = 6000;
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const int kOpusMaxBitrate = 510000;
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// Default audio dscp value.
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// See http://tools.ietf.org/html/rfc2474 for details.
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// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
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static const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
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const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
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// Ensure we open the file in a writeable path on ChromeOS and Android. This
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// workaround can be removed when it's possible to specify a filename for audio
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@ -140,29 +141,29 @@ static const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
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// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
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// below.
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#if defined(CHROMEOS)
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static const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
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const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
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#elif defined(ANDROID)
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static const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
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const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
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#else
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static const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
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const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
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#endif
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// Dumps an AudioCodec in RFC 2327-ish format.
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static std::string ToString(const AudioCodec& codec) {
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std::string ToString(const AudioCodec& codec) {
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std::stringstream ss;
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ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
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<< " (" << codec.id << ")";
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return ss.str();
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}
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static std::string ToString(const webrtc::CodecInst& codec) {
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std::string ToString(const webrtc::CodecInst& codec) {
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std::stringstream ss;
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ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
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<< " (" << codec.pltype << ")";
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return ss.str();
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}
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static void LogMultiline(rtc::LoggingSeverity sev, char* text) {
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void LogMultiline(rtc::LoggingSeverity sev, char* text) {
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const char* delim = "\r\n";
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for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
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LOG_V(sev) << tok;
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@ -170,7 +171,7 @@ static void LogMultiline(rtc::LoggingSeverity sev, char* text) {
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}
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// Severity is an integer because it comes is assumed to be from command line.
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static int SeverityToFilter(int severity) {
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int SeverityToFilter(int severity) {
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int filter = webrtc::kTraceNone;
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switch (severity) {
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case rtc::LS_VERBOSE:
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@ -188,15 +189,15 @@ static int SeverityToFilter(int severity) {
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return filter;
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}
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static bool IsCodec(const AudioCodec& codec, const char* ref_name) {
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bool IsCodec(const AudioCodec& codec, const char* ref_name) {
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return (_stricmp(codec.name.c_str(), ref_name) == 0);
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}
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static bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
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bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
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return (_stricmp(codec.plname, ref_name) == 0);
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}
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static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
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if (IsCodec(codec, kCodecPrefs[i].name) &&
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kCodecPrefs[i].clockrate == codec.plfreq) {
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@ -206,7 +207,7 @@ static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
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return false;
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}
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static bool FindCodec(const std::vector<AudioCodec>& codecs,
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bool FindCodec(const std::vector<AudioCodec>& codecs,
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const AudioCodec& codec,
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AudioCodec* found_codec) {
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for (const AudioCodec& c : codecs) {
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@ -220,12 +221,12 @@ static bool FindCodec(const std::vector<AudioCodec>& codecs,
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return false;
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}
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static bool IsNackEnabled(const AudioCodec& codec) {
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bool IsNackEnabled(const AudioCodec& codec) {
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return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
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kParamValueEmpty));
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}
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static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
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for (int packet_size_ms : codec_pref.packet_sizes_ms) {
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if (packet_size_ms && packet_size_ms <= ptime_ms) {
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@ -238,7 +239,7 @@ static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
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// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec
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// pacsize if it's valid, or we will pick the next smallest value we support.
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// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
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static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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for (const CodecPref& codec_pref : kCodecPrefs) {
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if ((IsCodec(*codec, codec_pref.name) &&
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codec_pref.clockrate == codec->plfreq) ||
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@ -255,7 +256,7 @@ static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
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}
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// Return true if codec.params[feature] == "1", false otherwise.
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static bool IsCodecFeatureEnabled(const AudioCodec& codec,
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bool IsCodecFeatureEnabled(const AudioCodec& codec,
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const char* feature) {
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int value;
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return codec.GetParam(feature, &value) && value == 1;
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@ -265,7 +266,7 @@ static bool IsCodecFeatureEnabled(const AudioCodec& codec,
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// otherwise. If the value (either from params or codec.bitrate) <=0, use the
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// default configuration. If the value is beyond feasible bit rate of Opus,
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// clamp it. Returns the Opus bit rate for operation.
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static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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int bitrate = 0;
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bool use_param = true;
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if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
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@ -298,7 +299,7 @@ static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
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// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
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// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
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static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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int value;
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if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
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return value;
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@ -306,7 +307,7 @@ static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
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return kOpusDefaultMaxPlaybackRate;
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}
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static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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bool* enable_codec_fec, int* max_playback_rate,
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bool* enable_codec_dtx) {
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*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
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@ -326,7 +327,7 @@ static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
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// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
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// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
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// codec.
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static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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if (IsCodec(*voe_codec, kG722CodecName)) {
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// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
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// has changed, and this special case is no longer needed.
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@ -338,7 +339,7 @@ static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
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// Gets the default set of options applied to the engine. Historically, these
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// were supplied as a combination of flags from the channel manager (ec, agc,
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// ns, and highpass) and the rest hardcoded in InitInternal.
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static AudioOptions GetDefaultEngineOptions() {
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AudioOptions GetDefaultEngineOptions() {
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AudioOptions options;
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options.echo_cancellation.Set(true);
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options.auto_gain_control.Set(true);
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@ -358,9 +359,10 @@ static AudioOptions GetDefaultEngineOptions() {
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return options;
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}
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static std::string GetEnableString(bool enable) {
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std::string GetEnableString(bool enable) {
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return enable ? "enable" : "disable";
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}
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} // namespace {
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WebRtcVoiceEngine::WebRtcVoiceEngine()
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: voe_wrapper_(new VoEWrapper()),
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@ -862,18 +864,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
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return true;
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}
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struct ResumeEntry {
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ResumeEntry(WebRtcVoiceMediaChannel *c, bool p, SendFlags s)
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: channel(c),
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playout(p),
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send(s) {
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}
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WebRtcVoiceMediaChannel *channel;
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bool playout;
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SendFlags send;
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};
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// TODO(juberti): Refactor this so that the core logic can be used to set the
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// soundclip device. At that time, reinstate the soundclip pause/resume code.
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bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
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@ -1186,40 +1176,18 @@ void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
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}
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}
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void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) {
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rtc::CritScope lock(&channels_cs_);
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WebRtcVoiceMediaChannel* channel = NULL;
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uint32 ssrc = 0;
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void WebRtcVoiceEngine::CallbackOnError(int channel_id, int err_code) {
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RTC_DCHECK(channel_id == -1);
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LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
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<< channel_num << ".";
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if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
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RTC_DCHECK(channel != NULL);
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channel->OnError(ssrc, err_code);
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} else {
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LOG(LS_ERROR) << "VoiceEngine channel " << channel_num
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<< " could not be found in channel list when error reported.";
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<< channel_id << ".";
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rtc::CritScope lock(&channels_cs_);
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for (WebRtcVoiceMediaChannel* channel : channels_) {
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channel->OnError(err_code);
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}
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}
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bool WebRtcVoiceEngine::FindChannelAndSsrc(
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int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
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RTC_DCHECK(channel != NULL && ssrc != NULL);
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*channel = NULL;
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*ssrc = 0;
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// Find corresponding channel and ssrc
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for (WebRtcVoiceMediaChannel* ch : channels_) {
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RTC_DCHECK(ch != NULL);
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if (ch->FindSsrc(channel_num, ssrc)) {
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*channel = ch;
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return true;
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}
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}
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return false;
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}
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void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
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RTC_DCHECK(channel != NULL);
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rtc::CritScope lock(&channels_cs_);
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channels_.push_back(channel);
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}
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@ -1416,6 +1384,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
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send_(SEND_NOTHING),
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call_(call),
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default_receive_ssrc_(0) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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engine->RegisterChannel(this);
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LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
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<< voe_channel();
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@ -1425,16 +1394,19 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
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}
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WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
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<< voe_channel();
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// Remove any remaining send streams, the default channel will be deleted
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// later.
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while (!send_channels_.empty())
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while (!send_channels_.empty()) {
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RemoveSendStream(send_channels_.begin()->first);
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}
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// Unregister ourselves from the engine.
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engine()->UnregisterChannel(this);
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// Remove any remaining streams.
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while (!receive_channels_.empty()) {
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RemoveRecvStream(receive_channels_.begin()->first);
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@ -1447,6 +1419,7 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
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bool WebRtcVoiceMediaChannel::SetSendParameters(
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const AudioSendParameters& params) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(pthatcher): Refactor this to be more clean now that we have
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// all the information at once.
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return (SetSendCodecs(params.codecs) &&
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@ -1457,6 +1430,7 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
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bool WebRtcVoiceMediaChannel::SetRecvParameters(
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const AudioRecvParameters& params) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(pthatcher): Refactor this to be more clean now that we have
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// all the information at once.
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return (SetRecvCodecs(params.codecs) &&
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@ -1464,6 +1438,7 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
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}
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bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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LOG(LS_INFO) << "Setting voice channel options: "
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<< options.ToString();
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@ -1553,6 +1528,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
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bool WebRtcVoiceMediaChannel::SetRecvCodecs(
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const std::vector<AudioCodec>& codecs) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// Set the payload types to be used for incoming media.
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LOG(LS_INFO) << "Setting receive voice codecs:";
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@ -1804,6 +1780,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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bool WebRtcVoiceMediaChannel::SetSendCodecs(
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const std::vector<AudioCodec>& codecs) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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dtmf_allowed_ = false;
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for (const AudioCodec& codec : codecs) {
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// Find the DTMF telephone event "codec".
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@ -1875,6 +1853,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodec(
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bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (receive_extensions_ == extensions) {
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return true;
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}
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@ -1942,6 +1921,7 @@ bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
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|
||||
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
|
||||
const std::vector<RtpHeaderExtension>& extensions) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
if (send_extensions_ == extensions) {
|
||||
return true;
|
||||
}
|
||||
@ -2000,6 +1980,7 @@ bool WebRtcVoiceMediaChannel::ResumePlayout() {
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
if (playout_ == playout) {
|
||||
return true;
|
||||
}
|
||||
@ -2088,6 +2069,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
|
||||
bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool enable,
|
||||
const AudioOptions* options,
|
||||
AudioRenderer* renderer) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
// TODO(solenberg): The state change should be fully rolled back if any one of
|
||||
// these calls fail.
|
||||
if (!SetLocalRenderer(ssrc, renderer)) {
|
||||
@ -2133,9 +2115,10 @@ bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
// If the default channel is already used for sending create a new channel
|
||||
// otherwise use the default channel for sending.
|
||||
int channel = GetSendChannelNum(sp.first_ssrc());
|
||||
int channel = GetSendChannelId(sp.first_ssrc());
|
||||
if (channel != -1) {
|
||||
LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc();
|
||||
return false;
|
||||
@ -2243,6 +2226,8 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
|
||||
|
||||
rtc::CritScope lock(&receive_channels_cs_);
|
||||
|
||||
if (!VERIFY(sp.ssrcs.size() == 1))
|
||||
@ -2301,6 +2286,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
// Configure to use external transport, like our default channel.
|
||||
if (engine()->voe()->network()->RegisterExternalTransport(
|
||||
channel, *this) == -1) {
|
||||
@ -2373,6 +2359,8 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
|
||||
|
||||
rtc::CritScope lock(&receive_channels_cs_);
|
||||
ChannelMap::iterator it = receive_channels_.find(ssrc);
|
||||
if (it == receive_channels_.end()) {
|
||||
@ -2431,6 +2419,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
|
||||
AudioRenderer* renderer) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::iterator it = receive_channels_.find(ssrc);
|
||||
if (it == receive_channels_.end()) {
|
||||
if (renderer) {
|
||||
@ -2475,6 +2464,7 @@ bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
|
||||
|
||||
bool WebRtcVoiceMediaChannel::GetActiveStreams(
|
||||
AudioInfo::StreamList* actives) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
// In conference mode, the default channel should not be in
|
||||
// |receive_channels_|.
|
||||
actives->clear();
|
||||
@ -2488,6 +2478,7 @@ bool WebRtcVoiceMediaChannel::GetActiveStreams(
|
||||
}
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetOutputLevel() {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
// return the highest output level of all streams
|
||||
int highest = GetOutputLevel(voe_channel());
|
||||
for (const auto& ch : receive_channels_) {
|
||||
@ -2524,6 +2515,7 @@ void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
|
||||
|
||||
bool WebRtcVoiceMediaChannel::SetOutputScaling(
|
||||
uint32 ssrc, double left, double right) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
rtc::CritScope lock(&receive_channels_cs_);
|
||||
// Collect the channels to scale the output volume.
|
||||
std::vector<int> channels;
|
||||
@ -2536,7 +2528,7 @@ bool WebRtcVoiceMediaChannel::SetOutputScaling(
|
||||
channels.push_back(ch.second->channel());
|
||||
}
|
||||
} else { // Collect only the channel of the specified ssrc.
|
||||
int channel = GetReceiveChannelNum(ssrc);
|
||||
int channel = GetReceiveChannelId(ssrc);
|
||||
if (-1 == channel) {
|
||||
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
|
||||
return false;
|
||||
@ -2597,7 +2589,7 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
|
||||
channel = send_channels_.begin()->second->channel();
|
||||
}
|
||||
} else {
|
||||
channel = GetSendChannelNum(ssrc);
|
||||
channel = GetSendChannelId(ssrc);
|
||||
}
|
||||
if (channel == -1) {
|
||||
LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
|
||||
@ -2639,7 +2631,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
|
||||
// any multiplexed streams, just send it to the default channel. Otherwise,
|
||||
// send it to the specific decoder instance for that stream.
|
||||
int which_channel =
|
||||
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), false));
|
||||
GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), false));
|
||||
if (which_channel == -1) {
|
||||
which_channel = voe_channel();
|
||||
}
|
||||
@ -2675,7 +2667,7 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
|
||||
bool has_sent_to_default_channel = false;
|
||||
if (type == kRtcpTypeSR) {
|
||||
int which_channel =
|
||||
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), true));
|
||||
GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), true));
|
||||
if (which_channel != -1) {
|
||||
engine()->voe()->network()->ReceivedRTCPPacket(
|
||||
which_channel, packet->data(), packet->size());
|
||||
@ -2700,7 +2692,7 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
|
||||
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
|
||||
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelId(ssrc);
|
||||
if (channel == -1) {
|
||||
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
|
||||
return false;
|
||||
@ -2784,6 +2776,8 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
|
||||
bool echo_metrics_on = false;
|
||||
// These can take on valid negative values, so use the lowest possible level
|
||||
// as default rather than -1.
|
||||
@ -2970,42 +2964,10 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
||||
return true;
|
||||
}
|
||||
|
||||
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
|
||||
rtc::CritScope lock(&receive_channels_cs_);
|
||||
RTC_DCHECK(ssrc != NULL);
|
||||
if (channel_num == -1 && send_ != SEND_NOTHING) {
|
||||
// Sometimes the VoiceEngine core will throw error with channel_num = -1.
|
||||
// This means the error is not limited to a specific channel. Signal the
|
||||
// message using ssrc=0. If the current channel is sending, use this
|
||||
// channel for sending the message.
|
||||
*ssrc = 0;
|
||||
return true;
|
||||
} else {
|
||||
// Check whether this is a sending channel.
|
||||
for (const auto& ch : send_channels_) {
|
||||
if (ch.second->channel() == channel_num) {
|
||||
// This is a sending channel.
|
||||
uint32 local_ssrc = 0;
|
||||
if (engine()->voe()->rtp()->GetLocalSSRC(
|
||||
channel_num, local_ssrc) != -1) {
|
||||
*ssrc = local_ssrc;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
}
|
||||
|
||||
// Check whether this is a receiving channel.
|
||||
for (const auto& ch : receive_channels_) {
|
||||
if (ch.second->channel() == channel_num) {
|
||||
*ssrc = ch.first;
|
||||
return true;
|
||||
}
|
||||
}
|
||||
void WebRtcVoiceMediaChannel::OnError(int error) {
|
||||
if (send_ == SEND_NOTHING) {
|
||||
return;
|
||||
}
|
||||
return false;
|
||||
}
|
||||
|
||||
void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
|
||||
if (error == VE_TYPING_NOISE_WARNING) {
|
||||
typing_noise_detected_ = true;
|
||||
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
|
||||
@ -3014,20 +2976,21 @@ void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
|
||||
}
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
|
||||
unsigned int ulevel;
|
||||
int ret =
|
||||
engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
|
||||
unsigned int ulevel = 0;
|
||||
int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
|
||||
return (ret == 0) ? static_cast<int>(ulevel) : -1;
|
||||
}
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) const {
|
||||
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32 ssrc) const {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
|
||||
if (it != receive_channels_.end())
|
||||
return it->second->channel();
|
||||
return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
|
||||
}
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) const {
|
||||
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32 ssrc) const {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::const_iterator it = send_channels_.find(ssrc);
|
||||
if (it != send_channels_.end())
|
||||
return it->second->channel();
|
||||
@ -3219,6 +3182,7 @@ void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32 ssrc) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
|
||||
const std::vector<AudioCodec>& new_codecs) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
for (const AudioCodec& codec : new_codecs) {
|
||||
webrtc::CodecInst voe_codec;
|
||||
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
|
||||
|
||||
@ -127,15 +127,12 @@ class WebRtcVoiceEngine
|
||||
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
|
||||
|
||||
// webrtc::VoiceEngineObserver:
|
||||
void CallbackOnError(int channel, int errCode) override;
|
||||
void CallbackOnError(int channel_id, int errCode) override;
|
||||
|
||||
// Given the device type, name, and id, find device id. Return true and
|
||||
// set the output parameter rtc_id if successful.
|
||||
bool FindWebRtcAudioDeviceId(
|
||||
bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
|
||||
bool FindChannelAndSsrc(int channel_num,
|
||||
WebRtcVoiceMediaChannel** channel,
|
||||
uint32* ssrc) const;
|
||||
|
||||
void StartAecDump(const std::string& filename);
|
||||
void StopAecDump();
|
||||
@ -237,11 +234,10 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
||||
return VoiceMediaChannel::SendRtcp(&packet);
|
||||
}
|
||||
|
||||
bool FindSsrc(int channel_num, uint32* ssrc);
|
||||
void OnError(uint32 ssrc, int error);
|
||||
void OnError(int error);
|
||||
|
||||
int GetReceiveChannelNum(uint32 ssrc) const;
|
||||
int GetSendChannelNum(uint32 ssrc) const;
|
||||
int GetReceiveChannelId(uint32 ssrc) const;
|
||||
int GetSendChannelId(uint32 ssrc) const;
|
||||
|
||||
private:
|
||||
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
|
||||
|
||||
@ -2866,23 +2866,23 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
|
||||
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
|
||||
}
|
||||
|
||||
// Test that GetReceiveChannelNum returns the default channel for the first
|
||||
// Test that GetReceiveChannelId returns the default channel for the first
|
||||
// recv stream in 1-1 calls.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelNumIn1To1Calls) {
|
||||
TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelIdIn1To1Calls) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
cricket::WebRtcVoiceMediaChannel* media_channel =
|
||||
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
|
||||
// Test that GetChannelNum returns the default channel if the SSRC is unknown.
|
||||
EXPECT_EQ(media_channel->voe_channel(),
|
||||
media_channel->GetReceiveChannelNum(0));
|
||||
media_channel->GetReceiveChannelId(0));
|
||||
cricket::StreamParams stream;
|
||||
stream.ssrcs.push_back(kSsrc2);
|
||||
EXPECT_TRUE(channel_->AddRecvStream(stream));
|
||||
EXPECT_EQ(media_channel->voe_channel(),
|
||||
media_channel->GetReceiveChannelNum(kSsrc2));
|
||||
media_channel->GetReceiveChannelId(kSsrc2));
|
||||
}
|
||||
|
||||
// Test that GetReceiveChannelNum doesn't return the default channel for the
|
||||
// Test that GetReceiveChannelId doesn't return the default channel for the
|
||||
// first recv stream in conference calls.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, TestGetChannelNumInConferenceCalls) {
|
||||
EXPECT_TRUE(SetupEngine());
|
||||
@ -2894,7 +2894,7 @@ TEST_F(WebRtcVoiceEngineTestFake, TestGetChannelNumInConferenceCalls) {
|
||||
cricket::WebRtcVoiceMediaChannel* media_channel =
|
||||
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
|
||||
EXPECT_LT(media_channel->voe_channel(),
|
||||
media_channel->GetReceiveChannelNum(kSsrc2));
|
||||
media_channel->GetReceiveChannelId(kSsrc2));
|
||||
}
|
||||
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetOutputScaling) {
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user