Searched defs:send_codec (Results 1 - 10 of 10) sorted by relevance
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
H A D | file_test.cc | 64 webrtc::CodecInst send_codec; local 65 voe_codec_->GetSendCodec(channel_, send_codec); 72 channel_, recording_filename.c_str(), &send_codec));
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | codec_manager.cc | 25 int IsValidSendCodec(const CodecInst& send_codec) { argument 27 if ((send_codec.channels != 1) && (send_codec.channels != 2)) { 31 send_codec.channels); 35 auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec); 43 if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) { 49 if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels) 53 send_codec.channels, send_codec.plname); 75 bool CodecManager::RegisterEncoder(const CodecInst& send_codec) { argument [all...] |
H A D | audio_coding_module_impl.cc | 199 int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) { argument 201 if (!codec_manager_.RegisterEncoder(send_codec)) {
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/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | TestStereo.cc | 832 auto send_codec = acm_a_->SendCodec(); local 834 ASSERT_TRUE(send_codec); 835 printf("%s -> ", send_codec->plname);
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | codec_database.cc | 193 bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec, argument 196 RTC_DCHECK(send_codec); 201 RTC_DCHECK_GE(send_codec->plType, 1); 203 RTC_DCHECK_LE(send_codec->startBitrate, 1000000u); 204 RTC_DCHECK(send_codec->codecType != kVideoCodecUnknown); 216 memcpy(&new_send_codec, send_codec, sizeof(new_send_codec)); 220 new_send_codec.maxBitrate = (static_cast<int>(send_codec->height) * 221 static_cast<int>(send_codec->width) * 222 static_cast<int>(send_codec->maxFramerate)) / 224 if (send_codec [all...] |
/external/webrtc/talk/media/webrtc/ |
H A D | fakewebrtcvoiceengine.h | 167 memset(&send_codec, 0, sizeof(send_codec)); 186 webrtc::CodecInst send_codec; member in struct:cricket::FakeWebRtcVoiceEngine::Channel 410 channels_[channel]->send_codec = codec; 416 codec = channels_[channel]->send_codec; 478 if (channels_[channel]->send_codec.channels == 2) { 490 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { 506 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { 525 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
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H A D | webrtcvoiceengine.cc | 1480 webrtc::CodecInst send_codec; local 1481 memset(&send_codec, 0, sizeof(send_codec)); 1510 if (!GetRedSendCodec(codec, codecs, &send_codec)) { 1522 send_codec = voe_codec; 1527 GetOpusConfig(codec, &send_codec, &enable_codec_fec, 1534 if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(&send_codec, ptime_ms)) { 1536 << send_codec.plname; 1557 if (!SetSendCodec(channel, send_codec)) 1571 if (IsCodec(send_codec, kOpusCodecNam 1711 SetSendCodec( int channel, const webrtc::CodecInst& send_codec) argument 2450 GetRedSendCodec(const AudioCodec& red_codec, const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) argument [all...] |
/external/webrtc/webrtc/video/ |
H A D | vie_encoder.cc | 247 VideoCodec send_codec; local 257 send_codec = encoder_config_; 266 if (send_codec.numberOfSimulcastStreams == 0) { 267 pad_up_to_bitrate_bps = send_codec.minBitrate * 1000; 269 SimulcastStream* stream_configs = send_codec.simulcastStream; 271 stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate * 273 for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) { 280 if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1) 580 VideoCodec send_codec; local 587 send_codec [all...] |
/external/webrtc/talk/media/base/ |
H A D | fakemediaengine.h | 499 virtual bool GetSendCodec(VideoCodec* send_codec) { argument 503 *send_codec = send_codecs_[0];
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/external/webrtc/webrtc/voice_engine/ |
H A D | channel.cc | 1375 auto send_codec = audio_coding_->SendCodec(); local 1376 if (send_codec) { 1377 codec = *send_codec;
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