| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| audio_coding_module.cc | 45 int AudioCodingModule::Codec(const char* payload_name,
53 payload_name, sampling_freq_hz, channels);
76 int AudioCodingModule::Codec(const char* payload_name,
79 return acm2::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
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| acm_codec_database.cc | 510 int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) {
519 name_match = (STR_CASE_CMP(database_[id].plname, payload_name) == 0);
522 if (STR_CASE_CMP(payload_name, "opus") != 0) {
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| acm_codec_database.h | 267 static int CodecId(const char* payload_name, int frequency, int channels);
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_audio.h | 72 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
80 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
87 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
| rtp_payload_registry.cc | 39 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
46 assert(payload_name);
68 size_t payload_name_length = strlen(payload_name);
87 payload->name, payload_name, payload_name_length)) {
100 payload_name, payload_name_length, frequency, channels, rate);
106 if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) {
111 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
112 } else if (ModuleRTPUtility::StringCompare(payload_name, "ulpfec", 3)) {
117 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
121 payload_name, payload_type, frequency, channels, rate)
[all...] |
| rtp_receiver_video.h | 50 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
58 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| rtp_receiver_impl.cc | 110 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
122 payload_name, payload_type, frequency, channels, rate,
125 if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
127 LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
279 char payload_name[RTP_PAYLOAD_NAME_SIZE]; local
311 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
312 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
331 id_, rtp_header.payloadType, payload_name,
355 char payload_name[RTP_PAYLOAD_NAME_SIZE]; local
402 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0
[all...] |
| rtp_receiver_audio.cc | 157 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
162 if (ModuleRTPUtility::StringCompare(payload_name, "telephone-event", 15)) {
165 if (ModuleRTPUtility::StringCompare(payload_name, "cn", 2)) {
272 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
276 payload_name,
281 << payload_name << "/" << payload_type;
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| rtp_receiver_strategy.h | 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| rtp_receiver_impl.h | 40 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| rtp_receiver_video.cc | 43 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
90 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
94 id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
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| rtp_sender_unittest.cc | 736 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local
738 ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
820 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local
875 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local
962 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local
1011 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; local
1040 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; local
[all...] |
| rtp_sender.cc | 220 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
223 assert(payload_name);
235 if (ModuleRTPUtility::StringCompare(payload->name, payload_name,
254 ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
257 ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
[all...] |
| rtp_sender.h | 93 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_payload_registry.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
72 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
145 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| rtp_receiver.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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| /external/chromium_org/third_party/webrtc/test/ |
| encoder_settings.cc | 61 strcpy(codec.plName, encoder_settings.payload_name.c_str());
63 (encoder_settings.payload_name == "VP8" ? kVideoCodecVP8
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| codec.cc | 172 const char* payload_name = name.c_str(); local
173 if (_stricmp(payload_name, kRedCodecName) == 0) {
176 if (_stricmp(payload_name, kUlpfecCodecName) == 0) {
179 if (_stricmp(payload_name, kRtxCodecName) == 0) {
|
| /external/chromium_org/third_party/webrtc/ |
| video_send_stream.h | 71 std::string payload_name; member in struct:webrtc::VideoSendStream::Config::EncoderSettings
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/interface/ |
| audio_coding_module.h | 140 // -payload_name : name of the codec.
153 static int Codec(const char* payload_name, CodecInst* codec,
163 // -payload_name : name of the codec.
172 static int Codec(const char* payload_name, int sampling_freq_hz,
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| loopback.cc | 78 send_config.encoder_settings.payload_name = "VP8";
|
| video_send_stream.cc | 33 ss << "{payload_name: " << payload_name; local
302 (config_.encoder_settings.payload_name == "VP8" ? kVideoCodecVP8
316 config_.encoder_settings.payload_name.c_str(),
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| rampup_tests.cc | 456 send_config.encoder_settings.payload_name = "FAKE";
531 send_config.encoder_settings.payload_name = "FAKE";
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| bitrate_estimator_tests.cc | 74 send_config_.encoder_settings.payload_name = "FAKE";
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_channel.h | 236 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
