Searched refs:frame_size_samples (Results 1 - 12 of 12) sorted by relevance
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| H A D | acm_send_test.h | 39 int frame_size_samples);
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| H A D | acm_send_test_oldapi.h | 41 int frame_size_samples);
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| H A D | acm_send_test.cc | 53 int frame_size_samples) {
55 acm_->RegisterSendCodec(codec_type, payload_type, frame_size_samples);
50 RegisterCodec(int codec_type, int channels, int payload_type, int frame_size_samples) argument
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| H A D | acm_send_test_oldapi.cc | 52 int frame_size_samples) {
57 codec_.pacsize = frame_size_samples;
48 RegisterCodec(const char* payload_name, int sampling_freq_hz, int channels, int payload_type, int frame_size_samples) argument
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| H A D | acm_codec_database.cc | 846 bool ACMCodecDB::IsILBCRateValid(int rate, int frame_size_samples) { argument
847 if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
850 } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
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| H A D | acm_codec_database.h | 324 // [frame_size_samples] - (used for iLBC) specifies which frame size to go
328 static bool IsILBCRateValid(int rate, int frame_size_samples);
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| H A D | audio_coding_module_unittest.cc | 635 int frame_size_samples,
640 codec_type, channels, payload_type, frame_size_samples);
632 RegisterSendCodec(int codec_type, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) argument
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| H A D | audio_coding_module_unittest_oldapi.cc | 646 int frame_size_samples,
654 frame_size_samples);
642 RegisterSendCodec(const char* payload_name, int sampling_freq_hz, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) argument
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| H A D | audio_coding_module_impl.h | 410 int frame_size_samples = 0) OVERRIDE;
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| H A D | audio_coding_module_impl.cc | 2054 int frame_size_samples) {
2066 if (frame_size_samples > 0) {
2067 codec.pacsize = frame_size_samples;
2052 RegisterSendCodec(int encoder_type, uint8_t payload_type, int frame_size_samples) argument
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| H A D | neteq_rtpplay.cc | 106 size_t* frame_size_samples,
494 size_t* frame_size_samples,
512 if (*frame_size_samples !=
514 *frame_size_samples =
517 new int16_t[*frame_size_samples]);
518 *payload_mem_size_bytes = 2 * *frame_size_samples;
535 assert(*frame_size_samples > 0);
536 if (!replacement_audio_file->Read(*frame_size_samples,
545 static_cast<int16_t>(*frame_size_samples),
547 assert(payload_len == 2 * *frame_size_samples);
490 ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file, webrtc::scoped_ptr<int16_t[]>* replacement_audio, webrtc::scoped_ptr<uint8_t[]>* payload, size_t* payload_mem_size_bytes, size_t* frame_size_samples, WebRtcRTPHeader* rtp_header, const webrtc::test::Packet* next_packet) argument
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/interface/ |
| H A D | audio_coding_module.h | 1051 int frame_size_samples = 0) = 0;
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