Cross Reference: /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/initial_delay

Lines Matching refs:codec
68     CodecInst codec;
73         AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
74         acm_b_->RegisterReceiveCodec(codec);
85     CodecInst codec;
86     AudioCodingModule::Codec("L16", &codec, 8000, 1);
87     codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
88     Run(codec, 1000);
92     CodecInst codec;
93     AudioCodingModule::Codec("L16", &codec, 16000, 1);
94     codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
95     Run(codec, 1000);
99     CodecInst codec;
100     AudioCodingModule::Codec("L16", &codec, 32000, 1);
101     codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
102     Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
106     CodecInst codec;
107     AudioCodingModule::Codec("L16", &codec, 8000, 2);
108     codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
109     Run(codec, 1000);
113     CodecInst codec;
114     AudioCodingModule::Codec("L16", &codec, 16000, 2);
115     codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
116     Run(codec, 1000);
120     CodecInst codec;
121     AudioCodingModule::Codec("L16", &codec, 32000, 2);
122     codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
123     Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
127   void Run(CodecInst codec, int initial_delay_ms) {
132     in_audio_frame.sample_rate_hz_ = codec.plfreq;
133     in_audio_frame.num_channels_ = codec.channels;
134     in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
143     ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
150       ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));