/dports/net/sems/sems-f89581a/core/ |
H A D | AmAudioMixIn.cpp | 45 int output_sample_rate, unsigned int nb_samples) { in get() argument 66 return A->get(system_ts, buffer, output_sample_rate, nb_samples); in get() 70 int res = B->get(system_ts, buffer, output_sample_rate, nb_samples); in get() 72 res = A->get(system_ts, buffer, output_sample_rate, nb_samples); in get() 90 output_sample_rate, nb_samples); in get() 113 output_sample_rate, nb_samples); in get()
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H A D | AmAudio.cpp | 290 int output_sample_rate, unsigned int nb_samples) 293 / (float)output_sample_rate)); 310 getSampleRate(), output_sample_rate); 417 …resampleInput(unsigned char* buffer, unsigned int s, int input_sample_rate, int output_sample_rate) 419 if ((input_sample_rate == output_sample_rate) && !input_resampling_state.get()) { 440 return resample(*input_resampling_state, buffer, s, input_sample_rate, output_sample_rate); 443 …esampleOutput(unsigned char* buffer, unsigned int s, int input_sample_rate, int output_sample_rate) 445 if ((input_sample_rate == output_sample_rate) 467 return resample(*output_resampling_state, buffer, s, input_sample_rate, output_sample_rate); 470 …tate& rstate, unsigned char* buffer, unsigned int s, int input_sample_rate, int output_sample_rate) [all …]
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H A D | AmAdvancedAudio.cpp | 93 int output_sample_rate, unsigned int nb_samples) in get() argument 103 output_sample_rate, size_trav); in get() 108 output_sample_rate = it->audio->getSampleRate(); in get() 110 output_sample_rate, size_trav>>1); in get() 299 int output_sample_rate, unsigned int nb_samples) { in get() argument 306 res = back_audio->get(system_ts, buffer, output_sample_rate, nb_samples); in get() 308 res = AmPlaylist::get(system_ts, buffer, output_sample_rate, nb_samples); in get() 316 int output_sample_rate, unsigned int size) in put() argument 336 int output_sample_rate, unsigned int nb_samples) in get() argument
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H A D | AmConferenceChannel.cpp | 69 int output_sample_rate, unsigned int nb_samples) in get() argument 71 if (!nb_samples || !output_sample_rate) in get() 76 unsigned int size = output_sample_rate ? in get() 77 PCM16_S2B(nb_samples * mixer->GetCurrentSampleRate() / output_sample_rate) : 0; in get() 97 size = resampleOutput(buffer,size,mixer_sample_rate,output_sample_rate); in get()
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H A D | AmBufferedAudio.cpp | 81 int output_sample_rate, unsigned int nb_samples) in get() argument 84 return AmAudio::get(system_ts, buffer, output_sample_rate, nb_samples); in get() 90 size_t nget = PCM16_S2B(nb_samples * getSampleRate() / output_sample_rate); in get() 106 int size = resampleOutput(samples,nget,getSampleRate(),output_sample_rate); in get()
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H A D | AmAudio.h | 307 …mpleInput(unsigned char *buffer, unsigned int size, int input_sample_rate, int output_sample_rate); in ~AmInternalResamplerState() 315 …pleOutput(unsigned char *buffer, unsigned int size, int input_sample_rate, int output_sample_rate); in ~AmInternalResamplerState() 326 …& rstate, unsigned char *buffer, unsigned int size, int input_sample_rate, int output_sample_rate); in ~AmInternalResamplerState() 358 int output_sample_rate, unsigned int nb_samples); in ~AmInternalResamplerState()
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/content/renderer/media/audio/ |
H A D | audio_renderer_mixer_manager.cc | 35 int output_sample_rate, preferred_output_buffer_size; in GetMixerOutputParams() local 39 output_sample_rate = input_params.sample_rate(); in GetMixerOutputParams() 44 output_sample_rate = input_params.sample_rate(); in GetMixerOutputParams() 57 output_sample_rate = hardware_params.sample_rate(); in GetMixerOutputParams() 71 output_sample_rate, preferred_output_buffer_size); in GetMixerOutputParams() 75 output_sample_rate, preferred_output_buffer_size); in GetMixerOutputParams() 87 output_sample_rate, output_buffer_size); in GetMixerOutputParams()
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/dports/multimedia/libxine/xine-lib-1.2.11/src/audio_out/ |
H A D | audio_esd_out.c | 58 int32_t output_sample_rate, input_sample_rate; member 121 return this->output_sample_rate; in ao_esd_open() 128 this->output_sample_rate = rate; in ao_esd_open() 153 if (this->output_sample_rate > this->server_sample_rate) in ao_esd_open() 154 this->output_sample_rate = this->server_sample_rate; in ao_esd_open() 157 this->output_sample_rate = this->server_sample_rate; in ao_esd_open() 159 this->output_sample_k_rate = this->output_sample_rate / 1000; in ao_esd_open() 170 return this->output_sample_rate; in ao_esd_open() 201 * this->output_sample_rate; in ao_esd_delay() 237 * this->output_sample_rate; in ao_esd_write() [all …]
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H A D | audio_irixal_out.c | 68 int32_t output_sample_rate, input_sample_rate; 184 this->output_sample_rate = this->input_sample_rate; 187 this->output_sample_rate = alFixedToInt (parvalue.value.ll); 189 if (this->input_sample_rate != this->output_sample_rate) 191 this->input_sample_rate, this->output_sample_rate); 193 return this->output_sample_rate;
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/blink/renderer/modules/media/audio/ |
H A D | audio_renderer_mixer_manager.cc | 38 int output_sample_rate, preferred_output_buffer_size; in GetMixerOutputParams() local 42 output_sample_rate = input_params.sample_rate(); in GetMixerOutputParams() 47 output_sample_rate = input_params.sample_rate(); in GetMixerOutputParams() 60 output_sample_rate = hardware_params.sample_rate(); in GetMixerOutputParams() 74 output_sample_rate, preferred_output_buffer_size); in GetMixerOutputParams() 78 output_sample_rate, preferred_output_buffer_size); in GetMixerOutputParams() 90 output_sample_rate, output_buffer_size); in GetMixerOutputParams()
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/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 212 int output_sample_rate = msg.output_sample_rate(); 213 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); 230 if (output_sample_rate == 0) { 231 output_sample_rate = input_sample_rate; 236 output_samples_per_channel = output_sample_rate / 100; 248 output_sample_rate,
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/dports/multimedia/vapoursynth-l-smash-works/L-SMASH-Works-0.0-940-g198cc78/common/ |
H A D | libavsmash_audio.c | 436 int output_sample_rate in count_sequence_output_pcm_samples() argument 440 if( output_sample_rate == current_sample_rate ) in count_sequence_output_pcm_samples() 443 …resampled_sample_count = av_rescale_rnd( sequence_pcm_count, output_sample_rate, current_sample_ra… in count_sequence_output_pcm_samples() 482 int output_sample_rate, in libavsmash_audio_count_overall_pcm_samples() argument 509 output_sample_rate ); in libavsmash_audio_count_overall_pcm_samples() 525 output_sample_rate ); in libavsmash_audio_count_overall_pcm_samples() 526 overall_pcm_count -= av_rescale( start_time, output_sample_rate, adhp->media_timescale ); in libavsmash_audio_count_overall_pcm_samples() 580 int output_sample_rate, in find_start_audio_frame() argument 616 output_sample_rate ); in find_start_audio_frame() 623 …*start_offset = av_rescale_rnd( *start_offset, current_sample_rate, output_sample_rate, AV_ROUND_… in find_start_audio_frame() [all …]
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H A D | lwlibav_audio.c | 220 int output_sample_rate in lwlibav_audio_count_overall_pcm_samples() argument 233 …uint64_t resampled_sample_count = output_sample_rate == current_sample_rate || pcm_sample_count ==… in lwlibav_audio_count_overall_pcm_samples() 235 … : (pcm_sample_count * output_sample_rate - 1) / current_sample_rate + 1; in lwlibav_audio_count_overall_pcm_samples() 247 … overall_pcm_sample_count += (pcm_sample_count * output_sample_rate - 1) / current_sample_rate + 1; in lwlibav_audio_count_overall_pcm_samples() 256 int output_sample_rate, in find_start_audio_frame() argument 283 resampled_sample_count = output_sample_rate == current_sample_rate || pcm_sample_count == 0 in find_start_audio_frame() 285 … : (pcm_sample_count * output_sample_rate - 1) / current_sample_rate + 1; in find_start_audio_frame() 292 if( *start_offset && current_sample_rate != output_sample_rate ) in find_start_audio_frame() 293 *start_offset = (*start_offset * current_sample_rate - 1) / output_sample_rate + 1; in find_start_audio_frame() 545 …frame_number = find_start_audio_frame( adhp, aohp->output_sample_rate, start_frame_pos, &aohp->out… in lwlibav_audio_get_pcm_samples()
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/dports/devel/py-aiortc/aiortc-1.2.1/tests/ |
H A D | codecs.py | 66 def roundtrip_audio(self, codec, output_layout, output_sample_rate, drop=[]): argument 77 output_sample_count = int(output_sample_rate * AUDIO_PTIME) 94 self.assertEqual(frames[0].samples, output_sample_rate * AUDIO_PTIME) 95 self.assertEqual(frames[0].sample_rate, output_sample_rate) 98 frames[0].time_base, fractions.Fraction(1, output_sample_rate)
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H A D | test_g711.py | 60 self.roundtrip_audio(PCMA_CODEC, output_layout="mono", output_sample_rate=8000) 64 PCMA_CODEC, output_layout="mono", output_sample_rate=8000, drop=[1] 118 self.roundtrip_audio(PCMU_CODEC, output_layout="mono", output_sample_rate=8000) 122 PCMU_CODEC, output_layout="mono", output_sample_rate=8000, drop=[1]
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 264 int output_sample_rate = msg.output_sample_rate(); in do_main() local 265 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main() 282 if (output_sample_rate == 0) { in do_main() 283 output_sample_rate = input_sample_rate; in do_main() 288 output_samples_per_channel = output_sample_rate / 100; in do_main() 300 output_sample_rate, in do_main()
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/dports/www/chromium-legacy/chromium-88.0.4324.182/chromecast/media/cma/backend/mixer/ |
H A D | filter_group.cc | 89 output_config_.output_sample_rate; in Initialize() 90 DCHECK_EQ(input_frames_per_write_ * output_config_.output_sample_rate, in Initialize() 94 << output_config_.output_sample_rate; in Initialize() 97 input_config.output_sample_rate = input_samples_per_second_; in Initialize() 162 DCHECK_NE(output_config_.output_sample_rate, 0); in MixAndFilter() 260 if (output_config_.output_sample_rate == 0) { in GetRenderingDelayMicroseconds() 343 << output_config_.output_sample_rate << "hz"; in PrintTopology()
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 285 int output_sample_rate = msg.output_sample_rate(); in do_main() local 286 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main() 306 if (output_sample_rate == 0) { in do_main() 307 output_sample_rate = input_sample_rate; in do_main() 315 static_cast<size_t>(output_sample_rate / 100); in do_main() 333 output_sample_rate, in do_main()
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 275 int output_sample_rate = msg.output_sample_rate(); in do_main() local 276 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main() 296 if (output_sample_rate == 0) { in do_main() 297 output_sample_rate = input_sample_rate; in do_main() 305 static_cast<size_t>(output_sample_rate / 100); in do_main() 323 output_sample_rate, in do_main()
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 285 int output_sample_rate = msg.output_sample_rate(); in do_main() local 286 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main() 306 if (output_sample_rate == 0) { in do_main() 307 output_sample_rate = input_sample_rate; in do_main() 315 static_cast<size_t>(output_sample_rate / 100); in do_main() 333 output_sample_rate, in do_main()
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 285 int output_sample_rate = msg.output_sample_rate(); in do_main() local 286 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); in do_main() 306 if (output_sample_rate == 0) { in do_main() 307 output_sample_rate = input_sample_rate; in do_main() 315 static_cast<size_t>(output_sample_rate / 100); in do_main() 333 output_sample_rate, in do_main()
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/dports/www/chromium-legacy/chromium-88.0.4324.182/chromecast/media/audio/ |
H A D | cast_audio_resampler_impl.cc | 27 int output_sample_rate) in CastAudioResamplerImpl() argument 30 static_cast<double>(input_sample_rate) / output_sample_rate, in CastAudioResamplerImpl() 149 int output_sample_rate) { in Create() argument 151 channel_count, input_sample_rate, output_sample_rate); in Create()
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/dports/audio/musicpd/mpd-0.23.6/src/lib/alsa/ |
H A D | HwSetup.cxx | 216 unsigned output_sample_rate = requested_sample_rate; in SetupHw() local 219 &output_sample_rate, nullptr); in SetupHw() 225 if (output_sample_rate == 0) in SetupHw() 229 if (output_sample_rate != requested_sample_rate) in SetupHw() 230 audio_format.sample_rate = params.CalcInputSampleRate(output_sample_rate); in SetupHw()
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/dports/multimedia/vapoursynth-l-smash-works/L-SMASH-Works-0.0-940-g198cc78/AviSynth/ |
H A D | audio_output.cpp | 72 aohp->output_sample_rate = sample_rate; in as_setup_audio_rendering() 109 av_opt_set_int( avr_ctx, "out_sample_rate", aohp->output_sample_rate, 0 ); in as_setup_audio_rendering() 115 vi->audio_samples_per_second = aohp->output_sample_rate; in as_setup_audio_rendering()
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/dports/net/sems/sems-f89581a/apps/rtmp/ |
H A D | RtmpAudio.cpp | 85 int output_sample_rate, unsigned int nb_samples) in get() argument 96 / (float)output_sample_rate); in get() 103 if(output_sample_rate != getSampleRate()) { in get() 105 getSampleRate(), output_sample_rate); in get()
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