| /external/chromium_org/media/base/ |
| H A D | audio_buffer_queue_unittest.cc | 9 #include "media/base/audio_buffer.h"
355 scoped_refptr<AudioBuffer> audio_buffer = local
368 buffer.Append(audio_buffer);
379 frames_read * audio_buffer->duration() / audio_buffer->frame_count(),
387 frames_read * audio_buffer->duration() / audio_buffer->frame_count(),
406 frames_read * audio_buffer->duration() / audio_buffer->frame_count(),
418 frames_read * audio_buffer
[all...] |
| /external/chromium_org/media/audio/mac/ |
| H A D | audio_input_mac.cc | 174 AudioQueueBufferRef audio_buffer) {
176 return AudioQueueEnqueueBuffer(audio_queue_, audio_buffer, 0, NULL);
183 AudioQueueBufferRef audio_buffer,
188 HandleInputBuffer(audio_queue, audio_buffer, start_time,
194 AudioQueueBufferRef audio_buffer,
199 DCHECK(audio_buffer->mAudioData);
202 DCHECK_EQ(0U, audio_buffer->mAudioDataByteSize);
206 if (audio_buffer->mAudioDataByteSize) {
221 uint8* audio_data = reinterpret_cast<uint8*>(audio_buffer->mAudioData);
225 this, audio_bus_.get(), audio_buffer
173 QueueNextBuffer( AudioQueueBufferRef audio_buffer) argument
180 HandleInputBufferStatic( void* data, AudioQueueRef audio_queue, AudioQueueBufferRef audio_buffer, const AudioTimeStamp* start_time, UInt32 num_packets, const AudioStreamPacketDescription* desc) argument
192 HandleInputBuffer( AudioQueueRef audio_queue, AudioQueueBufferRef audio_buffer, const AudioTimeStamp* start_time, UInt32 num_packets, const AudioStreamPacketDescription* packet_desc) argument
[all...] |
| H A D | audio_low_latency_input_mac.cc | 81 AudioBuffer* audio_buffer = audio_buffer_list_.mBuffers; local
82 audio_buffer->mNumberChannels = input_params.channels();
83 audio_buffer->mDataByteSize = data_byte_size;
84 audio_buffer->mData = audio_data_buffer_.get();
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| H A D | fakeaudiocapturemodule_unittest.cc | 116 int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) { argument
117 memset(audio_buffer, 0, audio_buffer_size);
120 int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) { argument
123 memcpy(audio_buffer, rec_buffer_, min_buffer_size);
|
| /external/chromium_org/content/renderer/media/ |
| H A D | webrtc_audio_device_impl.cc | 82 const int16* audio_buffer = audio_data; local
102 audio_buffer,
112 audio_buffer += frames_per_10_ms * number_of_channels;
|
| /external/srec/srec/test/SRecTest/src/ |
| H A D | SRecTest.c | 144 asr_int16_t audio_buffer [MAX_AUDIO_BUFFER_SIZE]; member in struct:ApplicationData_t
2138 data->num_samples_read = pfread ( data->audio_buffer, sizeof ( asr_int16_t ), data->audio_buffer_requested_size, audio_file );
2169 esr_status = SR_RecognizerPutAudio ( data->recognizer, data->audio_buffer, &data->num_samples_read, hit_eof );
|
| /external/srec/srec/test/SRecTestAudio/src/ |
| H A D | SRecTestAudio.c | 143 asr_int16_t audio_buffer [MAX_AUDIO_BUFFER_SIZE]; member in struct:ApplicationData_t
1831 data->num_samples_read = pfread ( data->audio_buffer, sizeof ( asr_int16_t ), data->audio_buffer_requested_size, audio_file );
1862 esr_status = SR_RecognizerPutAudio ( data->recognizer, data->audio_buffer, &data->num_samples_read, hit_eof );
2403 audio_status = lhs_audioinGetSamples ( audio_input_handle, &data__num_samples_read, data->audio_buffer, &input_status );
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