918b554789
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that OpenSL ES is not accidentally activated in existing clients. There are still some unresolved issues to sort out before it can be utilized. Enables possibility to use OpenSL ES based audio in both directions for WebRTC. All unit tests and demo clients have been tested with the new implementation but the new support is behind a flag (see above). More testing is needed before it can be used in the field and additional support for hardware effects is still missing. BUG=webrtc:5925 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/2119633004 . Cr-Commit-Position: refs/heads/master@{#14290}
104 lines
3.5 KiB
C++
104 lines
3.5 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_device/android/opensles_common.h"
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#include <SLES/OpenSLES.h>
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#include "webrtc/base/arraysize.h"
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#include "webrtc/base/checks.h"
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namespace webrtc {
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// Returns a string representation given an integer SL_RESULT_XXX code.
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// The mapping can be found in <SLES/OpenSLES.h>.
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const char* GetSLErrorString(size_t code) {
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static const char* sl_error_strings[] = {
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"SL_RESULT_SUCCESS", // 0
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"SL_RESULT_PRECONDITIONS_VIOLATED", // 1
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"SL_RESULT_PARAMETER_INVALID", // 2
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"SL_RESULT_MEMORY_FAILURE", // 3
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"SL_RESULT_RESOURCE_ERROR", // 4
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"SL_RESULT_RESOURCE_LOST", // 5
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"SL_RESULT_IO_ERROR", // 6
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"SL_RESULT_BUFFER_INSUFFICIENT", // 7
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"SL_RESULT_CONTENT_CORRUPTED", // 8
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"SL_RESULT_CONTENT_UNSUPPORTED", // 9
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"SL_RESULT_CONTENT_NOT_FOUND", // 10
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"SL_RESULT_PERMISSION_DENIED", // 11
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"SL_RESULT_FEATURE_UNSUPPORTED", // 12
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"SL_RESULT_INTERNAL_ERROR", // 13
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"SL_RESULT_UNKNOWN_ERROR", // 14
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"SL_RESULT_OPERATION_ABORTED", // 15
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"SL_RESULT_CONTROL_LOST", // 16
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};
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if (code >= arraysize(sl_error_strings)) {
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return "SL_RESULT_UNKNOWN_ERROR";
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}
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return sl_error_strings[code];
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}
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SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
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int sample_rate,
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size_t bits_per_sample) {
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RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
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SLDataFormat_PCM format;
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format.formatType = SL_DATAFORMAT_PCM;
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format.numChannels = static_cast<SLuint32>(channels);
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// Note that, the unit of sample rate is actually in milliHertz and not Hertz.
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switch (sample_rate) {
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case 8000:
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format.samplesPerSec = SL_SAMPLINGRATE_8;
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break;
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case 16000:
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format.samplesPerSec = SL_SAMPLINGRATE_16;
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break;
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case 22050:
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format.samplesPerSec = SL_SAMPLINGRATE_22_05;
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break;
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case 32000:
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format.samplesPerSec = SL_SAMPLINGRATE_32;
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break;
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case 44100:
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format.samplesPerSec = SL_SAMPLINGRATE_44_1;
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break;
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case 48000:
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format.samplesPerSec = SL_SAMPLINGRATE_48;
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break;
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case 64000:
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format.samplesPerSec = SL_SAMPLINGRATE_64;
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break;
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case 88200:
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format.samplesPerSec = SL_SAMPLINGRATE_88_2;
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break;
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case 96000:
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format.samplesPerSec = SL_SAMPLINGRATE_96;
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break;
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default:
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RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
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break;
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}
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format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
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format.endianness = SL_BYTEORDER_LITTLEENDIAN;
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if (format.numChannels == 1) {
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format.channelMask = SL_SPEAKER_FRONT_CENTER;
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} else if (format.numChannels == 2) {
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format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
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} else {
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RTC_CHECK(false) << "Unsupported number of channels: "
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<< format.numChannels;
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}
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return format;
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}
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} // namespace webrtc
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