Index: webrtc/api/audio_codecs/L16/audio_encoder_L16.cc |
diff --git a/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..bd243897a1ad235645ab3a2cf8033353080da002 |
--- /dev/null |
+++ b/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc |
@@ -0,0 +1,55 @@ |
+/* |
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h" |
+ |
+#include "webrtc/common_types.h" |
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" |
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h" |
+#include "webrtc/rtc_base/ptr_util.h" |
+ |
+namespace webrtc { |
+ |
+rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig( |
+ const SdpAudioFormat& format) { |
+ Config config; |
+ config.sample_rate_hz = format.clockrate_hz; |
+ config.num_channels = format.num_channels; |
+ return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk() |
+ ? rtc::Optional<Config>(config) |
+ : rtc::Optional<Config>(); |
+} |
+ |
+void AudioEncoderL16::AppendSupportedEncoders( |
+ std::vector<AudioCodecSpec>* specs) { |
+ Pcm16BAppendSupportedCodecSpecs(specs); |
+} |
+ |
+AudioCodecInfo AudioEncoderL16::QueryAudioEncoder( |
+ const AudioEncoderL16::Config& config) { |
+ RTC_DCHECK(config.IsOk()); |
+ return {config.sample_rate_hz, |
+ rtc::dchecked_cast<size_t>(config.num_channels), |
+ config.sample_rate_hz * config.num_channels * 16}; |
+} |
+ |
+std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder( |
+ const AudioEncoderL16::Config& config, |
+ int payload_type) { |
+ RTC_DCHECK(config.IsOk()); |
+ AudioEncoderPcm16B::Config c; |
+ c.sample_rate_hz = config.sample_rate_hz; |
+ c.num_channels = config.num_channels; |
+ c.frame_size_ms = config.frame_size_ms; |
+ c.payload_type = payload_type; |
+ return rtc::MakeUnique<AudioEncoderPcm16B>(c); |
+} |
+ |
+} // namespace webrtc |