Index: webrtc/audio/audio_transport_proxy.h |
diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..a4fc99415071d3625cdd85d56bf36b173680f39f |
--- /dev/null |
+++ b/webrtc/audio/audio_transport_proxy.h |
@@ -0,0 +1,130 @@ |
+/* |
+ * Copyright (c) 2016 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. |
+ */ |
+ |
+#ifndef WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |
+#define WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |
+ |
+#include "webrtc/api/audio/audio_mixer.h" |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/modules/audio_device/include/audio_device_defines.h" |
+#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+ |
+namespace webrtc { |
+ |
+class AudioTransportProxy : public AudioTransport { |
+ public: |
+ AudioTransportProxy(AudioTransport* voe_audio_transport, |
+ AudioProcessing* apm, |
+ AudioMixer* mixer) |
+ : voe_audio_transport_(voe_audio_transport) {} |
+ |
+ ~AudioTransportProxy() override {} |
+ |
+ int32_t RecordedDataIsAvailable(const void* audioSamples, |
+ const size_t nSamples, |
+ const size_t nBytesPerSample, |
+ const size_t nChannels, |
+ const uint32_t samplesPerSec, |
+ const uint32_t totalDelayMS, |
+ const int32_t clockDrift, |
+ const uint32_t currentMicLevel, |
+ const bool keyPressed, |
+ uint32_t& newMicLevel) override { |
+ // Pass call through to original audio transport instance. |
+ if (voe_audio_transport_) { |
+ return voe_audio_transport_->RecordedDataIsAvailable( |
+ audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec, |
+ totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel); |
+ } |
+ |
+ RTC_NOTREACHED() |
+ << "AudioTransport proxy doesn't know where to send recorded data: " |
+ "no Audio Transport provided."; |
+ return -1; |
+ } |
+ |
+ int32_t NeedMorePlayData(const size_t nSamples, |
+ const size_t nBytesPerSample, |
+ const size_t nChannels, |
+ const uint32_t samplesPerSec, |
+ void* audioSamples, |
+ size_t& nSamplesOut, |
+ int64_t* elapsed_time_ms, |
+ int64_t* ntp_time_ms) override { |
+ RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample); |
+ RTC_DCHECK_LE(1u, nChannels); |
ossu
2016/11/03 14:02:04
I'd prefer if the parameters in the two nChannels
aleloi
2016/11/08 11:46:41
Done.
|
+ RTC_DCHECK_LE(nChannels, 2u); |
+ RTC_DCHECK_LE( |
ossu
2016/11/03 14:02:04
Alright, so maybe a bit nitpicky, but don't we usu
aleloi
2016/11/08 11:46:41
I made comparisons to constants consistent in the
|
+ static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz), |
+ samplesPerSec); |
+ RTC_DCHECK_EQ(nSamples * 100, samplesPerSec); |
+ RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels, |
+ sizeof(AudioFrame::data_)); |
+ |
+ // Pass call through to original audio transport instance. |
+ if (voe_audio_transport_) { |
+ return voe_audio_transport_->NeedMorePlayData( |
+ nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples, |
+ nSamplesOut, elapsed_time_ms, ntp_time_ms); |
+ } |
+ |
+ RTC_NOTREACHED() |
+ << "AudioTransport proxy doesn't know from where to get play data: " |
+ "no Audio Transport provided."; |
+ return -1; |
+ } |
+ |
+ void PushCaptureData(int voe_channel, |
+ const void* audio_data, |
+ int bits_per_sample, |
+ int sample_rate, |
+ size_t number_of_channels, |
+ size_t number_of_frames) override { |
+ // This is part of deprecated VoE interface operating on specific |
+ // VoE channels. It should not be used. |
+ RTC_NOTREACHED(); |
+ } |
+ |
+ void PullRenderData(int bits_per_sample, |
+ int sample_rate, |
+ size_t number_of_channels, |
+ size_t number_of_frames, |
+ void* audio_data, |
+ int64_t* elapsed_time_ms, |
+ int64_t* ntp_time_ms) override { |
+ RTC_DCHECK_EQ(8 * sizeof(int16_t), |
+ static_cast<size_t>(bits_per_sample)); |
+ RTC_DCHECK_LE(1u, number_of_frames); |
+ RTC_DCHECK_LE(number_of_frames, 2u); |
+ RTC_DCHECK_LE(AudioProcessing::NativeRate::kSampleRate8kHz, |
+ static_cast<int>(sample_rate)); |
+ RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate); |
+ RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, |
+ sizeof(AudioFrame::data_)); |
+ if (voe_audio_transport_) { |
+ return voe_audio_transport_->PullRenderData( |
+ bits_per_sample, sample_rate, number_of_channels, number_of_frames, |
+ audio_data, elapsed_time_ms, ntp_time_ms); |
+ } |
+ |
+ RTC_NOTREACHED() |
+ << "AudioTransport proxy doesn't know from where to get play data: " |
+ "no Audio Transport provided."; |
+ } |
+ |
+ private: |
+ AudioTransport* voe_audio_transport_; |
+ AudioFrame frame_for_mixing_; |
+ |
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy); |
+}; |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |