990d6b875e
This reverts commit 90bace095806a635411edd40fb8490a144e59e63. Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it. Original change's description: > Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API > > (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) > > This SetAudioPlayout method lets applications disable audio playout while > still processing incoming audio data and generating statistics on the > received audio. > > This may be useful if the application wants to set up media flows as > soon as possible, but isn't ready to play audio yet. Currently, native > applications don't have any API point to control this, unless they > completely implement their own AudioDeviceModule. > > The SetAudioRecording works in a similar fashion but for the recorded > audio. One difference is that calling SetAudioRecording(false) does not > keep any audio processing alive. > > TBR=solenberg > > Bug: webrtc:7313 > Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa > Reviewed-on: https://webrtc-review.googlesource.com/16180 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20499} TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7313 Reviewed-on: https://webrtc-review.googlesource.com/17701 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20512}
111 lines
4.1 KiB
C++
111 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef VOICE_ENGINE_VOE_BASE_IMPL_H_
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#define VOICE_ENGINE_VOE_BASE_IMPL_H_
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#include "voice_engine/include/voe_base.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/criticalsection.h"
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#include "voice_engine/shared_data.h"
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namespace webrtc {
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class ProcessThread;
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class VoEBaseImpl : public VoEBase,
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public AudioTransport {
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public:
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int Init(
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AudioDeviceModule* external_adm,
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AudioProcessing* audio_processing,
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const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override;
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AudioDeviceModule* audio_device_module() override {
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return shared_->audio_device();
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}
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voe::TransmitMixer* transmit_mixer() override {
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return shared_->transmit_mixer();
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}
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int Terminate() override;
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int CreateChannel() override;
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int CreateChannel(const ChannelConfig& config) override;
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int DeleteChannel(int channel) override;
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int StartPlayout(int channel) override;
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int StartSend(int channel) override;
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int StopPlayout(int channel) override;
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int StopSend(int channel) override;
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AudioTransport* audio_transport() override { return this; }
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// AudioTransport
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int32_t RecordedDataIsAvailable(const void* audio_data,
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const size_t number_of_frames,
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const size_t bytes_per_sample,
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const size_t number_of_channels,
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const uint32_t sample_rate,
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const uint32_t audio_delay_milliseconds,
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const int32_t clock_drift,
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const uint32_t volume,
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const bool key_pressed,
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uint32_t& new_mic_volume) override;
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RTC_DEPRECATED int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void PushCaptureData(int voe_channel,
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const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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RTC_DEPRECATED void PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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protected:
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VoEBaseImpl(voe::SharedData* shared);
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~VoEBaseImpl() override;
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private:
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int32_t StartPlayout();
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int32_t StopPlayout();
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int32_t StartSend();
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int32_t StopSend();
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int32_t TerminateInternal();
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void GetPlayoutData(int sample_rate, size_t number_of_channels,
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size_t number_of_frames, bool feed_data_to_apm,
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void* audio_data, int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms);
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// Initialize channel by setting Engine Information then initializing
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// channel.
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int InitializeChannel(voe::ChannelOwner* channel_owner);
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
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AudioFrame audioFrame_;
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voe::SharedData* shared_;
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};
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} // namespace webrtc
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#endif // VOICE_ENGINE_VOE_BASE_IMPL_H_
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