Introduce CopyToFileAudioCapturer.
It will be used to dump generated audio from TestAudioDeviceModule into user defuned file in peer connection level test framework. Bug: webrtc:10138 Change-Id: I6e3db36aaf1303ab148e8812937c4f9cd1b49315 Reviewed-on: https://webrtc-review.googlesource.com/c/117220 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26267}
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@ -64,12 +64,12 @@ class TestAudioDeviceModule : public AudioDeviceModule {
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// -max_amplitude and +max_amplitude.
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class PulsedNoiseCapturer : public Capturer {
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public:
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virtual ~PulsedNoiseCapturer() {}
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~PulsedNoiseCapturer() override {}
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virtual void SetMaxAmplitude(int16_t amplitude) = 0;
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};
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virtual ~TestAudioDeviceModule() {}
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~TestAudioDeviceModule() override {}
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// Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
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// frames will be processed every 10ms / |speed|.
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@ -150,16 +150,16 @@ class TestAudioDeviceModule : public AudioDeviceModule {
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int sampling_frequency_in_hz,
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int num_channels = 1);
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virtual int32_t Init() = 0;
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virtual int32_t RegisterAudioCallback(AudioTransport* callback) = 0;
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int32_t Init() override = 0;
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int32_t RegisterAudioCallback(AudioTransport* callback) override = 0;
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virtual int32_t StartPlayout() = 0;
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virtual int32_t StopPlayout() = 0;
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virtual int32_t StartRecording() = 0;
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virtual int32_t StopRecording() = 0;
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int32_t StartPlayout() override = 0;
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int32_t StopPlayout() override = 0;
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int32_t StartRecording() override = 0;
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int32_t StopRecording() override = 0;
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virtual bool Playing() const = 0;
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virtual bool Recording() const = 0;
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bool Playing() const override = 0;
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bool Recording() const override = 0;
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// Blocks until the Renderer refuses to receive data.
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// Returns false if |timeout_ms| passes before that happens.
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