Line data Source code
1 : /*
2 : * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 : *
4 : * Use of this source code is governed by a BSD-style license
5 : * that can be found in the LICENSE file in the root of the source
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : */
10 :
11 : #include "webrtc/audio/audio_transport_proxy.h"
12 :
13 : namespace webrtc {
14 :
15 : namespace {
16 : // Resample audio in |frame| to given sample rate preserving the
17 : // channel count and place the result in |destination|.
18 0 : int Resample(const AudioFrame& frame,
19 : const int destination_sample_rate,
20 : PushResampler<int16_t>* resampler,
21 : int16_t* destination) {
22 0 : const int number_of_channels = static_cast<int>(frame.num_channels_);
23 : const int target_number_of_samples_per_channel =
24 0 : destination_sample_rate / 100;
25 0 : resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
26 0 : number_of_channels);
27 :
28 0 : return resampler->Resample(
29 0 : frame.data_, frame.samples_per_channel_ * number_of_channels, destination,
30 0 : number_of_channels * target_number_of_samples_per_channel);
31 : }
32 : } // namespace
33 :
34 0 : AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
35 : AudioProcessing* apm,
36 0 : AudioMixer* mixer)
37 0 : : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) {
38 0 : RTC_DCHECK(voe_audio_transport);
39 0 : RTC_DCHECK(apm);
40 0 : RTC_DCHECK(mixer);
41 0 : }
42 :
43 0 : AudioTransportProxy::~AudioTransportProxy() {}
44 :
45 0 : int32_t AudioTransportProxy::RecordedDataIsAvailable(
46 : const void* audioSamples,
47 : const size_t nSamples,
48 : const size_t nBytesPerSample,
49 : const size_t nChannels,
50 : const uint32_t samplesPerSec,
51 : const uint32_t totalDelayMS,
52 : const int32_t clockDrift,
53 : const uint32_t currentMicLevel,
54 : const bool keyPressed,
55 : uint32_t& newMicLevel) {
56 : // Pass call through to original audio transport instance.
57 0 : return voe_audio_transport_->RecordedDataIsAvailable(
58 : audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
59 0 : totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
60 : }
61 :
62 0 : int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
63 : const size_t nBytesPerSample,
64 : const size_t nChannels,
65 : const uint32_t samplesPerSec,
66 : void* audioSamples,
67 : size_t& nSamplesOut,
68 : int64_t* elapsed_time_ms,
69 : int64_t* ntp_time_ms) {
70 0 : RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
71 0 : RTC_DCHECK_GE(nChannels, 1);
72 0 : RTC_DCHECK_LE(nChannels, 2);
73 0 : RTC_DCHECK_GE(
74 : samplesPerSec,
75 0 : static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
76 :
77 : // 100 = 1 second / data duration (10 ms).
78 0 : RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
79 0 : RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
80 0 : sizeof(AudioFrame::data_));
81 :
82 0 : mixer_->Mix(nChannels, &mixed_frame_);
83 0 : *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
84 0 : *ntp_time_ms = mixed_frame_.ntp_time_ms_;
85 :
86 0 : const auto error = apm_->ProcessReverseStream(&mixed_frame_);
87 0 : RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
88 :
89 0 : nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_,
90 : static_cast<int16_t*>(audioSamples));
91 0 : RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
92 0 : return 0;
93 : }
94 :
95 0 : void AudioTransportProxy::PushCaptureData(int voe_channel,
96 : const void* audio_data,
97 : int bits_per_sample,
98 : int sample_rate,
99 : size_t number_of_channels,
100 : size_t number_of_frames) {
101 : // This is part of deprecated VoE interface operating on specific
102 : // VoE channels. It should not be used.
103 0 : RTC_NOTREACHED();
104 0 : }
105 :
106 0 : void AudioTransportProxy::PullRenderData(int bits_per_sample,
107 : int sample_rate,
108 : size_t number_of_channels,
109 : size_t number_of_frames,
110 : void* audio_data,
111 : int64_t* elapsed_time_ms,
112 : int64_t* ntp_time_ms) {
113 0 : RTC_DCHECK_EQ(bits_per_sample, 16);
114 0 : RTC_DCHECK_GE(number_of_channels, 1);
115 0 : RTC_DCHECK_LE(number_of_channels, 2);
116 0 : RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
117 :
118 : // 100 = 1 second / data duration (10 ms).
119 0 : RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
120 :
121 : // 8 = bits per byte.
122 0 : RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
123 0 : sizeof(AudioFrame::data_));
124 0 : mixer_->Mix(number_of_channels, &mixed_frame_);
125 0 : *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
126 0 : *ntp_time_ms = mixed_frame_.ntp_time_ms_;
127 :
128 0 : const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
129 0 : static_cast<int16_t*>(audio_data));
130 0 : RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
131 0 : }
132 :
133 : } // namespace webrtc
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