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authorMoonchild <moonchild@palemoon.org>2019-11-03 11:20:08 +0100
committerGitHub <noreply@github.com>2019-11-03 11:20:08 +0100
commitb7c6c779ee1d0100842822a1a9c63cd97d27644b (patch)
treecd6504982a432efda7dcd0b5208efde4c04847b1 /media/libcubeb/src/cubeb_resampler_internal.h
parent21b3f6247403c06f85e1f45d219f87549862198f (diff)
parent22b35fa8e923d52a3fa785993c28c3e63cd1ee1e (diff)
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Merge pull request #1270 from g4jc/libcubeb
Update libcubeb
Diffstat (limited to 'media/libcubeb/src/cubeb_resampler_internal.h')
-rw-r--r--media/libcubeb/src/cubeb_resampler_internal.h123
1 files changed, 87 insertions, 36 deletions
diff --git a/media/libcubeb/src/cubeb_resampler_internal.h b/media/libcubeb/src/cubeb_resampler_internal.h
index 3c37a04b9..fb69992ff 100644
--- a/media/libcubeb/src/cubeb_resampler_internal.h
+++ b/media/libcubeb/src/cubeb_resampler_internal.h
@@ -39,6 +39,13 @@ MOZ_END_STD_NAMESPACE
/* This header file contains the internal C++ API of the resamplers, for testing. */
+// When dropping audio input frames to prevent building
+// an input delay, this function returns the number of frames
+// to keep in the buffer.
+// @parameter sample_rate The sample rate of the stream.
+// @return A number of frames to keep.
+uint32_t min_buffered_audio_frame(uint32_t sample_rate);
+
int to_speex_quality(cubeb_resampler_quality q);
struct cubeb_resampler {
@@ -48,31 +55,6 @@ struct cubeb_resampler {
virtual ~cubeb_resampler() {}
};
-class noop_resampler : public cubeb_resampler {
-public:
- noop_resampler(cubeb_stream * s,
- cubeb_data_callback cb,
- void * ptr)
- : stream(s)
- , data_callback(cb)
- , user_ptr(ptr)
- {
- }
-
- virtual long fill(void * input_buffer, long * input_frames_count,
- void * output_buffer, long output_frames);
-
- virtual long latency()
- {
- return 0;
- }
-
-private:
- cubeb_stream * const stream;
- const cubeb_data_callback data_callback;
- void * const user_ptr;
-};
-
/** Base class for processors. This is just used to share methods for now. */
class processor {
public:
@@ -80,11 +62,11 @@ public:
: channels(channels)
{}
protected:
- size_t frames_to_samples(size_t frames)
+ size_t frames_to_samples(size_t frames) const
{
return frames * channels;
}
- size_t samples_to_frames(size_t samples)
+ size_t samples_to_frames(size_t samples) const
{
assert(!(samples % channels));
return samples / channels;
@@ -93,6 +75,43 @@ protected:
const uint32_t channels;
};
+template<typename T>
+class passthrough_resampler : public cubeb_resampler
+ , public processor {
+public:
+ passthrough_resampler(cubeb_stream * s,
+ cubeb_data_callback cb,
+ void * ptr,
+ uint32_t input_channels,
+ uint32_t sample_rate);
+
+ virtual long fill(void * input_buffer, long * input_frames_count,
+ void * output_buffer, long output_frames);
+
+ virtual long latency()
+ {
+ return 0;
+ }
+
+ void drop_audio_if_needed()
+ {
+ uint32_t to_keep = min_buffered_audio_frame(sample_rate);
+ uint32_t available = samples_to_frames(internal_input_buffer.length());
+ if (available > to_keep) {
+ internal_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
+ }
+ }
+
+private:
+ cubeb_stream * const stream;
+ const cubeb_data_callback data_callback;
+ void * const user_ptr;
+ /* This allows to buffer some input to account for the fact that we buffer
+ * some inputs. */
+ auto_array<T> internal_input_buffer;
+ uint32_t sample_rate;
+};
+
/** Bidirectional resampler, can resample an input and an output stream, or just
* an input stream or output stream. In this case a delay is inserted in the
* opposite direction to keep the streams synchronized. */
@@ -138,6 +157,7 @@ private:
cubeb_stream * const stream;
const cubeb_data_callback data_callback;
void * const user_ptr;
+ bool draining = false;
};
/** Handles one way of a (possibly) duplex resampler, working on interleaved
@@ -163,6 +183,7 @@ public:
int quality)
: processor(channels)
, resampling_ratio(static_cast<float>(source_rate) / target_rate)
+ , source_rate(source_rate)
, additional_latency(0)
, leftover_samples(0)
{
@@ -221,7 +242,7 @@ public:
/** Returns a buffer containing exactly `output_frame_count` resampled frames.
* The consumer should not hold onto the pointer. */
- T * output(size_t output_frame_count)
+ T * output(size_t output_frame_count, size_t * input_frames_used)
{
if (resampling_out_buffer.capacity() < frames_to_samples(output_frame_count)) {
resampling_out_buffer.reserve(frames_to_samples(output_frame_count));
@@ -238,6 +259,7 @@ public:
/* This shifts back any unresampled samples to the beginning of the input
buffer. */
resampling_in_buffer.pop(nullptr, frames_to_samples(in_len));
+ *input_frames_used = in_len;
return resampling_out_buffer.data();
}
@@ -261,7 +283,7 @@ public:
* exactly `output_frame_count` resampled frames. This can return a number
* slightly bigger than what is strictly necessary, but it guaranteed that the
* number of output frames will be exactly equal. */
- uint32_t input_needed_for_output(uint32_t output_frame_count)
+ uint32_t input_needed_for_output(uint32_t output_frame_count) const
{
int32_t unresampled_frames_left = samples_to_frames(resampling_in_buffer.length());
int32_t resampled_frames_left = samples_to_frames(resampling_out_buffer.length());
@@ -294,6 +316,16 @@ public:
resampling_in_buffer.set_length(leftover_samples +
frames_to_samples(written_frames));
}
+
+ void drop_audio_if_needed()
+ {
+ // Keep at most 100ms buffered.
+ uint32_t available = samples_to_frames(resampling_in_buffer.length());
+ uint32_t to_keep = min_buffered_audio_frame(source_rate);
+ if (available > to_keep) {
+ resampling_in_buffer.pop(nullptr, frames_to_samples(available - to_keep));
+ }
+ }
private:
/** Wrapper for the speex resampling functions to have a typed
* interface. */
@@ -330,6 +362,7 @@ private:
SpeexResamplerState * speex_resampler;
/** Source rate / target rate. */
const float resampling_ratio;
+ const uint32_t source_rate;
/** Storage for the input frames, to be resampled. Also contains
* any unresampled frames after resampling. */
auto_array<T> resampling_in_buffer;
@@ -348,11 +381,13 @@ class delay_line : public processor {
public:
/** Constructor
* @parameter frames the number of frames of delay.
- * @parameter channels the number of channels of this delay line. */
- delay_line(uint32_t frames, uint32_t channels)
+ * @parameter channels the number of channels of this delay line.
+ * @parameter sample_rate sample-rate of the audio going through this delay line */
+ delay_line(uint32_t frames, uint32_t channels, uint32_t sample_rate)
: processor(channels)
, length(frames)
, leftover_samples(0)
+ , sample_rate(sample_rate)
{
/* Fill the delay line with some silent frames to add latency. */
delay_input_buffer.push_silence(frames * channels);
@@ -375,7 +410,7 @@ public:
* @parameter frames_needed the number of frames to be returned.
* @return a buffer containing the delayed frames. The consumer should not
* hold onto the pointer. */
- T * output(uint32_t frames_needed)
+ T * output(uint32_t frames_needed, size_t * input_frames_used)
{
if (delay_output_buffer.capacity() < frames_to_samples(frames_needed)) {
delay_output_buffer.reserve(frames_to_samples(frames_needed));
@@ -385,6 +420,7 @@ public:
delay_output_buffer.push(delay_input_buffer.data(),
frames_to_samples(frames_needed));
delay_input_buffer.pop(nullptr, frames_to_samples(frames_needed));
+ *input_frames_used = frames_needed;
return delay_output_buffer.data();
}
@@ -426,7 +462,7 @@ public:
* @parameter frames_needed the number of frames one want to write into the
* delay_line
* @returns the number of frames one will get. */
- size_t input_needed_for_output(uint32_t frames_needed)
+ size_t input_needed_for_output(uint32_t frames_needed) const
{
return frames_needed;
}
@@ -441,6 +477,15 @@ public:
{
return length;
}
+
+ void drop_audio_if_needed()
+ {
+ size_t available = samples_to_frames(delay_input_buffer.length());
+ uint32_t to_keep = min_buffered_audio_frame(sample_rate);
+ if (available > to_keep) {
+ delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
+ }
+ }
private:
/** The length, in frames, of this delay line */
uint32_t length;
@@ -452,6 +497,7 @@ private:
/** The output buffer. This is only ever used if using the ::output with a
* single argument. */
auto_array<T> delay_output_buffer;
+ uint32_t sample_rate;
};
/** This sits behind the C API and is more typed. */
@@ -480,7 +526,10 @@ cubeb_resampler_create_internal(cubeb_stream * stream,
(output_params && output_params->rate == target_rate)) ||
(input_params && !output_params && (input_params->rate == target_rate)) ||
(output_params && !input_params && (output_params->rate == target_rate))) {
- return new noop_resampler(stream, callback, user_ptr);
+ return new passthrough_resampler<T>(stream, callback,
+ user_ptr,
+ input_params ? input_params->channels : 0,
+ target_rate);
}
/* Determine if we need to resampler one or both directions, and create the
@@ -512,13 +561,15 @@ cubeb_resampler_create_internal(cubeb_stream * stream,
* other direction so that the streams are synchronized. */
if (input_resampler && !output_resampler && input_params && output_params) {
output_delay.reset(new delay_line<T>(input_resampler->latency(),
- output_params->channels));
+ output_params->channels,
+ output_params->rate));
if (!output_delay) {
return NULL;
}
} else if (output_resampler && !input_resampler && input_params && output_params) {
input_delay.reset(new delay_line<T>(output_resampler->latency(),
- input_params->channels));
+ input_params->channels,
+ output_params->rate));
if (!input_delay) {
return NULL;
}